Rename cricket::MEDIA_TYPE_(AUDIO/VIDEO)
to webrtc::MediaType::(AUDIO/VIDEO)
This was generated using multiple passes of git grep -l | sed.
No-Iwyu: https://issues.webrtc.org/406288660
Bug: webrtc:42222911
Change-Id: I9a697bb6f40c60ffe0e6b61a2a3e07120c2f619e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/382500
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44222}
diff --git a/api/crypto/BUILD.gn b/api/crypto/BUILD.gn
index d1f9b13..2970f34 100644
--- a/api/crypto/BUILD.gn
+++ b/api/crypto/BUILD.gn
@@ -34,6 +34,7 @@
sources = [ "frame_decryptor_interface.h" ]
deps = [
"..:array_view",
+ "..:ref_count",
"..:rtp_parameters",
"../../rtc_base:refcount",
]
@@ -44,6 +45,7 @@
sources = [ "frame_encryptor_interface.h" ]
deps = [
"..:array_view",
+ "..:ref_count",
"..:rtp_parameters",
"../../rtc_base:refcount",
]
diff --git a/api/crypto/frame_decryptor_interface.h b/api/crypto/frame_decryptor_interface.h
index f412300..d37ed91 100644
--- a/api/crypto/frame_decryptor_interface.h
+++ b/api/crypto/frame_decryptor_interface.h
@@ -11,11 +11,13 @@
#ifndef API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_
#define API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_
+#include <cstddef>
+#include <cstdint>
#include <vector>
#include "api/array_view.h"
#include "api/media_types.h"
-#include "rtc_base/ref_count.h"
+#include "api/ref_count.h"
namespace webrtc {
@@ -58,7 +60,7 @@
// kRecoverable should be returned if the failure was due to something other
// than a decryption failure. kFailedToDecrypt should be returned in all other
// cases.
- virtual Result Decrypt(cricket::MediaType media_type,
+ virtual Result Decrypt(webrtc::MediaType media_type,
const std::vector<uint32_t>& csrcs,
rtc::ArrayView<const uint8_t> additional_data,
rtc::ArrayView<const uint8_t> encrypted_frame,
@@ -67,7 +69,7 @@
// Returns the total required length in bytes for the output of the
// decryption. This can be larger than the actual number of bytes you need but
// must never be smaller as it informs the size of the frame buffer.
- virtual size_t GetMaxPlaintextByteSize(cricket::MediaType media_type,
+ virtual size_t GetMaxPlaintextByteSize(webrtc::MediaType media_type,
size_t encrypted_frame_size) = 0;
};
diff --git a/api/crypto/frame_encryptor_interface.h b/api/crypto/frame_encryptor_interface.h
index 99148bd..dd38356 100644
--- a/api/crypto/frame_encryptor_interface.h
+++ b/api/crypto/frame_encryptor_interface.h
@@ -11,9 +11,12 @@
#ifndef API_CRYPTO_FRAME_ENCRYPTOR_INTERFACE_H_
#define API_CRYPTO_FRAME_ENCRYPTOR_INTERFACE_H_
+#include <cstddef>
+#include <cstdint>
+
#include "api/array_view.h"
#include "api/media_types.h"
-#include "rtc_base/ref_count.h"
+#include "api/ref_count.h"
namespace webrtc {
@@ -35,7 +38,7 @@
// must set bytes_written to the number of bytes you wrote in the
// encrypted_frame. 0 must be returned if successful all other numbers can be
// selected by the implementer to represent error codes.
- virtual int Encrypt(cricket::MediaType media_type,
+ virtual int Encrypt(webrtc::MediaType media_type,
uint32_t ssrc,
rtc::ArrayView<const uint8_t> additional_data,
rtc::ArrayView<const uint8_t> frame,
@@ -45,7 +48,7 @@
// Returns the total required length in bytes for the output of the
// encryption. This can be larger than the actual number of bytes you need but
// must never be smaller as it informs the size of the encrypted_frame buffer.
- virtual size_t GetMaxCiphertextByteSize(cricket::MediaType media_type,
+ virtual size_t GetMaxCiphertextByteSize(webrtc::MediaType media_type,
size_t frame_size) = 0;
};
diff --git a/api/media_types.h b/api/media_types.h
index 339c263..2e1ab86 100644
--- a/api/media_types.h
+++ b/api/media_types.h
@@ -23,7 +23,7 @@
DATA,
UNSUPPORTED,
ANY,
- // Backwards compatibility values for cricket::MediaType users
+ // Backwards compatibility values for webrtc::MediaType users
// TODO: https://issues.webrtc.org/42222911 - deprecate and remove
MEDIA_TYPE_AUDIO = AUDIO,
MEDIA_TYPE_VIDEO = VIDEO,
@@ -56,7 +56,7 @@
using webrtc::kMediaTypeVideo;
using webrtc::MediaTypeToString;
-// Backwards compatibility values for cricket::MediaType users
+// Backwards compatibility values for webrtc::MediaType users
// TODO: https://issues.webrtc.org/42222911 - deprecate and remove
constexpr MediaType MEDIA_TYPE_AUDIO = webrtc::MediaType::AUDIO;
constexpr MediaType MEDIA_TYPE_VIDEO = webrtc::MediaType::VIDEO;
diff --git a/api/media_types_unittest.cc b/api/media_types_unittest.cc
deleted file mode 100644
index aefb506..0000000
--- a/api/media_types_unittest.cc
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "api/media_types.h"
-
-#include "test/gmock.h"
-#include "test/gtest.h"
-
-namespace {
-
-TEST(MediaTypeTest, Assignment) {
- webrtc::MediaType type_w;
- cricket::MediaType type_c;
- // If this compiles, the test passes.
- type_w = webrtc::MediaType::VIDEO;
- type_c = webrtc::MediaType::VIDEO;
- type_w = type_c;
- type_c = type_w;
- // The older constant names.
- type_w = cricket::MediaType::MEDIA_TYPE_VIDEO;
- type_w = cricket::MEDIA_TYPE_VIDEOO;
- type_c = cricket::MediaType::MEDIA_TYPE_VIDEO;
- type_c = cricket::MEDIA_TYPE_VIDEO;
-}
-
-TEST(MediaTypeTest, AutomaticConversionFromInteger) {
- webrtc::MediaType type_w;
- type_w = 4;
-}
-
-TEST(MediaTypeTest, AutomaticConversionToInteger) {
- webrtc::MediaType type_w;
- cricket::MediaType type_c;
- int type_i;
- // If this compiles, the test passes.
- type_w = webrtc::MediaType::VIDEO;
- type_c = webrtc::MediaType::VIDEO;
- type_i = webrtc::MediaType::VIDEO;
- // Explicitly invoking the converter works.
- type_i = cricket::MediaTypeToInt(webrtc::MediaType::VIDEO);
- type_i = type_w;
- type_i = type_c;
-}
-
-} // namespace
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index b8a12a6..0039dd2 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -861,15 +861,14 @@
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) = 0;
- // Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
- // MEDIA_TYPE_VIDEO.
- // Errors:
- // - INVALID_PARAMETER: `media_type` is not MEDIA_TYPE_AUDIO or
- // MEDIA_TYPE_VIDEO.
+ // Adds a transceiver with the given kind. Can either be
+ // webrtc::MediaType::AUDIO or webrtc::MediaType::VIDEO. Errors:
+ // - INVALID_PARAMETER: `media_type` is not webrtc::MediaType::AUDIO or
+ // webrtc::MediaType::VIDEO.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
- AddTransceiver(cricket::MediaType media_type) = 0;
+ AddTransceiver(webrtc::MediaType media_type) = 0;
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
- AddTransceiver(cricket::MediaType media_type,
+ AddTransceiver(webrtc::MediaType media_type,
const RtpTransceiverInit& init) = 0;
// Creates a sender without a track. Can be used for "early media"/"warmup"
@@ -1541,14 +1540,16 @@
PeerConnectionDependencies dependencies) = 0;
// Returns the capabilities of an RTP sender of type `kind`.
- // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
+ // If for some reason you pass in webrtc::MediaType::DATA, returns an empty
+ // structure.
virtual RtpCapabilities GetRtpSenderCapabilities(
- cricket::MediaType kind) const = 0;
+ webrtc::MediaType kind) const = 0;
// Returns the capabilities of an RTP receiver of type `kind`.
- // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
+ // If for some reason you pass in webrtc::MediaType::DATA, returns an empty
+ // structure.
virtual RtpCapabilities GetRtpReceiverCapabilities(
- cricket::MediaType kind) const = 0;
+ webrtc::MediaType kind) const = 0;
virtual rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
const std::string& stream_id) = 0;
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index a1fce6a..57d77f2 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -137,7 +137,7 @@
std::string name;
// The media type of this codec. Equivalent to MIME top-level type.
- cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
+ webrtc::MediaType kind = webrtc::MediaType::AUDIO;
// If unset, the implementation default is used.
std::optional<int> clock_rate;
@@ -195,7 +195,7 @@
template <typename Sink>
friend void AbslStringify(Sink& sink, const RtpCodecCapability& cap) {
- if (cap.kind == cricket::MEDIA_TYPE_AUDIO) {
+ if (cap.kind == webrtc::MediaType::AUDIO) {
absl::Format(&sink, "[audio/%s/%d/%d]", cap.name,
cap.clock_rate.value_or(0), cap.num_channels.value_or(1));
} else {
@@ -213,8 +213,9 @@
// RtpHeaderExtensionParameters.
//
// Note that ORTC includes a "kind" field, but we omit this because it's
-// redundant; if you call "RtpReceiver::GetCapabilities(MEDIA_TYPE_AUDIO)",
-// you know you're getting audio capabilities.
+// redundant; if you call
+// "RtpReceiver::GetCapabilities(webrtc::MediaType::AUDIO)", you know you're
+// getting audio capabilities.
struct RTC_EXPORT RtpHeaderExtensionCapability {
// URI of this extension, as defined in RFC8285.
std::string uri;
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index bad3d2c..a37d69a 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -40,7 +40,7 @@
// In the future, it's likely that an RtpReceiver will only call
// OnFirstPacketReceived when a packet is received specifically for its
// SSRC/mid.
- virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
+ virtual void OnFirstPacketReceived(webrtc::MediaType media_type) = 0;
protected:
virtual ~RtpReceiverObserverInterface() {}
@@ -68,7 +68,7 @@
virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const;
// Audio or video receiver?
- virtual cricket::MediaType media_type() const = 0;
+ virtual webrtc::MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index 91d7b3c..524c654 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -40,7 +40,7 @@
public:
// The observer is called when the first media packet is sent for the observed
// sender. It is called immediately if the first packet was already sent.
- virtual void OnFirstPacketSent(cricket::MediaType media_type) = 0;
+ virtual void OnFirstPacketSent(webrtc::MediaType media_type) = 0;
protected:
virtual ~RtpSenderObserverInterface() {}
@@ -68,7 +68,7 @@
virtual uint32_t ssrc() const = 0;
// Audio or video sender?
- virtual cricket::MediaType media_type() const = 0;
+ virtual webrtc::MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
diff --git a/api/rtp_transceiver_interface.h b/api/rtp_transceiver_interface.h
index 2d0bf29..c9c4145 100644
--- a/api/rtp_transceiver_interface.h
+++ b/api/rtp_transceiver_interface.h
@@ -63,7 +63,7 @@
public:
// Media type of the transceiver. Any sender(s)/receiver(s) will have this
// type as well.
- virtual cricket::MediaType media_type() const = 0;
+ virtual webrtc::MediaType media_type() const = 0;
// The mid attribute is the mid negotiated and present in the local and
// remote descriptions. Before negotiation is complete, the mid value may be
diff --git a/api/test/fake_frame_decryptor.cc b/api/test/fake_frame_decryptor.cc
index e5bcc57..4241d99 100644
--- a/api/test/fake_frame_decryptor.cc
+++ b/api/test/fake_frame_decryptor.cc
@@ -25,7 +25,7 @@
: fake_key_(fake_key), expected_postfix_byte_(expected_postfix_byte) {}
FakeFrameDecryptor::Result FakeFrameDecryptor::Decrypt(
- cricket::MediaType /* media_type */,
+ webrtc::MediaType /* media_type */,
const std::vector<uint32_t>& /* csrcs */,
rtc::ArrayView<const uint8_t> /* additional_data */,
rtc::ArrayView<const uint8_t> encrypted_frame,
@@ -47,7 +47,7 @@
}
size_t FakeFrameDecryptor::GetMaxPlaintextByteSize(
- cricket::MediaType /* media_type */,
+ webrtc::MediaType /* media_type */,
size_t encrypted_frame_size) {
return encrypted_frame_size - 1;
}
diff --git a/api/test/fake_frame_decryptor.h b/api/test/fake_frame_decryptor.h
index 783bc80..bbbeb39 100644
--- a/api/test/fake_frame_decryptor.h
+++ b/api/test/fake_frame_decryptor.h
@@ -34,13 +34,13 @@
uint8_t expected_postfix_byte = 255);
// Fake decryption that just xors the payload with the 1 byte key and checks
// the postfix byte. This will always fail if fail_decryption_ is set to true.
- Result Decrypt(cricket::MediaType media_type,
+ Result Decrypt(webrtc::MediaType media_type,
const std::vector<uint32_t>& csrcs,
rtc::ArrayView<const uint8_t> additional_data,
rtc::ArrayView<const uint8_t> encrypted_frame,
rtc::ArrayView<uint8_t> frame) override;
// Always returns 1 less than the size of the encrypted frame.
- size_t GetMaxPlaintextByteSize(cricket::MediaType media_type,
+ size_t GetMaxPlaintextByteSize(webrtc::MediaType media_type,
size_t encrypted_frame_size) override;
// Sets the fake key to use for encryption.
void SetFakeKey(uint8_t fake_key);
diff --git a/api/test/fake_frame_encryptor.cc b/api/test/fake_frame_encryptor.cc
index ec31596..cfa371e 100644
--- a/api/test/fake_frame_encryptor.cc
+++ b/api/test/fake_frame_encryptor.cc
@@ -23,7 +23,7 @@
// FrameEncryptorInterface implementation
int FakeFrameEncryptor::Encrypt(
- cricket::MediaType /* media_type */,
+ webrtc::MediaType /* media_type */,
uint32_t /* ssrc */,
rtc::ArrayView<const uint8_t> /* additional_data */,
rtc::ArrayView<const uint8_t> frame,
@@ -44,7 +44,7 @@
}
size_t FakeFrameEncryptor::GetMaxCiphertextByteSize(
- cricket::MediaType /* media_type */,
+ webrtc::MediaType /* media_type */,
size_t frame_size) {
return frame_size + 1;
}
diff --git a/api/test/fake_frame_encryptor.h b/api/test/fake_frame_encryptor.h
index 074981b..692f7ee 100644
--- a/api/test/fake_frame_encryptor.h
+++ b/api/test/fake_frame_encryptor.h
@@ -33,14 +33,14 @@
uint8_t postfix_byte = 255);
// Simply xors each payload with the provided fake key and adds the postfix
// bit to the end. This will always fail if fail_encryption_ is set to true.
- int Encrypt(cricket::MediaType media_type,
+ int Encrypt(webrtc::MediaType media_type,
uint32_t ssrc,
rtc::ArrayView<const uint8_t> additional_data,
rtc::ArrayView<const uint8_t> frame,
rtc::ArrayView<uint8_t> encrypted_frame,
size_t* bytes_written) override;
// Always returns 1 more than the size of the frame.
- size_t GetMaxCiphertextByteSize(cricket::MediaType media_type,
+ size_t GetMaxCiphertextByteSize(webrtc::MediaType media_type,
size_t frame_size) override;
// Sets the fake key to use during encryption.
void SetFakeKey(uint8_t fake_key);
diff --git a/api/test/mock_frame_decryptor.h b/api/test/mock_frame_decryptor.h
index 3f762ff..5e060c8 100644
--- a/api/test/mock_frame_decryptor.h
+++ b/api/test/mock_frame_decryptor.h
@@ -26,7 +26,7 @@
public:
MOCK_METHOD(Result,
Decrypt,
- (cricket::MediaType,
+ (webrtc::MediaType,
const std::vector<uint32_t>&,
rtc::ArrayView<const uint8_t>,
rtc::ArrayView<const uint8_t>,
@@ -35,7 +35,7 @@
MOCK_METHOD(size_t,
GetMaxPlaintextByteSize,
- (cricket::MediaType, size_t encrypted_frame_size),
+ (webrtc::MediaType, size_t encrypted_frame_size),
(override));
};
diff --git a/api/test/mock_frame_encryptor.h b/api/test/mock_frame_encryptor.h
index 5e99dff..ab0875b 100644
--- a/api/test/mock_frame_encryptor.h
+++ b/api/test/mock_frame_encryptor.h
@@ -25,7 +25,7 @@
public:
MOCK_METHOD(int,
Encrypt,
- (cricket::MediaType,
+ (webrtc::MediaType,
uint32_t,
rtc::ArrayView<const uint8_t>,
rtc::ArrayView<const uint8_t>,
@@ -35,7 +35,7 @@
MOCK_METHOD(size_t,
GetMaxCiphertextByteSize,
- (cricket::MediaType media_type, size_t frame_size),
+ (webrtc::MediaType media_type, size_t frame_size),
(override));
};
diff --git a/api/test/mock_peer_connection_factory_interface.h b/api/test/mock_peer_connection_factory_interface.h
index 2d46fe6..c5b1256 100644
--- a/api/test/mock_peer_connection_factory_interface.h
+++ b/api/test/mock_peer_connection_factory_interface.h
@@ -44,11 +44,11 @@
(override));
MOCK_METHOD(RtpCapabilities,
GetRtpSenderCapabilities,
- (cricket::MediaType),
+ (webrtc::MediaType),
(const, override));
MOCK_METHOD(RtpCapabilities,
GetRtpReceiverCapabilities,
- (cricket::MediaType),
+ (webrtc::MediaType),
(const, override));
MOCK_METHOD(rtc::scoped_refptr<MediaStreamInterface>,
CreateLocalMediaStream,
diff --git a/api/test/mock_peerconnectioninterface.h b/api/test/mock_peerconnectioninterface.h
index d0ea047..c1e9349 100644
--- a/api/test/mock_peerconnectioninterface.h
+++ b/api/test/mock_peerconnectioninterface.h
@@ -88,11 +88,11 @@
(override));
MOCK_METHOD(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
AddTransceiver,
- (cricket::MediaType),
+ (webrtc::MediaType),
(override));
MOCK_METHOD(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
AddTransceiver,
- (cricket::MediaType, const RtpTransceiverInit&),
+ (webrtc::MediaType, const RtpTransceiverInit&),
(override));
MOCK_METHOD(rtc::scoped_refptr<RtpSenderInterface>,
CreateSender,
diff --git a/api/test/mock_rtp_transceiver.h b/api/test/mock_rtp_transceiver.h
index 8bc2a6f..77bdf00 100644
--- a/api/test/mock_rtp_transceiver.h
+++ b/api/test/mock_rtp_transceiver.h
@@ -37,7 +37,7 @@
return rtc::make_ref_counted<MockRtpTransceiver>();
}
- MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
+ MOCK_METHOD(webrtc::MediaType, media_type, (), (const, override));
MOCK_METHOD(std::optional<std::string>, mid, (), (const, override));
MOCK_METHOD(rtc::scoped_refptr<RtpSenderInterface>,
sender,
diff --git a/api/test/mock_rtpreceiver.h b/api/test/mock_rtpreceiver.h
index 8f8c3b4..3e88d2e 100644
--- a/api/test/mock_rtpreceiver.h
+++ b/api/test/mock_rtpreceiver.h
@@ -37,7 +37,7 @@
streams,
(),
(const, override));
- MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
+ MOCK_METHOD(webrtc::MediaType, media_type, (), (const, override));
MOCK_METHOD(std::string, id, (), (const, override));
MOCK_METHOD(RtpParameters, GetParameters, (), (const, override));
MOCK_METHOD(bool,
diff --git a/api/test/mock_rtpsender.h b/api/test/mock_rtpsender.h
index ba1c7c0..144ce04 100644
--- a/api/test/mock_rtpsender.h
+++ b/api/test/mock_rtpsender.h
@@ -50,7 +50,7 @@
(),
(const, override));
MOCK_METHOD(uint32_t, ssrc, (), (const, override));
- MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
+ MOCK_METHOD(webrtc::MediaType, media_type, (), (const, override));
MOCK_METHOD(std::string, id, (), (const, override));
MOCK_METHOD(std::vector<std::string>, stream_ids, (), (const, override));
MOCK_METHOD(void, SetStreams, (const std::vector<std::string>&), (override));
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 4d7e47d..6d32265 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -716,14 +716,14 @@
rtc::Buffer decrypted_audio_payload;
if (frame_decryptor_ != nullptr) {
const size_t max_plaintext_size = frame_decryptor_->GetMaxPlaintextByteSize(
- cricket::MEDIA_TYPE_AUDIO, payload_length);
+ webrtc::MediaType::AUDIO, payload_length);
decrypted_audio_payload.SetSize(max_plaintext_size);
const std::vector<uint32_t> csrcs(header.arrOfCSRCs,
header.arrOfCSRCs + header.numCSRCs);
const FrameDecryptorInterface::Result decrypt_result =
frame_decryptor_->Decrypt(
- cricket::MEDIA_TYPE_AUDIO, csrcs,
+ webrtc::MediaType::AUDIO, csrcs,
/*additional_data=*/
nullptr,
rtc::ArrayView<const uint8_t>(payload, payload_data_length),
@@ -751,7 +751,7 @@
char buf[1024];
SimpleStringBuilder mime_type(buf);
auto it = payload_type_map_.find(header.payloadType);
- mime_type << cricket::MediaTypeToString(cricket::MEDIA_TYPE_AUDIO) << "/"
+ mime_type << webrtc::MediaTypeToString(webrtc::MediaType::AUDIO) << "/"
<< (it != payload_type_map_.end() ? it->second.name
: "x-unknown");
frame_transformer_delegate_->Transform(payload_data, header, remote_ssrc_,
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 994e843..1a487d9 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -385,7 +385,7 @@
// is transformed, the delegate will call SendRtpAudio to send it.
char buf[1024];
SimpleStringBuilder mime_type(buf);
- mime_type << cricket::MediaTypeToString(cricket::MEDIA_TYPE_AUDIO) << "/"
+ mime_type << webrtc::MediaTypeToString(webrtc::MediaType::AUDIO) << "/"
<< encoder_format_.name;
frame_transformer_delegate_->Transform(
frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
@@ -417,15 +417,15 @@
// TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
// Allocate a buffer to hold the maximum possible encrypted payload.
size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize(
- cricket::MEDIA_TYPE_AUDIO, payload.size());
+ webrtc::MediaType::AUDIO, payload.size());
encrypted_audio_payload.SetSize(max_ciphertext_size);
// Encrypt the audio payload into the buffer.
size_t bytes_written = 0;
- int encrypt_status = frame_encryptor_->Encrypt(
- cricket::MEDIA_TYPE_AUDIO, rtp_rtcp_->SSRC(),
- /*additional_data=*/nullptr, payload, encrypted_audio_payload,
- &bytes_written);
+ int encrypt_status =
+ frame_encryptor_->Encrypt(webrtc::MediaType::AUDIO, rtp_rtcp_->SSRC(),
+ /*additional_data=*/nullptr, payload,
+ encrypted_audio_payload, &bytes_written);
if (encrypt_status != 0) {
RTC_DLOG(LS_ERROR)
<< "Channel::SendData() failed encrypt audio payload: "
diff --git a/media/base/codec.cc b/media/base/codec.cc
index ac749e2..5eb8daa 100644
--- a/media/base/codec.cc
+++ b/media/base/codec.cc
@@ -203,11 +203,11 @@
switch (type) {
case Type::kAudio: {
codec_params.num_channels = static_cast<int>(channels);
- codec_params.kind = MEDIA_TYPE_AUDIO;
+ codec_params.kind = webrtc::MediaType::AUDIO;
break;
}
case Type::kVideo: {
- codec_params.kind = MEDIA_TYPE_VIDEO;
+ codec_params.kind = webrtc::MediaType::VIDEO;
break;
}
}
diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc
index 5420116..7cc55e4 100644
--- a/media/base/codec_comparators.cc
+++ b/media/base/codec_comparators.cc
@@ -403,14 +403,13 @@
// Some video codecs are compatible with others (e.g. same profile but
// different level). This comparison looks at the relevant parameters,
// ignoring ones that are either irrelevant or unrecognized.
- if (rtp_codec.kind == cricket::MediaType::MEDIA_TYPE_VIDEO &&
- rtp_codec.IsMediaCodec()) {
+ if (rtp_codec.kind == webrtc::MediaType::VIDEO && rtp_codec.IsMediaCodec()) {
return IsSameCodecSpecific(rtp_codec.name, params1, rtp_codec2.name,
params2);
}
// audio/RED should ignore the parameters which specify payload types so
// can not be compared.
- if (rtp_codec.kind == cricket::MediaType::MEDIA_TYPE_AUDIO &&
+ if (rtp_codec.kind == webrtc::MediaType::AUDIO &&
rtp_codec.name == cricket::kRedCodecName) {
return true;
}
diff --git a/media/base/codec_unittest.cc b/media/base/codec_unittest.cc
index 4cb78fa..cfe2552 100644
--- a/media/base/codec_unittest.cc
+++ b/media/base/codec_unittest.cc
@@ -259,7 +259,7 @@
v.SetParam("p1", "v1");
webrtc::RtpCodecParameters codec_params_1 = v.ToCodecParameters();
EXPECT_EQ(96, codec_params_1.payload_type);
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, codec_params_1.kind);
+ EXPECT_EQ(webrtc::MediaType::VIDEO, codec_params_1.kind);
EXPECT_EQ("V", codec_params_1.name);
EXPECT_EQ(cricket::kVideoCodecClockrate, codec_params_1.clock_rate);
EXPECT_EQ(std::nullopt, codec_params_1.num_channels);
@@ -271,7 +271,7 @@
a.SetParam("p1", "a1");
webrtc::RtpCodecParameters codec_params_2 = a.ToCodecParameters();
EXPECT_EQ(97, codec_params_2.payload_type);
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, codec_params_2.kind);
+ EXPECT_EQ(webrtc::MediaType::AUDIO, codec_params_2.kind);
EXPECT_EQ("A", codec_params_2.name);
EXPECT_EQ(44100, codec_params_2.clock_rate);
EXPECT_EQ(2, codec_params_2.num_channels);
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index 5fac7cf..ee5a777 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -503,8 +503,8 @@
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
return this;
}
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_AUDIO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::AUDIO;
}
bool SetReceiverParameters(const AudioReceiverParameters& params) override;
@@ -595,8 +595,8 @@
return nullptr;
}
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; }
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_AUDIO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::AUDIO;
}
bool SetSenderParameters(const AudioSenderParameter& params) override;
@@ -673,8 +673,8 @@
VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override {
return nullptr;
}
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_VIDEO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::VIDEO;
}
const std::vector<Codec>& recv_codecs() const;
@@ -747,8 +747,8 @@
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override {
return nullptr;
}
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_VIDEO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::VIDEO;
}
const std::vector<Codec>& send_codecs() const;
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index e1e8f54..e0cba40 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -191,7 +191,7 @@
virtual VideoMediaSendChannelInterface* AsVideoSendChannel() = 0;
virtual VoiceMediaSendChannelInterface* AsVoiceSendChannel() = 0;
- virtual cricket::MediaType media_type() const = 0;
+ virtual webrtc::MediaType media_type() const = 0;
// Gets the currently set codecs/payload types to be used for outgoing media.
virtual std::optional<Codec> GetSendCodec() const = 0;
@@ -267,7 +267,7 @@
virtual VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() = 0;
virtual VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() = 0;
- virtual cricket::MediaType media_type() const = 0;
+ virtual webrtc::MediaType media_type() const = 0;
// Creates a new incoming media stream with SSRCs, CNAME as described
// by sp. In the case of a sp without SSRCs, the unsignaled sp is cached
// to be used later for unsignaled streams received.
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 4844756..aab602a 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -170,7 +170,7 @@
webrtc::VideoBitrateAllocatorFactory* bitrate_allocator_factory);
~WebRtcVideoSendChannel() override;
- MediaType media_type() const override { return MEDIA_TYPE_VIDEO; }
+ MediaType media_type() const override { return webrtc::MediaType::VIDEO; }
// Type manipulations
VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; }
VoiceMediaSendChannelInterface* AsVoiceSendChannel() override {
@@ -574,7 +574,7 @@
~WebRtcVideoReceiveChannel() override;
public:
- MediaType media_type() const override { return MEDIA_TYPE_VIDEO; }
+ MediaType media_type() const override { return webrtc::MediaType::VIDEO; }
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
return this;
}
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 2429599..a7a7340 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -5108,7 +5108,7 @@
webrtc::RtpCodec vp9_rtp_codec;
vp9_rtp_codec.name = "VP9";
- vp9_rtp_codec.kind = cricket::MEDIA_TYPE_VIDEO;
+ vp9_rtp_codec.kind = webrtc::MediaType::VIDEO;
vp9_rtp_codec.clock_rate = 90000;
initial_params.encodings[0].codec = vp9_rtp_codec;
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 1f7f361..0f1be3e 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -200,7 +200,7 @@
~WebRtcVoiceSendChannel() override;
- MediaType media_type() const override { return MEDIA_TYPE_AUDIO; }
+ MediaType media_type() const override { return webrtc::MediaType::AUDIO; }
VideoMediaSendChannelInterface* AsVideoSendChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
@@ -363,7 +363,7 @@
~WebRtcVoiceReceiveChannel() override;
- MediaType media_type() const override { return MEDIA_TYPE_AUDIO; }
+ MediaType media_type() const override { return webrtc::MediaType::AUDIO; }
VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override {
RTC_CHECK_NOTREACHED();
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 74e3d1f..1db9c22 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -1449,7 +1449,7 @@
webrtc::RtpCodec opus_rtp_codec;
opus_rtp_codec.name = "opus";
- opus_rtp_codec.kind = cricket::MEDIA_TYPE_AUDIO;
+ opus_rtp_codec.kind = webrtc::MediaType::AUDIO;
opus_rtp_codec.num_channels = 2;
opus_rtp_codec.clock_rate = 48000;
initial_params.encodings[0].codec = opus_rtp_codec;
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 6d6d804..c3eeced 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -669,7 +669,7 @@
rtc::Buffer encrypted_video_payload;
if (frame_encryptor_ != nullptr) {
const size_t max_ciphertext_size =
- frame_encryptor_->GetMaxCiphertextByteSize(cricket::MEDIA_TYPE_VIDEO,
+ frame_encryptor_->GetMaxCiphertextByteSize(webrtc::MediaType::VIDEO,
payload.size());
encrypted_video_payload.SetSize(max_ciphertext_size);
@@ -682,7 +682,7 @@
}
if (frame_encryptor_->Encrypt(
- cricket::MEDIA_TYPE_VIDEO, first_packet->Ssrc(), additional_data,
+ webrtc::MediaType::VIDEO, first_packet->Ssrc(), additional_data,
payload, encrypted_video_payload, &bytes_written) != 0) {
return false;
}
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index c18cb39..24bf325 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -461,6 +461,7 @@
"../media:codec_list",
"../media:media_constants",
"../media:media_engine",
+ "../rtc_base:logging",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
@@ -2630,6 +2631,7 @@
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
+ "//third_party/jsoncpp",
]
# These deps are kept separately because they can't be automatically
diff --git a/pc/DEPS b/pc/DEPS
index 959885f..dd44ac1 100644
--- a/pc/DEPS
+++ b/pc/DEPS
@@ -16,3 +16,10 @@
"+system_wrappers",
"+absl/strings/str_cat.h",
]
+
+specific_include_rules = {
+ "rtc_stats_collector_unittest.cc": [
+ "+json/reader.h",
+ "+json/value.h",
+ ],
+}
\ No newline at end of file
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index e5942e7..d146bc8 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -82,8 +82,8 @@
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override;
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_AUDIO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::AUDIO;
}
std::string id() const override { return id_; }
diff --git a/pc/channel.cc b/pc/channel.cc
index 012143d..889a894 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -100,8 +100,8 @@
const RtpHeaderExtensions& extensions,
bool is_stream_active,
MediaChannelParameters* params) {
- RTC_DCHECK(desc->type() == MEDIA_TYPE_AUDIO ||
- desc->type() == MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(desc->type() == webrtc::MediaType::AUDIO ||
+ desc->type() == webrtc::MediaType::VIDEO);
params->is_stream_active = is_stream_active;
params->codecs = desc->codecs();
// TODO(bugs.webrtc.org/11513): See if we really need
diff --git a/pc/channel.h b/pc/channel.h
index 3a3680f..7840693 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -412,8 +412,8 @@
return receive_channel();
}
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_AUDIO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::AUDIO;
}
private:
@@ -481,8 +481,8 @@
return receive_channel();
}
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_VIDEO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::VIDEO;
}
private:
diff --git a/pc/channel_interface.h b/pc/channel_interface.h
index 0500b87..7b8011d 100644
--- a/pc/channel_interface.h
+++ b/pc/channel_interface.h
@@ -49,7 +49,7 @@
class ChannelInterface {
public:
virtual ~ChannelInterface() = default;
- virtual cricket::MediaType media_type() const = 0;
+ virtual webrtc::MediaType media_type() const = 0;
virtual VideoChannel* AsVideoChannel() = 0;
virtual VoiceChannel* AsVoiceChannel() = 0;
diff --git a/pc/codec_vendor.cc b/pc/codec_vendor.cc
index 6db82b5..34e6edc 100644
--- a/pc/codec_vendor.cc
+++ b/pc/codec_vendor.cc
@@ -345,9 +345,9 @@
if (!checked_codec_list.ok()) {
RTC_LOG(LS_ERROR) << checked_codec_list.error();
}
- if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
+ if (IsMediaContentOfType(content, webrtc::MediaType::AUDIO)) {
MergeCodecs(checked_codec_list.value(), audio_codecs, used_pltypes);
- } else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
+ } else if (IsMediaContentOfType(content, webrtc::MediaType::VIDEO)) {
MergeCodecs(checked_codec_list.value(), video_codecs, used_pltypes);
}
}
@@ -387,7 +387,7 @@
// Ideally this should be done for all codecs, but RFCs of other codecs
// do not clear define the expected behavior for the level in the offer.
#ifdef RTC_ENABLE_H265
- if (media_description_options.type == MEDIA_TYPE_VIDEO) {
+ if (media_description_options.type == webrtc::MediaType::VIDEO) {
std::unordered_map<webrtc::H265Profile, webrtc::H265Level>
supported_h265_profiles;
// The assumption here is that H.265 codecs with the same profile and tier
@@ -577,7 +577,7 @@
const CodecList& codecs) {
CodecList filtered_codecs;
CodecList supported_codecs =
- media_description_options.type == MEDIA_TYPE_AUDIO
+ media_description_options.type == webrtc::MediaType::AUDIO
? GetAudioCodecsForOffer(media_description_options.direction)
: GetVideoCodecsForOffer(media_description_options.direction);
@@ -626,7 +626,7 @@
!FindMatchingCodec(supported_codecs, filtered_codecs, codec)) {
// Use the `found_codec` from `codecs` because it has the
// correctly mapped payload type (most of the time).
- if (media_description_options.type == MEDIA_TYPE_VIDEO &&
+ if (media_description_options.type == webrtc::MediaType::VIDEO &&
found_codec->GetResiliencyType() == Codec::ResiliencyType::kRtx) {
// For RTX we might need to adjust the apt parameter if we got a
// remote offer without RTX for a codec for which we support RTX.
@@ -651,11 +651,11 @@
}
}
- if (media_description_options.type == MEDIA_TYPE_AUDIO &&
+ if (media_description_options.type == webrtc::MediaType::AUDIO &&
!session_options.vad_enabled) {
// If application doesn't want CN codecs in offer.
StripCNCodecs(filtered_codecs);
- } else if (media_description_options.type == MEDIA_TYPE_VIDEO &&
+ } else if (media_description_options.type == webrtc::MediaType::VIDEO &&
session_options.raw_packetization_for_video) {
for (Codec& codec : filtered_codecs) {
if (codec.IsMediaCodec()) {
@@ -693,7 +693,7 @@
CodecList negotiated_codecs;
if (media_description_options.codecs_to_include.empty()) {
const CodecList& supported_codecs =
- media_description_options.type == MEDIA_TYPE_AUDIO
+ media_description_options.type == webrtc::MediaType::AUDIO
? GetAudioCodecsForAnswer(offer_rtd, answer_rtd)
: GetVideoCodecsForAnswer(offer_rtd, answer_rtd);
if (!media_description_options.codec_preferences.empty()) {
@@ -738,11 +738,11 @@
filtered_codecs = ComputeCodecsUnion(filtered_codecs, other_codecs);
}
- if (media_description_options.type == MEDIA_TYPE_AUDIO &&
+ if (media_description_options.type == webrtc::MediaType::AUDIO &&
!session_options.vad_enabled) {
// If application doesn't want CN codecs in offer.
StripCNCodecs(filtered_codecs);
- } else if (media_description_options.type == MEDIA_TYPE_VIDEO &&
+ } else if (media_description_options.type == webrtc::MediaType::VIDEO &&
session_options.raw_packetization_for_video) {
for (Codec& codec : filtered_codecs) {
if (codec.IsMediaCodec()) {
@@ -780,16 +780,16 @@
// the codecs are explicitly set by the test.
if (media_engine) {
audio_send_codecs_ =
- TypedCodecVendor(media_engine, MEDIA_TYPE_AUDIO,
+ TypedCodecVendor(media_engine, webrtc::MediaType::AUDIO,
/* is_sender= */ true, rtx_enabled, trials);
audio_recv_codecs_ =
- TypedCodecVendor(media_engine, MEDIA_TYPE_AUDIO,
+ TypedCodecVendor(media_engine, webrtc::MediaType::AUDIO,
/* is_sender= */ false, rtx_enabled, trials);
video_send_codecs_ =
- TypedCodecVendor(media_engine, MEDIA_TYPE_VIDEO,
+ TypedCodecVendor(media_engine, webrtc::MediaType::VIDEO,
/* is_sender= */ true, rtx_enabled, trials);
video_recv_codecs_ =
- TypedCodecVendor(media_engine, MEDIA_TYPE_VIDEO,
+ TypedCodecVendor(media_engine, webrtc::MediaType::VIDEO,
/* is_sender= */ false, rtx_enabled, trials);
}
}
@@ -876,7 +876,7 @@
if (!offered_codecs.ok()) {
return offered_codecs.MoveError();
}
- if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) {
+ if (IsMediaContentOfType(&content, webrtc::MediaType::AUDIO)) {
for (const Codec& offered_audio_codec : offered_codecs.value()) {
if (!FindMatchingCodec(offered_codecs.value(),
filtered_offered_audio_codecs,
@@ -886,7 +886,7 @@
filtered_offered_audio_codecs.push_back(offered_audio_codec);
}
}
- } else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) {
+ } else if (IsMediaContentOfType(&content, webrtc::MediaType::VIDEO)) {
std::vector<Codec> pending_rtx_codecs;
for (const Codec& offered_video_codec : offered_codecs.value()) {
if (!FindMatchingCodec(offered_codecs.value(),
diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc
index 300a4e1..398d620 100644
--- a/pc/legacy_stats_collector.cc
+++ b/pc/legacy_stats_collector.cc
@@ -15,9 +15,11 @@
#include <algorithm>
#include <cmath>
-#include <list>
+#include <map>
+#include <memory>
#include <optional>
#include <set>
+#include <string>
#include <utility>
#include <vector>
@@ -27,19 +29,24 @@
#include "api/candidate.h"
#include "api/data_channel_interface.h"
#include "api/field_trials_view.h"
+#include "api/legacy_stats_types.h"
+#include "api/media_stream_interface.h"
#include "api/media_types.h"
+#include "api/peer_connection_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/video/video_content_type.h"
-#include "api/video/video_timing.h"
#include "call/call.h"
#include "media/base/media_channel.h"
+#include "p2p/base/connection_info.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
+#include "p2p/base/port.h"
#include "pc/channel.h"
#include "pc/channel_interface.h"
#include "pc/data_channel_utils.h"
+#include "pc/peer_connection_internal.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_receiver_proxy.h"
#include "pc/rtp_sender_proxy.h"
@@ -48,8 +55,10 @@
#include "rtc_base/checks.h"
#include "rtc_base/ip_address.h"
#include "rtc_base/logging.h"
+#include "rtc_base/network_constants.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/socket_address.h"
+#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/thread.h"
@@ -1060,7 +1069,7 @@
auto transceivers = pc_->GetTransceiversInternal();
std::vector<cricket::VideoMediaSendChannelInterface*> video_media_channels;
for (const auto& transceiver : transceivers) {
- if (transceiver->media_type() != cricket::MEDIA_TYPE_VIDEO) {
+ if (transceiver->media_type() != webrtc::MediaType::VIDEO) {
continue;
}
auto* video_channel = transceiver->internal()->channel();
@@ -1193,11 +1202,11 @@
std::unique_ptr<ChannelStatsGatherer> CreateChannelStatsGatherer(
cricket::ChannelInterface* channel) {
RTC_DCHECK(channel);
- if (channel->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (channel->media_type() == webrtc::MediaType::AUDIO) {
return std::make_unique<VoiceChannelStatsGatherer>(
channel->AsVoiceChannel());
} else {
- RTC_DCHECK_EQ(channel->media_type(), cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK_EQ(channel->media_type(), webrtc::MediaType::VIDEO);
return std::make_unique<VideoChannelStatsGatherer>(
channel->AsVideoChannel());
}
diff --git a/pc/legacy_stats_collector_unittest.cc b/pc/legacy_stats_collector_unittest.cc
index 268eabf..9ab8511 100644
--- a/pc/legacy_stats_collector_unittest.cc
+++ b/pc/legacy_stats_collector_unittest.cc
@@ -13,23 +13,31 @@
#include <stdio.h>
#include <cstdint>
+#include <memory>
#include <optional>
+#include <string>
+#include <utility>
+#include <vector>
#include "absl/algorithm/container.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/candidate.h"
#include "api/data_channel_interface.h"
+#include "api/legacy_stats_types.h"
+#include "api/make_ref_counted.h"
+#include "api/media_stream_interface.h"
#include "api/media_stream_track.h"
#include "api/media_types.h"
+#include "api/peer_connection_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "call/call.h"
#include "media/base/media_channel.h"
+#include "p2p/base/connection_info.h"
#include "p2p/base/ice_transport_internal.h"
#include "pc/media_stream.h"
-#include "pc/rtp_receiver.h"
-#include "pc/rtp_sender.h"
+#include "pc/peer_connection_internal.h"
#include "pc/sctp_data_channel.h"
#include "pc/test/fake_peer_connection_for_stats.h"
#include "pc/test/fake_video_track_source.h"
@@ -37,9 +45,11 @@
#include "pc/test/mock_rtp_sender_internal.h"
#include "pc/transport_stats.h"
#include "pc/video_track.h"
+#include "rtc_base/checks.h"
#include "rtc_base/fake_ssl_identity.h"
#include "rtc_base/message_digest.h"
#include "rtc_base/net_helper.h"
+#include "rtc_base/network_constants.h"
#include "rtc_base/null_socket_server.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/socket_address.h"
@@ -736,8 +746,8 @@
EXPECT_CALL(*sender, media_type())
.WillRepeatedly(
Return(track->kind() == MediaStreamTrackInterface::kAudioKind
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO));
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO));
EXPECT_CALL(*sender, SetMediaChannel(_)).Times(AtMost(2));
EXPECT_CALL(*sender, SetTransceiverAsStopped()).Times(AtMost(1));
EXPECT_CALL(*sender, Stop());
@@ -753,8 +763,8 @@
EXPECT_CALL(*receiver, media_type())
.WillRepeatedly(
Return(track->kind() == MediaStreamTrackInterface::kAudioKind
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO));
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO));
EXPECT_CALL(*receiver, SetMediaChannel(_)).WillRepeatedly(Return());
EXPECT_CALL(*receiver, Stop()).WillRepeatedly(Return());
return receiver;
diff --git a/pc/media_options.cc b/pc/media_options.cc
index 5ad8795..6326ebb 100644
--- a/pc/media_options.cc
+++ b/pc/media_options.cc
@@ -38,7 +38,7 @@
void MediaDescriptionOptions::AddAudioSender(
const std::string& track_id,
const std::vector<std::string>& stream_ids) {
- RTC_DCHECK(type == MEDIA_TYPE_AUDIO);
+ RTC_DCHECK(type == webrtc::MediaType::AUDIO);
AddSenderInternal(track_id, stream_ids, {}, SimulcastLayerList(), 1);
}
@@ -48,7 +48,7 @@
const std::vector<RidDescription>& rids,
const SimulcastLayerList& simulcast_layers,
int num_sim_layers) {
- RTC_DCHECK(type == MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(type == webrtc::MediaType::VIDEO);
RTC_DCHECK(rids.empty() || num_sim_layers == 0)
<< "RIDs are the compliant way to indicate simulcast.";
RTC_DCHECK(ValidateSimulcastLayers(rids, simulcast_layers));
diff --git a/pc/media_options.h b/pc/media_options.h
index c70e956..0e268e0 100644
--- a/pc/media_options.h
+++ b/pc/media_options.h
@@ -90,9 +90,13 @@
struct MediaSessionOptions {
MediaSessionOptions() {}
- bool has_audio() const { return HasMediaDescription(MEDIA_TYPE_AUDIO); }
- bool has_video() const { return HasMediaDescription(MEDIA_TYPE_VIDEO); }
- bool has_data() const { return HasMediaDescription(MEDIA_TYPE_DATA); }
+ bool has_audio() const {
+ return HasMediaDescription(webrtc::MediaType::AUDIO);
+ }
+ bool has_video() const {
+ return HasMediaDescription(webrtc::MediaType::VIDEO);
+ }
+ bool has_data() const { return HasMediaDescription(webrtc::MediaType::DATA); }
bool HasMediaDescription(MediaType type) const;
diff --git a/pc/media_session.cc b/pc/media_session.cc
index ab41adc..90110aa 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -145,7 +145,7 @@
}
bool IsMediaContentOfType(const ContentInfo* content,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
if (!content || !content->media_description()) {
return false;
}
@@ -249,8 +249,8 @@
RTC_DCHECK_EQ(1, description->streams().size())
<< "RIDs are only supported in Unified Plan semantics.";
RTC_DCHECK_EQ(1, media_description_options.sender_options.size());
- RTC_DCHECK(description->type() == cricket::MediaType::MEDIA_TYPE_AUDIO ||
- description->type() == cricket::MediaType::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(description->type() == webrtc::MediaType::AUDIO ||
+ description->type() == webrtc::MediaType::VIDEO);
// One RID or less indicates that simulcast is not needed.
if (description->streams()[0].rids().size() <= 1) {
@@ -556,8 +556,8 @@
cricket::StreamParamsVec* current_streams,
MediaContentDescription* answer,
const FieldTrialsView& field_trials) {
- RTC_DCHECK(offer->type() == cricket::MEDIA_TYPE_AUDIO ||
- offer->type() == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(offer->type() == webrtc::MediaType::AUDIO ||
+ offer->type() == webrtc::MediaType::VIDEO);
answer->AddCodecs(local_codecs);
answer->set_protocol(offer->protocol());
if (!AddStreamParams(media_description_options.sender_options,
@@ -625,7 +625,7 @@
return true;
}
-bool IsMediaProtocolSupported(cricket::MediaType type,
+bool IsMediaProtocolSupported(webrtc::MediaType type,
const std::string& protocol,
bool secure_transport) {
// Since not all applications serialize and deserialize the media protocol,
@@ -634,7 +634,7 @@
return true;
}
- if (type == cricket::MEDIA_TYPE_DATA) {
+ if (type == webrtc::MediaType::DATA) {
// Check for SCTP
if (secure_transport) {
// Most likely scenarios first.
@@ -763,26 +763,26 @@
// Media type must match unless this media section is being recycled.
}
switch (media_description_options.type) {
- case cricket::MEDIA_TYPE_AUDIO:
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::AUDIO:
+ case webrtc::MediaType::VIDEO:
error = AddRtpContentForOffer(
media_description_options, session_options, current_content,
current_description,
- media_description_options.type == cricket::MEDIA_TYPE_AUDIO
+ media_description_options.type == webrtc::MediaType::AUDIO
? extensions_with_ids.audio
: extensions_with_ids.video,
- media_description_options.type == cricket::MEDIA_TYPE_AUDIO
+ media_description_options.type == webrtc::MediaType::AUDIO
? offer_audio_codecs
: offer_video_codecs,
¤t_streams, offer.get(), &ice_credentials);
break;
- case cricket::MEDIA_TYPE_DATA:
+ case webrtc::MediaType::DATA:
error = AddDataContentForOffer(media_description_options,
session_options, current_content,
current_description, ¤t_streams,
offer.get(), &ice_credentials);
break;
- case cricket::MEDIA_TYPE_UNSUPPORTED:
+ case webrtc::MediaType::UNSUPPORTED:
error = AddUnsupportedContentForOffer(
media_description_options, session_options, current_content,
current_description, offer.get(), &ice_credentials);
@@ -959,24 +959,24 @@
RtpHeaderExtensionsFromCapabilities(
UnstoppedRtpHeaderExtensionCapabilities(header_extensions_in));
switch (media_description_options.type) {
- case cricket::MEDIA_TYPE_AUDIO:
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::AUDIO:
+ case webrtc::MediaType::VIDEO:
error = AddRtpContentForAnswer(
media_description_options, session_options, offer_content, offer,
current_content, current_description, bundle_transport,
- media_description_options.type == cricket::MEDIA_TYPE_AUDIO
+ media_description_options.type == webrtc::MediaType::AUDIO
? answer_audio_codecs
: answer_video_codecs,
header_extensions, ¤t_streams, answer.get(),
&ice_credentials);
break;
- case cricket::MEDIA_TYPE_DATA:
+ case webrtc::MediaType::DATA:
error = AddDataContentForAnswer(
media_description_options, session_options, offer_content, offer,
current_content, current_description, bundle_transport,
¤t_streams, answer.get(), &ice_credentials);
break;
- case cricket::MEDIA_TYPE_UNSUPPORTED:
+ case webrtc::MediaType::UNSUPPORTED:
error = AddUnsupportedContentForAnswer(
media_description_options, session_options, offer_content, offer,
current_content, current_description, bundle_transport,
@@ -1094,12 +1094,12 @@
// Add them to `used_ids` so the local ids are not reused if a new media
// type is added.
for (const cricket::ContentInfo* content : current_active_contents) {
- if (IsMediaContentOfType(content, cricket::MEDIA_TYPE_AUDIO)) {
+ if (IsMediaContentOfType(content, webrtc::MediaType::AUDIO)) {
MergeRtpHdrExts(content->media_description()->rtp_header_extensions(),
enable_encrypted_rtp_header_extensions_,
&offered_extensions.audio, &all_encountered_extensions,
&used_ids);
- } else if (IsMediaContentOfType(content, cricket::MEDIA_TYPE_VIDEO)) {
+ } else if (IsMediaContentOfType(content, webrtc::MediaType::VIDEO)) {
MergeRtpHdrExts(content->media_description()->rtp_header_extensions(),
enable_encrypted_rtp_header_extensions_,
&offered_extensions.video, &all_encountered_extensions,
@@ -1114,11 +1114,11 @@
cricket::RtpHeaderExtensions filtered_extensions =
filtered_rtp_header_extensions(UnstoppedOrPresentRtpHeaderExtensions(
entry.header_extensions, all_encountered_extensions));
- if (entry.type == cricket::MEDIA_TYPE_AUDIO)
+ if (entry.type == webrtc::MediaType::AUDIO)
MergeRtpHdrExts(
filtered_extensions, enable_encrypted_rtp_header_extensions_,
&offered_extensions.audio, &all_encountered_extensions, &used_ids);
- else if (entry.type == cricket::MEDIA_TYPE_VIDEO)
+ else if (entry.type == webrtc::MediaType::VIDEO)
MergeRtpHdrExts(
filtered_extensions, enable_encrypted_rtp_header_extensions_,
&offered_extensions.video, &all_encountered_extensions, &used_ids);
@@ -1199,8 +1199,8 @@
cricket::StreamParamsVec* current_streams,
SessionDescription* session_description,
cricket::IceCredentialsIterator* ice_credentials) const {
- RTC_DCHECK(media_description_options.type == cricket::MEDIA_TYPE_AUDIO ||
- media_description_options.type == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(media_description_options.type == webrtc::MediaType::AUDIO ||
+ media_description_options.type == webrtc::MediaType::VIDEO);
std::vector<cricket::Codec> codecs_to_include;
std::string mid = media_description_options.mid;
@@ -1213,7 +1213,7 @@
}
codecs_to_include = error_or_filtered_codecs.MoveValue();
std::unique_ptr<MediaContentDescription> content_description;
- if (media_description_options.type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_description_options.type == webrtc::MediaType::AUDIO) {
content_description = std::make_unique<AudioContentDescription>();
} else {
content_description = std::make_unique<VideoContentDescription>();
@@ -1292,7 +1292,7 @@
SessionDescription* desc,
cricket::IceCredentialsIterator* ice_credentials) const {
RTC_CHECK(
- IsMediaContentOfType(current_content, cricket::MEDIA_TYPE_UNSUPPORTED));
+ IsMediaContentOfType(current_content, webrtc::MediaType::UNSUPPORTED));
const UnsupportedContentDescription* current_unsupported_description =
current_content->media_description()->as_unsupported();
@@ -1332,12 +1332,12 @@
cricket::StreamParamsVec* current_streams,
SessionDescription* answer,
cricket::IceCredentialsIterator* ice_credentials) const {
- RTC_DCHECK(media_description_options.type == cricket::MEDIA_TYPE_AUDIO ||
- media_description_options.type == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(media_description_options.type == webrtc::MediaType::AUDIO ||
+ media_description_options.type == webrtc::MediaType::VIDEO);
RTC_CHECK(
IsMediaContentOfType(offer_content, media_description_options.type));
const RtpMediaContentDescription* offer_content_description;
- if (media_description_options.type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_description_options.type == webrtc::MediaType::AUDIO) {
offer_content_description = offer_content->media_description()->as_audio();
} else {
offer_content_description = offer_content->media_description()->as_video();
@@ -1387,7 +1387,7 @@
offer_description->HasGroup(cricket::GROUP_TYPE_BUNDLE) &&
session_options.bundle_enabled;
std::unique_ptr<MediaContentDescription> answer_content;
- if (media_description_options.type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_description_options.type == webrtc::MediaType::AUDIO) {
answer_content = std::make_unique<AudioContentDescription>();
} else {
answer_content = std::make_unique<VideoContentDescription>();
@@ -1424,7 +1424,7 @@
: transport->secure();
bool rejected = media_description_options.stopped ||
offer_content->rejected || !has_usable_media_codecs ||
- !IsMediaProtocolSupported(cricket::MEDIA_TYPE_AUDIO,
+ !IsMediaProtocolSupported(webrtc::MediaType::AUDIO,
answer_content->protocol(), secure);
if (rejected) {
RTC_LOG(LS_INFO) << "m= section '" << media_description_options.mid
@@ -1468,7 +1468,7 @@
bool bundle_enabled =
offer_description->HasGroup(cricket::GROUP_TYPE_BUNDLE) &&
session_options.bundle_enabled;
- RTC_CHECK(IsMediaContentOfType(offer_content, cricket::MEDIA_TYPE_DATA));
+ RTC_CHECK(IsMediaContentOfType(offer_content, webrtc::MediaType::DATA));
std::unique_ptr<MediaContentDescription> data_answer;
if (offer_content->media_description()->as_sctp()) {
// SCTP data content
@@ -1509,7 +1509,7 @@
bool rejected = media_description_options.stopped ||
offer_content->rejected ||
- !IsMediaProtocolSupported(cricket::MEDIA_TYPE_DATA,
+ !IsMediaProtocolSupported(webrtc::MediaType::DATA,
data_answer->protocol(), secure);
auto error = AddTransportAnswer(media_description_options.mid,
*data_transport, answer);
@@ -1543,7 +1543,7 @@
"Failed to create transport answer, unsupported transport is missing");
}
RTC_CHECK(
- IsMediaContentOfType(offer_content, cricket::MEDIA_TYPE_UNSUPPORTED));
+ IsMediaContentOfType(offer_content, webrtc::MediaType::UNSUPPORTED));
const UnsupportedContentDescription* offer_unsupported_description =
offer_content->media_description()->as_unsupported();
@@ -1569,23 +1569,23 @@
}
bool IsAudioContent(const ContentInfo* content) {
- return IsMediaContentOfType(content, cricket::MEDIA_TYPE_AUDIO);
+ return IsMediaContentOfType(content, webrtc::MediaType::AUDIO);
}
bool IsVideoContent(const ContentInfo* content) {
- return IsMediaContentOfType(content, cricket::MEDIA_TYPE_VIDEO);
+ return IsMediaContentOfType(content, webrtc::MediaType::VIDEO);
}
bool IsDataContent(const ContentInfo* content) {
- return IsMediaContentOfType(content, cricket::MEDIA_TYPE_DATA);
+ return IsMediaContentOfType(content, webrtc::MediaType::DATA);
}
bool IsUnsupportedContent(const ContentInfo* content) {
- return IsMediaContentOfType(content, cricket::MEDIA_TYPE_UNSUPPORTED);
+ return IsMediaContentOfType(content, webrtc::MediaType::UNSUPPORTED);
}
const ContentInfo* GetFirstMediaContent(const cricket::ContentInfos& contents,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
for (const cricket::ContentInfo& content : contents) {
if (IsMediaContentOfType(&content, media_type)) {
return &content;
@@ -1595,19 +1595,19 @@
}
const ContentInfo* GetFirstAudioContent(const cricket::ContentInfos& contents) {
- return GetFirstMediaContent(contents, cricket::MEDIA_TYPE_AUDIO);
+ return GetFirstMediaContent(contents, webrtc::MediaType::AUDIO);
}
const ContentInfo* GetFirstVideoContent(const cricket::ContentInfos& contents) {
- return GetFirstMediaContent(contents, cricket::MEDIA_TYPE_VIDEO);
+ return GetFirstMediaContent(contents, webrtc::MediaType::VIDEO);
}
const ContentInfo* GetFirstDataContent(const cricket::ContentInfos& contents) {
- return GetFirstMediaContent(contents, cricket::MEDIA_TYPE_DATA);
+ return GetFirstMediaContent(contents, webrtc::MediaType::DATA);
}
const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
if (sdesc == nullptr) {
return nullptr;
}
@@ -1616,39 +1616,39 @@
}
const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) {
- return GetFirstMediaContent(sdesc, cricket::MEDIA_TYPE_AUDIO);
+ return GetFirstMediaContent(sdesc, webrtc::MediaType::AUDIO);
}
const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) {
- return GetFirstMediaContent(sdesc, cricket::MEDIA_TYPE_VIDEO);
+ return GetFirstMediaContent(sdesc, webrtc::MediaType::VIDEO);
}
const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) {
- return GetFirstMediaContent(sdesc, cricket::MEDIA_TYPE_DATA);
+ return GetFirstMediaContent(sdesc, webrtc::MediaType::DATA);
}
const MediaContentDescription* GetFirstMediaContentDescription(
const SessionDescription* sdesc,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
const ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
return (content ? content->media_description() : nullptr);
}
const AudioContentDescription* GetFirstAudioContentDescription(
const SessionDescription* sdesc) {
- auto desc = GetFirstMediaContentDescription(sdesc, cricket::MEDIA_TYPE_AUDIO);
+ auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::AUDIO);
return desc ? desc->as_audio() : nullptr;
}
const VideoContentDescription* GetFirstVideoContentDescription(
const SessionDescription* sdesc) {
- auto desc = GetFirstMediaContentDescription(sdesc, cricket::MEDIA_TYPE_VIDEO);
+ auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::VIDEO);
return desc ? desc->as_video() : nullptr;
}
const SctpDataContentDescription* GetFirstSctpDataContentDescription(
const SessionDescription* sdesc) {
- auto desc = GetFirstMediaContentDescription(sdesc, cricket::MEDIA_TYPE_DATA);
+ auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::DATA);
return desc ? desc->as_sctp() : nullptr;
}
@@ -1657,7 +1657,7 @@
//
ContentInfo* GetFirstMediaContent(cricket::ContentInfos* contents,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
for (cricket::ContentInfo& content : *contents) {
if (IsMediaContentOfType(&content, media_type)) {
return &content;
@@ -1667,19 +1667,19 @@
}
ContentInfo* GetFirstAudioContent(cricket::ContentInfos* contents) {
- return GetFirstMediaContent(contents, cricket::MEDIA_TYPE_AUDIO);
+ return GetFirstMediaContent(contents, webrtc::MediaType::AUDIO);
}
ContentInfo* GetFirstVideoContent(cricket::ContentInfos* contents) {
- return GetFirstMediaContent(contents, cricket::MEDIA_TYPE_VIDEO);
+ return GetFirstMediaContent(contents, webrtc::MediaType::VIDEO);
}
ContentInfo* GetFirstDataContent(cricket::ContentInfos* contents) {
- return GetFirstMediaContent(contents, cricket::MEDIA_TYPE_DATA);
+ return GetFirstMediaContent(contents, webrtc::MediaType::DATA);
}
ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
if (sdesc == nullptr) {
return nullptr;
}
@@ -1688,39 +1688,39 @@
}
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) {
- return GetFirstMediaContent(sdesc, cricket::MEDIA_TYPE_AUDIO);
+ return GetFirstMediaContent(sdesc, webrtc::MediaType::AUDIO);
}
ContentInfo* GetFirstVideoContent(SessionDescription* sdesc) {
- return GetFirstMediaContent(sdesc, cricket::MEDIA_TYPE_VIDEO);
+ return GetFirstMediaContent(sdesc, webrtc::MediaType::VIDEO);
}
ContentInfo* GetFirstDataContent(SessionDescription* sdesc) {
- return GetFirstMediaContent(sdesc, cricket::MEDIA_TYPE_DATA);
+ return GetFirstMediaContent(sdesc, webrtc::MediaType::DATA);
}
MediaContentDescription* GetFirstMediaContentDescription(
SessionDescription* sdesc,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
return (content ? content->media_description() : nullptr);
}
AudioContentDescription* GetFirstAudioContentDescription(
SessionDescription* sdesc) {
- auto desc = GetFirstMediaContentDescription(sdesc, cricket::MEDIA_TYPE_AUDIO);
+ auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::AUDIO);
return desc ? desc->as_audio() : nullptr;
}
VideoContentDescription* GetFirstVideoContentDescription(
SessionDescription* sdesc) {
- auto desc = GetFirstMediaContentDescription(sdesc, cricket::MEDIA_TYPE_VIDEO);
+ auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::VIDEO);
return desc ? desc->as_video() : nullptr;
}
SctpDataContentDescription* GetFirstSctpDataContentDescription(
SessionDescription* sdesc) {
- auto desc = GetFirstMediaContentDescription(sdesc, cricket::MEDIA_TYPE_DATA);
+ auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::DATA);
return desc ? desc->as_sctp() : nullptr;
}
diff --git a/pc/media_session.h b/pc/media_session.h
index ce2e19f..fcd4aea 100644
--- a/pc/media_session.h
+++ b/pc/media_session.h
@@ -197,12 +197,12 @@
bool IsDataContent(const ContentInfo* content);
bool IsUnsupportedContent(const ContentInfo* content);
const ContentInfo* GetFirstMediaContent(const cricket::ContentInfos& contents,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
const ContentInfo* GetFirstAudioContent(const cricket::ContentInfos& contents);
const ContentInfo* GetFirstVideoContent(const cricket::ContentInfos& contents);
const ContentInfo* GetFirstDataContent(const cricket::ContentInfos& contents);
const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc);
const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc);
const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc);
@@ -215,12 +215,12 @@
// Non-const versions of the above functions.
// Useful when modifying an existing description.
ContentInfo* GetFirstMediaContent(cricket::ContentInfos* contents,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
ContentInfo* GetFirstAudioContent(cricket::ContentInfos* contents);
ContentInfo* GetFirstVideoContent(cricket::ContentInfos* contents);
ContentInfo* GetFirstDataContent(cricket::ContentInfos* contents);
ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc);
ContentInfo* GetFirstVideoContent(SessionDescription* sdesc);
ContentInfo* GetFirstDataContent(SessionDescription* sdesc);
diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc
index cf79144..b558922 100644
--- a/pc/media_session_unittest.cc
+++ b/pc/media_session_unittest.cc
@@ -400,7 +400,7 @@
}
bool IsMediaContentOfType(const ContentInfo* content,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
RTC_DCHECK(content);
return content->media_description()->type() == media_type;
}
@@ -449,7 +449,7 @@
}
// Add a media section to the `session_options`.
-void AddMediaDescriptionOptions(cricket::MediaType type,
+void AddMediaDescriptionOptions(webrtc::MediaType type,
const std::string& mid,
RtpTransceiverDirection direction,
bool stopped,
@@ -460,21 +460,21 @@
void AddAudioVideoSections(RtpTransceiverDirection direction,
cricket::MediaSessionOptions* opts) {
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio", direction,
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio", direction,
kActive, opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video", direction,
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video", direction,
kActive, opts);
}
void AddDataSection(RtpTransceiverDirection direction,
cricket::MediaSessionOptions* opts) {
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_DATA, "data", direction,
+ AddMediaDescriptionOptions(webrtc::MediaType::DATA, "data", direction,
kActive, opts);
}
void AttachSenderToMediaDescriptionOptions(
const std::string& mid,
- cricket::MediaType type,
+ webrtc::MediaType type,
const std::string& track_id,
const std::vector<std::string>& stream_ids,
const std::vector<cricket::RidDescription>& rids,
@@ -483,10 +483,10 @@
cricket::MediaSessionOptions* session_options) {
auto it = FindFirstMediaDescriptionByMid(mid, session_options);
switch (type) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
it->AddAudioSender(track_id, stream_ids);
break;
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
it->AddVideoSender(track_id, stream_ids, rids, simulcast_layers,
num_sim_layer);
break;
@@ -497,7 +497,7 @@
void AttachSenderToMediaDescriptionOptions(
const std::string& mid,
- cricket::MediaType type,
+ webrtc::MediaType type,
const std::string& track_id,
const std::vector<std::string>& stream_ids,
int num_sim_layer,
@@ -525,7 +525,7 @@
// Helper function used to create recv-only audio MediaSessionOptions.
cricket::MediaSessionOptions CreateAudioMediaSession() {
cricket::MediaSessionOptions session_options;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&session_options);
return session_options;
@@ -779,10 +779,10 @@
HeaderExtensionCapabilitiesFromRtpExtensions(video_exts);
for (auto& entry : opts->media_description_options) {
switch (entry.type) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
entry.header_extensions = audio_caps;
break;
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
entry.header_extensions = video_caps;
break;
default:
@@ -814,7 +814,7 @@
EXPECT_FALSE(vc);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const MediaContentDescription* acd = ac->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_THAT(codec_lookup_helper_1_.CodecVendor("")->audio_sendrecv_codecs(),
ElementsAreArray(acd->codecs()));
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached.
@@ -847,7 +847,7 @@
ASSERT_TRUE(vc == NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const MediaContentDescription* acd = ac->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_EQ(2U, acd->codecs().size());
EXPECT_EQ("opus", acd->codecs()[0].name);
EXPECT_EQ("red", acd->codecs()[1].name);
@@ -875,7 +875,7 @@
ASSERT_TRUE(vc == NULL);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const MediaContentDescription* acd = ac->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_EQ(2U, acd->codecs().size());
EXPECT_EQ("red", acd->codecs()[0].name);
EXPECT_EQ("opus", acd->codecs()[1].name);
@@ -896,7 +896,7 @@
EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
const MediaContentDescription* acd = ac->media_description();
const MediaContentDescription* vcd = vc->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_EQ(
codec_lookup_helper_1_.CodecVendor("")->audio_sendrecv_codecs().codecs(),
acd->codecs());
@@ -905,7 +905,7 @@
acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
EXPECT_EQ(cricket::kMediaProtocolDtlsSavpf, acd->protocol());
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, vcd->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, vcd->type());
EXPECT_EQ(
codec_lookup_helper_1_.CodecVendor("")->video_sendrecv_codecs().codecs(),
vcd->codecs());
@@ -923,7 +923,7 @@
cricket::Codec custom_audio_codec = cricket::CreateAudioCodec(audio_format);
custom_audio_codec.id = 123; // picked at random, but valid
auto audio_options = cricket::MediaDescriptionOptions(
- cricket::MEDIA_TYPE_AUDIO, "0", RtpTransceiverDirection::kSendRecv,
+ webrtc::MediaType::AUDIO, "0", RtpTransceiverDirection::kSendRecv,
kActive);
audio_options.codecs_to_include.push_back(custom_audio_codec);
opts.media_description_options.push_back(audio_options);
@@ -931,7 +931,7 @@
cricket::Codec custom_video_codec = cricket::CreateVideoCodec("custom-video");
custom_video_codec.id = 124; // picked at random, but valid
auto video_options = cricket::MediaDescriptionOptions(
- cricket::MEDIA_TYPE_VIDEO, "1", RtpTransceiverDirection::kSendRecv,
+ webrtc::MediaType::VIDEO, "1", RtpTransceiverDirection::kSendRecv,
kActive);
video_options.codecs_to_include.push_back(custom_video_codec);
opts.media_description_options.push_back(video_options);
@@ -947,13 +947,13 @@
EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
const MediaContentDescription* acd = ac->media_description();
const MediaContentDescription* vcd = vc->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
ASSERT_EQ(acd->codecs().size(), 1U);
// Fields in codec are set during the gen process, so simple compare
// does not work.
EXPECT_EQ(acd->codecs()[0].name, custom_audio_codec.name);
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, vcd->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, vcd->type());
ASSERT_EQ(vcd->codecs().size(), 1U);
EXPECT_EQ(vcd->codecs()[0].name, custom_video_codec.name);
}
@@ -971,7 +971,7 @@
cricket::Codec custom_audio_codec = cricket::CreateAudioCodec(audio_format);
custom_audio_codec.id = 123; // picked at random, but valid
auto audio_options = cricket::MediaDescriptionOptions(
- cricket::MEDIA_TYPE_AUDIO, "audio", RtpTransceiverDirection::kSendRecv,
+ webrtc::MediaType::AUDIO, "audio", RtpTransceiverDirection::kSendRecv,
kActive);
audio_options.codecs_to_include.push_back(custom_audio_codec);
answer_opts.media_description_options.push_back(audio_options);
@@ -979,7 +979,7 @@
cricket::Codec custom_video_codec = cricket::CreateVideoCodec("custom-video");
custom_video_codec.id = 124;
auto video_options = cricket::MediaDescriptionOptions(
- cricket::MEDIA_TYPE_VIDEO, "video", RtpTransceiverDirection::kSendRecv,
+ webrtc::MediaType::VIDEO, "video", RtpTransceiverDirection::kSendRecv,
kActive);
video_options.codecs_to_include.push_back(custom_video_codec);
answer_opts.media_description_options.push_back(video_options);
@@ -997,13 +997,13 @@
EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
const MediaContentDescription* acd = ac->media_description();
const MediaContentDescription* vcd = vc->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
ASSERT_EQ(acd->codecs().size(), 1U);
// Fields in codec are set during the gen process, so simple compare
// does not work.
EXPECT_EQ(acd->codecs()[0].name, custom_audio_codec.name);
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, vcd->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, vcd->type());
ASSERT_EQ(vcd->codecs().size(), 1U);
EXPECT_EQ(vcd->codecs()[0].name, custom_video_codec.name);
}
@@ -1039,10 +1039,10 @@
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateUpdatedVideoOfferWithBundle) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kInactive, kStopped,
&opts);
opts.bundle_enabled = true;
@@ -1124,7 +1124,7 @@
TEST_F(MediaSessionDescriptionFactoryTest, ReOfferNoBundleGroupIfAllRejected) {
cricket::MediaSessionOptions opts;
opts.bundle_enabled = true;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -1143,7 +1143,7 @@
TEST_F(MediaSessionDescriptionFactoryTest, ReAnswerNoBundleGroupIfAllRejected) {
cricket::MediaSessionOptions opts;
opts.bundle_enabled = true;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -1166,7 +1166,7 @@
TEST_F(MediaSessionDescriptionFactoryTest, ReOfferChangeBundleOffererTagged) {
cricket::MediaSessionOptions opts;
opts.bundle_enabled = true;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -1174,7 +1174,7 @@
// Reject the audio m= section and add a video m= section.
opts.media_description_options[0].stopped = true;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> reoffer =
@@ -1193,7 +1193,7 @@
TEST_F(MediaSessionDescriptionFactoryTest, ReAnswerChangedBundleOffererTagged) {
cricket::MediaSessionOptions opts;
opts.bundle_enabled = true;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -1203,7 +1203,7 @@
// Reject the audio m= section and add a video m= section.
opts.media_description_options[0].stopped = true;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> reoffer =
@@ -1223,16 +1223,16 @@
// Create an offer with 4 m= sections, initially without BUNDLE groups.
cricket::MediaSessionOptions opts;
opts.bundle_enabled = false;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "1",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "2",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "3",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "3",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "4",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "4",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -1362,10 +1362,10 @@
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSendOnlyOffer) {
cricket::MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kSendOnly, &opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&opts);
@@ -1374,9 +1374,9 @@
ASSERT_TRUE(offer.get());
EXPECT_EQ(2u, offer->contents().size());
EXPECT_TRUE(
- IsMediaContentOfType(&offer->contents()[0], cricket::MEDIA_TYPE_AUDIO));
+ IsMediaContentOfType(&offer->contents()[0], webrtc::MediaType::AUDIO));
EXPECT_TRUE(
- IsMediaContentOfType(&offer->contents()[1], cricket::MEDIA_TYPE_VIDEO));
+ IsMediaContentOfType(&offer->contents()[1], webrtc::MediaType::VIDEO));
EXPECT_EQ(RtpTransceiverDirection::kSendOnly,
GetMediaDirection(&offer->contents()[0]));
@@ -1395,9 +1395,9 @@
ASSERT_TRUE(offer1.get());
EXPECT_EQ(1u, offer1->contents().size());
EXPECT_TRUE(
- IsMediaContentOfType(&offer1->contents()[0], cricket::MEDIA_TYPE_DATA));
+ IsMediaContentOfType(&offer1->contents()[0], webrtc::MediaType::DATA));
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer2(
@@ -1405,11 +1405,11 @@
ASSERT_TRUE(offer2.get());
EXPECT_EQ(2u, offer2->contents().size());
EXPECT_TRUE(
- IsMediaContentOfType(&offer2->contents()[0], cricket::MEDIA_TYPE_DATA));
+ IsMediaContentOfType(&offer2->contents()[0], webrtc::MediaType::DATA));
EXPECT_TRUE(
- IsMediaContentOfType(&offer2->contents()[1], cricket::MEDIA_TYPE_VIDEO));
+ IsMediaContentOfType(&offer2->contents()[1], webrtc::MediaType::VIDEO));
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer3(
@@ -1417,11 +1417,11 @@
ASSERT_TRUE(offer3.get());
EXPECT_EQ(3u, offer3->contents().size());
EXPECT_TRUE(
- IsMediaContentOfType(&offer3->contents()[0], cricket::MEDIA_TYPE_DATA));
+ IsMediaContentOfType(&offer3->contents()[0], webrtc::MediaType::DATA));
EXPECT_TRUE(
- IsMediaContentOfType(&offer3->contents()[1], cricket::MEDIA_TYPE_VIDEO));
+ IsMediaContentOfType(&offer3->contents()[1], webrtc::MediaType::VIDEO));
EXPECT_TRUE(
- IsMediaContentOfType(&offer3->contents()[2], cricket::MEDIA_TYPE_AUDIO));
+ IsMediaContentOfType(&offer3->contents()[2], webrtc::MediaType::AUDIO));
}
// Create a typical audio answer, and ensure it matches what we expect.
@@ -1438,7 +1438,7 @@
EXPECT_FALSE(vc);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const MediaContentDescription* acd = ac->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
@@ -1461,7 +1461,7 @@
EXPECT_FALSE(vc);
EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
const MediaContentDescription* acd = ac->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
@@ -1473,7 +1473,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateAudioAnswerWithNoCommonCodecs) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::vector f1_codecs = {cricket::CreateAudioCodec(96, "opus", 48000, 1)};
@@ -1510,12 +1510,12 @@
EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
const MediaContentDescription* acd = ac->media_description();
const MediaContentDescription* vcd = vc->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
EXPECT_EQ(kAutoBandwidth, acd->bandwidth()); // negotiated auto bw
EXPECT_EQ(0U, acd->first_ssrc()); // no sender is attached
EXPECT_TRUE(acd->rtcp_mux()); // negotiated rtcp-mux
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, vcd->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, vcd->type());
EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
EXPECT_EQ(0U, vcd->first_ssrc()); // no sender is attached
EXPECT_TRUE(vcd->rtcp_mux()); // negotiated rtcp-mux
@@ -1526,7 +1526,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateVideoAnswerWithNoCommonCodecs) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::vector f1_codecs = {cricket::CreateVideoCodec(96, "H264")};
@@ -1551,7 +1551,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateVideoAnswerWithOnlyFecCodecsCommon) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::vector f1_codecs = {cricket::CreateVideoCodec(96, "H264"),
@@ -1706,7 +1706,7 @@
ASSERT_TRUE(offer1.get());
// Appends audio to the offer.
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer2(
@@ -1714,7 +1714,7 @@
ASSERT_TRUE(offer2.get());
// Appends video to the offer.
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer3(
@@ -1726,11 +1726,11 @@
ASSERT_TRUE(answer.get());
EXPECT_EQ(3u, answer->contents().size());
EXPECT_TRUE(
- IsMediaContentOfType(&answer->contents()[0], cricket::MEDIA_TYPE_DATA));
+ IsMediaContentOfType(&answer->contents()[0], webrtc::MediaType::DATA));
EXPECT_TRUE(
- IsMediaContentOfType(&answer->contents()[1], cricket::MEDIA_TYPE_AUDIO));
+ IsMediaContentOfType(&answer->contents()[1], webrtc::MediaType::AUDIO));
EXPECT_TRUE(
- IsMediaContentOfType(&answer->contents()[2], cricket::MEDIA_TYPE_VIDEO));
+ IsMediaContentOfType(&answer->contents()[2], webrtc::MediaType::VIDEO));
}
// TODO(deadbeef): Extend these tests to ensure the correct direction with other
@@ -2014,7 +2014,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
OffersUnstoppedExtensionsWithAudioVideoExtensionStopped) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2022,7 +2022,7 @@
RtpTransceiverDirection::kStopped),
RtpHeaderExtensionCapability("uri2", 3,
RtpTransceiverDirection::kSendOnly)};
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2047,7 +2047,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
OffersUnstoppedExtensionsWithAudioExtensionStopped) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2055,7 +2055,7 @@
RtpTransceiverDirection::kSendOnly),
RtpHeaderExtensionCapability("uri2", 3,
RtpTransceiverDirection::kStopped)};
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2082,7 +2082,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
OffersUnstoppedExtensionsWithVideoExtensionStopped) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2090,7 +2090,7 @@
RtpTransceiverDirection::kSendOnly),
RtpHeaderExtensionCapability("uri2", 7,
RtpTransceiverDirection::kSendRecv)};
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2116,7 +2116,7 @@
TEST_F(MediaSessionDescriptionFactoryTest, AnswersUnstoppedExtensions) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2151,7 +2151,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
AppendsUnstoppedExtensionsToCurrentDescription) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2181,7 +2181,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
AllowsStoppedExtensionsToBeRemovedFromSubsequentOffer) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
opts.media_description_options.back().header_extensions = {
@@ -2503,10 +2503,10 @@
// Create an audio-only answer to a video offer.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerToVideo) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -2675,13 +2675,13 @@
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
cricket::MediaSessionOptions opts;
AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&opts);
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack2, {kMediaStream1}, 1,
&opts);
@@ -2695,7 +2695,7 @@
ASSERT_TRUE(vc);
const MediaContentDescription* acd = ac->media_description();
const MediaContentDescription* vcd = vc->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_EQ(
codec_lookup_helper_1_.CodecVendor("")->audio_sendrecv_codecs().codecs(),
acd->codecs());
@@ -2714,7 +2714,7 @@
acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, vcd->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, vcd->type());
EXPECT_EQ(
codec_lookup_helper_1_.CodecVendor("")->video_sendrecv_codecs().codecs(),
vcd->codecs());
@@ -2729,11 +2729,11 @@
// Update the offer. Add a new video track that is not synched to the
// other tracks and replace audio track 2 with audio track 3.
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack2, {kMediaStream2}, 1,
&opts);
DetachSenderFromMediaSection("audio", kAudioTrack2, &opts);
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack3, {kMediaStream1}, 1,
&opts);
std::unique_ptr<SessionDescription> updated_offer(
@@ -2773,14 +2773,14 @@
// Create an offer with simulcast video stream.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSimulcastVideoOffer) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
const int num_sim_layers = 3;
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1},
num_sim_layers, &opts);
std::unique_ptr<SessionDescription> offer =
@@ -2836,7 +2836,7 @@
// Create an offer with spec-compliant simulcast video stream.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateCompliantSimulcastOffer) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::vector<cricket::RidDescription> send_rids;
@@ -2850,7 +2850,7 @@
simulcast_layers.AddLayer(cricket::SimulcastLayer(send_rids[0].rid, false));
simulcast_layers.AddLayer(cricket::SimulcastLayer(send_rids[1].rid, true));
simulcast_layers.AddLayer(cricket::SimulcastLayer(send_rids[2].rid, false));
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1},
send_rids, simulcast_layers, 0, &opts);
std::unique_ptr<SessionDescription> offer =
@@ -2864,12 +2864,12 @@
// In this scenario, RIDs do not need to be negotiated (there is only one).
TEST_F(MediaSessionDescriptionFactoryTest, TestOfferWithRidsNoSimulcast) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
cricket::RidDescription rid("f", cricket::RidDirection::kSend);
AttachSenderToMediaDescriptionOptions(
- "video", cricket::MEDIA_TYPE_VIDEO, kVideoTrack1, {kMediaStream1}, {rid},
+ "video", webrtc::MediaType::VIDEO, kVideoTrack1, {kMediaStream1}, {rid},
cricket::SimulcastLayerList(), 0, &opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOfferOrError(opts, nullptr).MoveValue();
@@ -2891,17 +2891,17 @@
// In this scenario, the SFU is the caller requesting that we send Simulcast.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateCompliantSimulcastAnswer) {
cricket::MediaSessionOptions offer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOfferOrError(offer_opts, nullptr).MoveValue();
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
@@ -2918,7 +2918,7 @@
simulcast_layers.AddLayer(
cricket::SimulcastLayer(rid_descriptions[2].rid, false));
AttachSenderToMediaDescriptionOptions(
- "video", cricket::MEDIA_TYPE_VIDEO, kVideoTrack1, {kMediaStream1},
+ "video", webrtc::MediaType::VIDEO, kVideoTrack1, {kMediaStream1},
rid_descriptions, simulcast_layers, 0, &answer_opts);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswerOrError(offer.get(), answer_opts, nullptr).MoveValue();
@@ -2932,24 +2932,24 @@
// Note that RID Direction is not the same as the transceiver direction.
TEST_F(MediaSessionDescriptionFactoryTest, TestAnswerWithRidsNoSimulcast) {
cricket::MediaSessionOptions offer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
cricket::RidDescription rid_offer("f", cricket::RidDirection::kSend);
AttachSenderToMediaDescriptionOptions(
- "video", cricket::MEDIA_TYPE_VIDEO, kVideoTrack1, {kMediaStream1},
+ "video", webrtc::MediaType::VIDEO, kVideoTrack1, {kMediaStream1},
{rid_offer}, cricket::SimulcastLayerList(), 0, &offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOfferOrError(offer_opts, nullptr).MoveValue();
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
cricket::RidDescription rid_answer("f", cricket::RidDirection::kReceive);
AttachSenderToMediaDescriptionOptions(
- "video", cricket::MEDIA_TYPE_VIDEO, kVideoTrack1, {kMediaStream1},
+ "video", webrtc::MediaType::VIDEO, kVideoTrack1, {kMediaStream1},
{rid_answer}, cricket::SimulcastLayerList(), 0, &answer_opts);
std::unique_ptr<SessionDescription> answer =
f2_.CreateAnswerOrError(offer.get(), answer_opts, nullptr).MoveValue();
@@ -2975,29 +2975,29 @@
// adding a new video track and removes one of the audio tracks.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
cricket::MediaSessionOptions offer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&offer_opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&offer_opts);
std::unique_ptr<SessionDescription> offer =
f1_.CreateOfferOrError(offer_opts, nullptr).MoveValue();
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack2, {kMediaStream1}, 1,
&answer_opts);
@@ -3012,7 +3012,7 @@
const MediaContentDescription* acd = ac->media_description();
const MediaContentDescription* vcd = vc->media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, acd->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type());
EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
const cricket::StreamParamsVec& audio_streams = acd->streams();
@@ -3029,7 +3029,7 @@
acd->bandwidth()); // default bandwidth (auto)
EXPECT_TRUE(acd->rtcp_mux()); // rtcp-mux defaults on
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, vcd->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, vcd->type());
EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
const cricket::StreamParamsVec& video_streams = vcd->streams();
@@ -3042,7 +3042,7 @@
// Update the answer. Add a new video track that is not synched to the
// other tracks and remove 1 audio track.
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack2, {kMediaStream2}, 1,
&answer_opts);
DetachSenderFromMediaSection("audio", kAudioTrack2, &answer_opts);
@@ -3139,7 +3139,7 @@
cricket::CodecList{}, cricket::CodecList{});
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "a0",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "a0",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -3175,7 +3175,7 @@
cricket::CodecList{}, cricket::CodecList{});
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "v0",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "v0",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -3206,7 +3206,7 @@
// Perform initial offer/answer in reverse (`f2_` as offerer) so that the
// second offer/answer is forward (`f1_` as offerer).
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "a0",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "a0",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -3240,7 +3240,7 @@
// Perform initial offer/answer in reverse (`f2_` as offerer) so that the
// second offer/answer is forward (`f1_` as offerer).
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "v0",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "v0",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -3268,7 +3268,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
RespondentCreatesOfferAfterCreatingAnswerWithRtx) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<cricket::Codec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
@@ -3324,7 +3324,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
RespondentCreatesOfferAfterCreatingAnswerWithRemappedRtxPayloadType) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
// We specifically choose different preferred payload types for VP8 to
@@ -3390,7 +3390,7 @@
f1_codecs);
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
@@ -3487,7 +3487,7 @@
// Test that RTX is ignored when there is no associated payload type parameter.
TEST_F(MediaSessionDescriptionFactoryTest, RtxWithoutApt) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<cricket::Codec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
@@ -3534,7 +3534,7 @@
// type doesn't match the local value.
TEST_F(MediaSessionDescriptionFactoryTest, FilterOutRtxIfAptDoesntMatch) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<cricket::Codec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
@@ -3569,7 +3569,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
FilterOutUnsupportedRtxWhenCreatingAnswer) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<cricket::Codec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
@@ -3612,7 +3612,7 @@
// to add another.
TEST_F(MediaSessionDescriptionFactoryTest, AddSecondRtxInNewOffer) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
std::vector<cricket::Codec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
@@ -3653,11 +3653,11 @@
// generated for each simulcast ssrc and correctly grouped.
TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateMultipleRtxSsrcs) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Add simulcast streams.
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
"stream1", {"stream1label"}, 3, &opts);
// Use a single real codec, and then add RTX for it.
@@ -3698,11 +3698,11 @@
ScopedKeyValueConfig override_field_trials(field_trials,
"WebRTC-FlexFEC-03/Enabled/");
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Add single stream.
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
"stream1", {"stream1label"}, 1, &opts);
// Use a single real codec, and then add FlexFEC for it.
@@ -3742,11 +3742,11 @@
ScopedKeyValueConfig override_field_trials(field_trials,
"WebRTC-FlexFEC-03/Enabled/");
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Add simulcast streams.
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
"stream1", {"stream1label"}, 3, &opts);
// Use a single real codec, and then add FlexFEC for it.
@@ -3914,7 +3914,7 @@
// ensure the TransportInfo in the SessionDescription matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferAudio) {
cricket::MediaSessionOptions options;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
TestTransportInfo(true, options, false);
@@ -3923,7 +3923,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoOfferIceRenomination) {
cricket::MediaSessionOptions options;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
options.media_description_options[0]
@@ -3933,7 +3933,7 @@
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferAudioCurrent) {
cricket::MediaSessionOptions options;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
TestTransportInfo(true, options, true);
@@ -3969,7 +3969,7 @@
TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoAnswerAudio) {
cricket::MediaSessionOptions options;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
TestTransportInfo(false, options, false);
@@ -3978,7 +3978,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoAnswerIceRenomination) {
cricket::MediaSessionOptions options;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
options.media_description_options[0]
@@ -3989,7 +3989,7 @@
TEST_F(MediaSessionDescriptionFactoryTest,
TestTransportInfoAnswerAudioCurrent) {
cricket::MediaSessionOptions options;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kRecvOnly, kActive,
&options);
TestTransportInfo(false, options, true);
@@ -4112,13 +4112,13 @@
// Test that the generated MIDs match the existing offer.
TEST_F(MediaSessionDescriptionFactoryTest, TestMIDsMatchesExistingOffer) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio_modified",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio_modified",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video_modified",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video_modified",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_DATA, "data_modified",
+ AddMediaDescriptionOptions(webrtc::MediaType::DATA, "data_modified",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Create offer.
@@ -4144,31 +4144,31 @@
TEST_F(MediaSessionDescriptionFactoryTest,
CreateOfferWithMultipleAVMediaSections) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio_1",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio_1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("audio_1", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio_1", webrtc::MediaType::AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video_1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video_1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video_1", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video_1", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio_2",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio_2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("audio_2", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio_2", webrtc::MediaType::AUDIO,
kAudioTrack2, {kMediaStream2}, 1,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video_2",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video_2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video_2", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video_2", webrtc::MediaType::VIDEO,
kVideoTrack2, {kMediaStream2}, 1,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -4206,31 +4206,31 @@
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerWithMultipleAVMediaSections) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio_1",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio_1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("audio_1", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio_1", webrtc::MediaType::AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video_1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video_1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video_1", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video_1", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio_2",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio_2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("audio_2", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio_2", webrtc::MediaType::AUDIO,
kAudioTrack2, {kMediaStream2}, 1,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video_2",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video_2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video_2", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video_2", webrtc::MediaType::VIDEO,
kVideoTrack2, {kMediaStream2}, 1,
&opts);
@@ -4274,10 +4274,10 @@
CreateOfferWithMediaSectionStoppedByOfferer) {
// Create an offer with two audio sections and one of them is stopped.
cricket::MediaSessionOptions offer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio1",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio2",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio2",
RtpTransceiverDirection::kInactive, kStopped,
&offer_opts);
std::unique_ptr<SessionDescription> offer =
@@ -4294,10 +4294,10 @@
CreateAnswerWithMediaSectionStoppedByOfferer) {
// Create an offer with two audio sections and one of them is stopped.
cricket::MediaSessionOptions offer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio1",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio2",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio2",
RtpTransceiverDirection::kInactive, kStopped,
&offer_opts);
std::unique_ptr<SessionDescription> offer =
@@ -4309,10 +4309,10 @@
// Create an answer based on the offer.
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio1",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio2",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio2",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
std::unique_ptr<SessionDescription> answer =
@@ -4328,10 +4328,10 @@
CreateAnswerWithMediaSectionRejectedByAnswerer) {
// Create an offer with two audio sections.
cricket::MediaSessionOptions offer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio1",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio2",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio2",
RtpTransceiverDirection::kSendRecv, kActive,
&offer_opts);
std::unique_ptr<SessionDescription> offer =
@@ -4343,10 +4343,10 @@
// The answerer rejects one of the audio sections.
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio1",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio1",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio2",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio2",
RtpTransceiverDirection::kInactive, kStopped,
&answer_opts);
std::unique_ptr<SessionDescription> answer =
@@ -4367,10 +4367,10 @@
cricket::MediaSessionOptions opts;
// This tests put video section first because normally audio comes first by
// default.
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
std::unique_ptr<SessionDescription> offer =
@@ -4387,10 +4387,10 @@
TEST_F(MediaSessionDescriptionFactoryTest,
PayloadTypesSharedByMediaSectionsOfSameType) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video2",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Create an offer with two video sections using same codecs.
@@ -4437,7 +4437,7 @@
f2_codecs);
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
@@ -4475,7 +4475,7 @@
f2_codecs);
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
@@ -4514,7 +4514,7 @@
f2_codecs);
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
@@ -4552,7 +4552,7 @@
f2_codecs);
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
@@ -4581,10 +4581,10 @@
TEST_F(MediaSessionDescriptionFactoryTest,
CreateOfferRespectsCodecPreferenceOrder) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video2",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Create an offer with two video sections using same codecs.
@@ -4617,10 +4617,10 @@
TEST_F(MediaSessionDescriptionFactoryTest,
CreateAnswerRespectsCodecPreferenceOrder) {
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video1",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video1",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video2",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video2",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
// Create an offer with two video sections using same codecs.
@@ -4679,10 +4679,10 @@
video_codecs2);
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
@@ -4733,7 +4733,7 @@
{h264_pm1});
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
@@ -4853,12 +4853,12 @@
recv_codecs);
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio", direction,
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio", direction,
kActive, &opts);
if (direction == RtpTransceiverDirection::kSendRecv ||
direction == RtpTransceiverDirection::kSendOnly) {
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&opts);
}
@@ -4968,11 +4968,11 @@
VectorFromIndices(kOfferAnswerCodecs, kAnswerRecvCodecs));
cricket::MediaSessionOptions offer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
- offer_direction, kActive, &offer_opts);
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio", offer_direction,
+ kActive, &offer_opts);
if (webrtc::RtpTransceiverDirectionHasSend(offer_direction)) {
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&offer_opts);
}
@@ -4982,11 +4982,11 @@
ASSERT_TRUE(offer.get());
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO, "audio",
+ AddMediaDescriptionOptions(webrtc::MediaType::AUDIO, "audio",
answer_direction, kActive, &answer_opts);
if (webrtc::RtpTransceiverDirectionHasSend(answer_direction)) {
- AttachSenderToMediaDescriptionOptions("audio", cricket::MEDIA_TYPE_AUDIO,
+ AttachSenderToMediaDescriptionOptions("audio", webrtc::MediaType::AUDIO,
kAudioTrack1, {kMediaStream1}, 1,
&answer_opts);
}
@@ -5000,7 +5000,7 @@
// to send nor receive audio. The checks are still in place if at some point
// we'd instead create an inactive stream.
if (ac) {
- ASSERT_EQ(cricket::MEDIA_TYPE_AUDIO, ac->media_description()->type());
+ ASSERT_EQ(webrtc::MediaType::AUDIO, ac->media_description()->type());
const MediaContentDescription* acd = ac->media_description();
std::vector<cricket::Codec> target_codecs;
@@ -5162,11 +5162,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -5180,10 +5180,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5217,10 +5217,10 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -5234,7 +5234,7 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level6LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&answer_opts);
@@ -5268,7 +5268,7 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
@@ -5282,7 +5282,7 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&answer_opts);
@@ -5323,11 +5323,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -5341,10 +5341,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5385,11 +5385,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -5403,10 +5403,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5447,11 +5447,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -5465,10 +5465,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5519,11 +5519,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -5537,10 +5537,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5581,11 +5581,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -5599,10 +5599,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level4LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5643,7 +5643,7 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
@@ -5657,10 +5657,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level6LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5701,7 +5701,7 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
@@ -5715,10 +5715,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5759,7 +5759,7 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
@@ -5773,10 +5773,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5817,7 +5817,7 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
@@ -5831,10 +5831,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5875,7 +5875,7 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&opts);
@@ -5889,10 +5889,10 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level6LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&answer_opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&answer_opts);
@@ -5933,11 +5933,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -5951,7 +5951,7 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level52LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&answer_opts);
@@ -5992,11 +5992,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -6010,7 +6010,7 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level6LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&answer_opts);
@@ -6051,11 +6051,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -6069,7 +6069,7 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level6LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&answer_opts);
@@ -6110,11 +6110,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -6128,7 +6128,7 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level6LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&answer_opts);
@@ -6169,11 +6169,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendOnly, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
@@ -6187,7 +6187,7 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level4LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&answer_opts);
@@ -6224,11 +6224,11 @@
.codecs());
cricket::MediaSessionOptions opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kSendRecv, kActive,
&opts);
- AttachSenderToMediaDescriptionOptions("video", cricket::MEDIA_TYPE_VIDEO,
+ AttachSenderToMediaDescriptionOptions("video", webrtc::MediaType::VIDEO,
kVideoTrack1, {kMediaStream1}, 1,
&opts);
std::vector<RtpCodecCapability> preferences;
@@ -6248,7 +6248,7 @@
CheckH265Level(ocd->codecs(), kVideoCodecsH265Level4LevelId);
cricket::MediaSessionOptions answer_opts;
- AddMediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO, "video",
+ AddMediaDescriptionOptions(webrtc::MediaType::VIDEO, "video",
RtpTransceiverDirection::kRecvOnly, kActive,
&answer_opts);
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 9c7492d..edcaec0 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -598,12 +598,12 @@
rtp_manager_->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), rtc::make_ref_counted<RtpTransceiver>(
- cricket::MEDIA_TYPE_AUDIO, context_.get(),
+ webrtc::MediaType::AUDIO, context_.get(),
codec_lookup_helper_.get())));
rtp_manager_->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), rtc::make_ref_counted<RtpTransceiver>(
- cricket::MEDIA_TYPE_VIDEO, context_.get(),
+ webrtc::MediaType::VIDEO, context_.get(),
codec_lookup_helper_.get())));
}
@@ -937,11 +937,11 @@
}
} else {
bool removed;
- if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (sender->media_type() == webrtc::MediaType::AUDIO) {
removed = rtp_manager()->GetAudioTransceiver()->internal()->RemoveSender(
sender.get());
} else {
- RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
+ RTC_DCHECK_EQ(webrtc::MediaType::VIDEO, sender->media_type());
removed = rtp_manager()->GetVideoTransceiver()->internal()->RemoveSender(
sender.get());
}
@@ -986,11 +986,11 @@
if (!track) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
}
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
- media_type = cricket::MEDIA_TYPE_AUDIO;
+ media_type = webrtc::MediaType::AUDIO;
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
- media_type = cricket::MEDIA_TYPE_VIDEO;
+ media_type = webrtc::MediaType::VIDEO;
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track kind is not audio or video");
@@ -999,12 +999,12 @@
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
-PeerConnection::AddTransceiver(cricket::MediaType media_type) {
+PeerConnection::AddTransceiver(webrtc::MediaType media_type) {
return AddTransceiver(media_type, RtpTransceiverInit());
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
-PeerConnection::AddTransceiver(cricket::MediaType media_type,
+PeerConnection::AddTransceiver(webrtc::MediaType media_type,
const RtpTransceiverInit& init) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
@@ -1013,8 +1013,8 @@
}
RTC_CHECK(IsUnifiedPlan())
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
- if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO)) {
+ if (!(media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"media type is not audio or video");
}
@@ -1023,7 +1023,7 @@
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool update_negotiation_needed) {
@@ -1032,13 +1032,13 @@
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
- RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO));
+ RTC_DCHECK((media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO));
if (track) {
RTC_DCHECK_EQ(media_type,
(track->kind() == MediaStreamTrackInterface::kAudioKind
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO));
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO));
}
size_t num_rids = absl::c_count_if(init.send_encodings,
@@ -1073,7 +1073,7 @@
// Encodings are dropped from the tail if too many are provided.
size_t max_simulcast_streams =
- media_type == cricket::MEDIA_TYPE_VIDEO ? kMaxSimulcastStreams : 1u;
+ media_type == webrtc::MediaType::VIDEO ? kMaxSimulcastStreams : 1u;
if (parameters.encodings.size() > max_simulcast_streams) {
parameters.encodings.erase(
parameters.encodings.begin() + max_simulcast_streams,
@@ -1111,7 +1111,7 @@
// codec selection against supported values.
cricket::CodecVendor codec_vendor(context_->media_engine(), false,
context_->env().field_trials());
- if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type == webrtc::MediaType::VIDEO) {
codecs = codec_vendor.video_send_codecs().codecs();
} else {
codecs = codec_vendor.audio_send_codecs().codecs();
@@ -1126,7 +1126,7 @@
LOG_AND_RETURN_ERROR(result.type(), result.message());
}
- RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
+ RTC_LOG(LS_INFO) << "Adding " << webrtc::MediaTypeToString(media_type)
<< " transceiver in response to a call to AddTransceiver.";
// Set the sender ID equal to the track ID if the track is specified unless
// that sender ID is already in use.
@@ -2588,12 +2588,12 @@
int num_data_mlines = 0;
for (const ContentInfo& content :
remote_description.description()->contents()) {
- cricket::MediaType media_type = content.media_description()->type();
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ webrtc::MediaType media_type = content.media_description()->type();
+ if (media_type == webrtc::MediaType::AUDIO) {
num_audio_mlines += 1;
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ } else if (media_type == webrtc::MediaType::VIDEO) {
num_video_mlines += 1;
- } else if (media_type == cricket::MEDIA_TYPE_DATA) {
+ } else if (media_type == webrtc::MediaType::DATA) {
num_data_mlines += 1;
}
}
@@ -2739,7 +2739,7 @@
std::vector<RtpTransceiverProxyRefPtr> transceivers) {
TRACE_EVENT0("webrtc", "PeerConnection::ReportTransportStats");
Thread::ScopedDisallowBlockingCalls no_blocking_calls;
- std::map<std::string, std::set<cricket::MediaType>>
+ std::map<std::string, std::set<webrtc::MediaType>>
media_types_by_transport_name;
for (const auto& transceiver : transceivers) {
if (transceiver->internal()->channel()) {
@@ -2755,13 +2755,13 @@
transport_controller_->GetDtlsTransport(*sctp_mid_n_);
if (dtls_transport) {
media_types_by_transport_name[dtls_transport->transport_name()].insert(
- cricket::MEDIA_TYPE_DATA);
+ webrtc::MediaType::DATA);
}
}
for (const auto& entry : media_types_by_transport_name) {
const std::string& transport_name = entry.first;
- const std::set<cricket::MediaType> media_types = entry.second;
+ const std::set<webrtc::MediaType> media_types = entry.second;
cricket::TransportStats stats;
if (transport_controller_->GetStats(transport_name, &stats)) {
ReportBestConnectionState(stats);
@@ -2824,7 +2824,7 @@
void PeerConnection::ReportNegotiatedCiphers(
bool dtls_enabled,
const cricket::TransportStats& stats,
- const std::set<cricket::MediaType>& media_types) {
+ const std::set<webrtc::MediaType>& media_types) {
if (!dtls_enabled || stats.channel_stats.empty()) {
return;
}
@@ -2837,19 +2837,19 @@
}
if (ssl_cipher_suite != kTlsNullWithNullNull) {
- for (cricket::MediaType media_type : media_types) {
+ for (webrtc::MediaType media_type : media_types) {
switch (media_type) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
kSslCipherSuiteMaxValue);
break;
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
kSslCipherSuiteMaxValue);
break;
- case cricket::MEDIA_TYPE_DATA:
+ case webrtc::MediaType::DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
kSslCipherSuiteMaxValue);
@@ -2864,19 +2864,19 @@
uint16_t ssl_peer_signature_algorithm =
stats.channel_stats[0].ssl_peer_signature_algorithm;
if (ssl_peer_signature_algorithm != kSslSignatureAlgorithmUnknown) {
- for (cricket::MediaType media_type : media_types) {
+ for (webrtc::MediaType media_type : media_types) {
switch (media_type) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Audio",
ssl_peer_signature_algorithm, kSslSignatureAlgorithmMaxValue);
break;
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Video",
ssl_peer_signature_algorithm, kSslSignatureAlgorithmMaxValue);
break;
- case cricket::MEDIA_TYPE_DATA:
+ case webrtc::MediaType::DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Data",
ssl_peer_signature_algorithm, kSslSignatureAlgorithmMaxValue);
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 0a78847..b3679bf 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -150,9 +150,9 @@
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
- cricket::MediaType media_type) override;
+ webrtc::MediaType media_type) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const RtpTransceiverInit& init) override;
rtc::scoped_refptr<RtpSenderInterface> CreateSender(
@@ -416,7 +416,7 @@
// Internal implementation for AddTransceiver family of methods. If
// `fire_callback` is set, fires OnRenegotiationNeeded callback if successful.
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool fire_callback = true) override;
@@ -587,7 +587,7 @@
static void ReportNegotiatedCiphers(
bool dtls_enabled,
const cricket::TransportStats& stats,
- const std::set<cricket::MediaType>& media_types);
+ const std::set<webrtc::MediaType>& media_types);
void ReportIceCandidateCollected(const Candidate& candidate)
RTC_RUN_ON(signaling_thread());
diff --git a/pc/peer_connection_bundle_unittest.cc b/pc/peer_connection_bundle_unittest.cc
index 54e24ab..029dedb 100644
--- a/pc/peer_connection_bundle_unittest.cc
+++ b/pc/peer_connection_bundle_unittest.cc
@@ -104,7 +104,7 @@
using PeerConnectionWrapper::PeerConnectionWrapper;
bool AddIceCandidateToMedia(Candidate* candidate,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
auto* desc = pc()->remote_description()->description();
for (size_t i = 0; i < desc->contents().size(); i++) {
const auto& content = desc->contents()[i];
@@ -126,7 +126,7 @@
cricket::VoiceChannel* voice_channel() {
auto transceivers = GetInternalPeerConnection()->GetTransceiversInternal();
for (const auto& transceiver : transceivers) {
- if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (transceiver->media_type() == webrtc::MediaType::AUDIO) {
return static_cast<cricket::VoiceChannel*>(
transceiver->internal()->channel());
}
@@ -141,7 +141,7 @@
cricket::VideoChannel* video_channel() {
auto transceivers = GetInternalPeerConnection()->GetTransceiversInternal();
for (const auto& transceiver : transceivers) {
- if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
+ if (transceiver->media_type() == webrtc::MediaType::VIDEO) {
return static_cast<cricket::VideoChannel*>(
transceiver->internal()->channel());
}
@@ -662,15 +662,15 @@
Candidate audio_candidate1 = CreateLocalUdpCandidate(kAudioAddress1);
ASSERT_TRUE(caller->AddIceCandidateToMedia(&audio_candidate1,
- cricket::MEDIA_TYPE_AUDIO));
+ webrtc::MediaType::AUDIO));
Candidate video_candidate = CreateLocalUdpCandidate(kVideoAddress);
ASSERT_TRUE(caller->AddIceCandidateToMedia(&video_candidate,
- cricket::MEDIA_TYPE_VIDEO));
+ webrtc::MediaType::VIDEO));
Candidate audio_candidate2 = CreateLocalUdpCandidate(kAudioAddress2);
ASSERT_TRUE(caller->AddIceCandidateToMedia(&audio_candidate2,
- cricket::MEDIA_TYPE_AUDIO));
+ webrtc::MediaType::AUDIO));
EXPECT_THAT(
WaitUntil(
diff --git a/pc/peer_connection_data_channel_unittest.cc b/pc/peer_connection_data_channel_unittest.cc
index d55afe9..d6b8838 100644
--- a/pc/peer_connection_data_channel_unittest.cc
+++ b/pc/peer_connection_data_channel_unittest.cc
@@ -210,10 +210,10 @@
auto offer = caller->CreateOffer();
const auto& offer_contents = offer->description()->contents();
- ASSERT_EQ(cricket::MEDIA_TYPE_AUDIO,
+ ASSERT_EQ(webrtc::MediaType::AUDIO,
offer_contents[0].media_description()->type());
auto audio_mid = offer_contents[0].mid();
- ASSERT_EQ(cricket::MEDIA_TYPE_DATA,
+ ASSERT_EQ(webrtc::MediaType::DATA,
offer_contents[2].media_description()->type());
auto data_mid = offer_contents[2].mid();
diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc
index 0e9ed6a..c1f784e 100644
--- a/pc/peer_connection_encodings_integrationtest.cc
+++ b/pc/peer_connection_encodings_integrationtest.cc
@@ -261,7 +261,7 @@
absl::string_view codec_name) {
std::vector<RtpCodecCapability> codecs =
pc_wrapper->pc_factory()
- ->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO)
.codecs;
return std::find_if(codecs.begin(), codecs.end(),
[&codec_name](const RtpCodecCapability& codec) {
@@ -274,7 +274,7 @@
absl::string_view codec_name) {
std::vector<RtpCodecCapability> codecs =
pc_wrapper->pc_factory()
- ->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO)
.codecs;
codecs.erase(std::remove_if(codecs.begin(), codecs.end(),
[&codec_name](const RtpCodecCapability& codec) {
@@ -590,7 +590,7 @@
// codec preferences because we want the sender to think SVC is a possibility.
std::vector<RtpCodecCapability> codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
EXPECT_THAT(codecs[0].name, StrCaseEq("VP8"));
// Attempt SVC (L3T3_KEY), which is not possible with VP8, but the sender does
@@ -1365,7 +1365,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
RtpParameters parameters = audio_transceiver->sender()->GetParameters();
@@ -1377,7 +1377,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
RtpParameters parameters = video_transceiver->sender()->GetParameters();
@@ -1396,7 +1396,7 @@
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
std::optional<RtpCodecCapability> pcmu =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO,
"pcmu");
ASSERT_TRUE(pcmu);
@@ -1437,7 +1437,7 @@
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
std::optional<RtpCodecCapability> vp9 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp9");
ASSERT_TRUE(vp9);
@@ -1484,7 +1484,7 @@
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
std::optional<RtpCodecCapability> pcmu =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO,
"pcmu");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
@@ -1521,7 +1521,7 @@
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
std::optional<RtpCodecCapability> pcmu =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO,
"pcmu");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
@@ -1570,7 +1570,7 @@
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
std::optional<RtpCodecCapability> vp9 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp9");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
@@ -1614,7 +1614,7 @@
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
std::optional<RtpCodecCapability> vp9 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp9");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
@@ -1660,7 +1660,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
RtpCodec dummy_codec;
- dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO;
+ dummy_codec.kind = webrtc::MediaType::AUDIO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
dummy_codec.num_channels = 2;
@@ -1672,7 +1672,7 @@
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO, init);
EXPECT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
RTCErrorType::UNSUPPORTED_OPERATION);
@@ -1683,7 +1683,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
RtpCodec dummy_codec;
- dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO;
+ dummy_codec.kind = webrtc::MediaType::VIDEO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
@@ -1694,7 +1694,7 @@
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
EXPECT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
RTCErrorType::UNSUPPORTED_OPERATION);
@@ -1705,13 +1705,13 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
RtpCodec dummy_codec;
- dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO;
+ dummy_codec.kind = webrtc::MediaType::AUDIO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
dummy_codec.num_channels = 2;
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
@@ -1727,12 +1727,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
RtpCodec dummy_codec;
- dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO;
+ dummy_codec.kind = webrtc::MediaType::VIDEO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
@@ -1750,13 +1750,13 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::optional<RtpCodecCapability> opus =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO,
"opus");
ASSERT_TRUE(opus);
std::vector<RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
.codecs;
not_opus_codecs.erase(
std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(),
@@ -1766,7 +1766,7 @@
not_opus_codecs.end());
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
@@ -1789,13 +1789,13 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::optional<RtpCodecCapability> opus =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO,
"opus");
ASSERT_TRUE(opus);
std::vector<RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
.codecs;
not_opus_codecs.erase(
std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(),
@@ -1805,7 +1805,7 @@
not_opus_codecs.end());
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
@@ -1856,7 +1856,7 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
@@ -1917,13 +1917,13 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::optional<RtpCodecCapability> vp8 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp8");
ASSERT_TRUE(vp8);
std::vector<RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
not_vp8_codecs.erase(
std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(),
@@ -1933,7 +1933,7 @@
not_vp8_codecs.end());
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
@@ -1956,13 +1956,13 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::optional<RtpCodecCapability> vp8 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp8");
ASSERT_TRUE(vp8);
std::vector<RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
not_vp8_codecs.erase(
std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(),
@@ -1972,7 +1972,7 @@
not_vp8_codecs.end());
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
@@ -2018,13 +2018,13 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::optional<RtpCodecCapability> opus =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO,
"opus");
ASSERT_TRUE(opus);
std::vector<RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
.codecs;
not_opus_codecs.erase(
std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(),
@@ -2040,7 +2040,7 @@
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO, init);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
@@ -2067,16 +2067,16 @@
std::vector<RtpCodecCapability> send_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
.codecs;
std::optional<RtpCodecCapability> opus =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO,
"opus");
ASSERT_TRUE(opus);
std::optional<RtpCodecCapability> red =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO,
"red");
ASSERT_TRUE(red);
@@ -2087,7 +2087,7 @@
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO, init);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
@@ -2122,7 +2122,7 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
std::optional<RtpCodecCapability> vp8 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp8");
ASSERT_TRUE(vp8);
@@ -2134,7 +2134,7 @@
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
@@ -2152,13 +2152,13 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::optional<RtpCodecCapability> vp8 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp8");
ASSERT_TRUE(vp8);
std::vector<RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
not_vp8_codecs.erase(
std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(),
@@ -2179,7 +2179,7 @@
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
ASSERT_TRUE(transceiver_or_error.ok());
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
@@ -2210,11 +2210,11 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::optional<RtpCodecCapability> vp8 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp8");
ASSERT_TRUE(vp8);
std::optional<RtpCodecCapability> vp9 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp9");
ASSERT_TRUE(vp9);
@@ -2231,7 +2231,7 @@
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
ASSERT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
RTCErrorType::UNSUPPORTED_OPERATION);
@@ -2248,11 +2248,11 @@
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
std::optional<RtpCodecCapability> vp8 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp8");
ASSERT_TRUE(vp8);
std::optional<RtpCodecCapability> vp9 =
- local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
"vp9");
ASSERT_TRUE(vp9);
@@ -2269,7 +2269,7 @@
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
- local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
ASSERT_TRUE(transceiver_or_error.ok());
}
@@ -2285,7 +2285,7 @@
encoding.scale_resolution_down_to = {.width = 1280, .height = 720};
init.send_encodings.push_back(encoding);
auto transceiver_or_error =
- pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
EXPECT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
RTCErrorType::UNSUPPORTED_OPERATION);
@@ -2295,7 +2295,7 @@
.height = 0};
init.send_encodings[1].scale_resolution_down_to = {.width = 0, .height = 720};
transceiver_or_error =
- pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
EXPECT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
RTCErrorType::UNSUPPORTED_OPERATION);
@@ -2309,7 +2309,7 @@
.height = 720};
init.send_encodings[1].scale_resolution_down_by = 2.0;
transceiver_or_error =
- pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
ASSERT_TRUE(transceiver_or_error.ok());
// SetParameters: If `scale_resolution_down_to` is specified on any active
@@ -3161,7 +3161,7 @@
rtc::scoped_refptr<RtpTransceiverInterface> transceiver =
local_pc_wrapper->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO)
+ ->AddTransceiver(webrtc::MediaType::VIDEO)
.MoveValue();
std::vector<RtpCodecCapability> preferred_codecs =
GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "H265");
@@ -3248,7 +3248,7 @@
// H265.
std::vector<RtpCodecCapability> sender_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
auto it = std::find_if(sender_codecs.begin(), sender_codecs.end(),
[](const RtpCodecCapability codec_capability) {
@@ -3294,7 +3294,7 @@
rtc::scoped_refptr<RtpTransceiverInterface> transceiver =
local_pc_wrapper->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO)
+ ->AddTransceiver(webrtc::MediaType::VIDEO)
.MoveValue();
// Filter on codec name and assert that sender capabilities have codecs for
@@ -3302,7 +3302,7 @@
// {sendrecv, recvonly}.
std::vector<RtpCodecCapability> send_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
send_codecs.erase(std::remove_if(send_codecs.begin(), send_codecs.end(),
[](const RtpCodecCapability& codec) {
@@ -3311,7 +3311,7 @@
send_codecs.end());
std::vector<RtpCodecCapability> recv_codecs =
local_pc_wrapper->pc_factory()
- ->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO)
.codecs;
recv_codecs.erase(std::remove_if(recv_codecs.begin(), recv_codecs.end(),
[](const RtpCodecCapability& codec) {
diff --git a/pc/peer_connection_factory.cc b/pc/peer_connection_factory.cc
index 45fcefd..80b3fe7 100644
--- a/pc/peer_connection_factory.cc
+++ b/pc/peer_connection_factory.cc
@@ -149,17 +149,17 @@
}
RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities(
- cricket::MediaType kind) const {
+ webrtc::MediaType kind) const {
RTC_DCHECK_RUN_ON(signaling_thread());
switch (kind) {
- case cricket::MEDIA_TYPE_AUDIO: {
+ case webrtc::MediaType::AUDIO: {
cricket::Codecs cricket_codecs;
cricket_codecs = codec_vendor_.audio_send_codecs().codecs();
auto extensions =
GetDefaultEnabledRtpHeaderExtensions(media_engine()->voice());
return ToRtpCapabilities(cricket_codecs, extensions);
}
- case cricket::MEDIA_TYPE_VIDEO: {
+ case webrtc::MediaType::VIDEO: {
cricket::Codecs cricket_codecs;
cricket_codecs = codec_vendor_.video_send_codecs().codecs();
auto extensions =
@@ -174,17 +174,17 @@
}
RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
- cricket::MediaType kind) const {
+ webrtc::MediaType kind) const {
RTC_DCHECK_RUN_ON(signaling_thread());
switch (kind) {
- case cricket::MEDIA_TYPE_AUDIO: {
+ case webrtc::MediaType::AUDIO: {
cricket::Codecs cricket_codecs;
cricket_codecs = codec_vendor_.audio_recv_codecs().codecs();
auto extensions =
GetDefaultEnabledRtpHeaderExtensions(media_engine()->voice());
return ToRtpCapabilities(cricket_codecs, extensions);
}
- case cricket::MEDIA_TYPE_VIDEO: {
+ case webrtc::MediaType::VIDEO: {
cricket::Codecs cricket_codecs =
codec_vendor_.video_recv_codecs().codecs();
auto extensions =
diff --git a/pc/peer_connection_factory.h b/pc/peer_connection_factory.h
index fe04e0f..651ab21 100644
--- a/pc/peer_connection_factory.h
+++ b/pc/peer_connection_factory.h
@@ -65,10 +65,10 @@
PeerConnectionDependencies dependencies) override;
RtpCapabilities GetRtpSenderCapabilities(
- cricket::MediaType kind) const override;
+ webrtc::MediaType kind) const override;
RtpCapabilities GetRtpReceiverCapabilities(
- cricket::MediaType kind) const override;
+ webrtc::MediaType kind) const override;
rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
const std::string& stream_id) override;
diff --git a/pc/peer_connection_factory_proxy.h b/pc/peer_connection_factory_proxy.h
index 88ee3b1..3552c7c 100644
--- a/pc/peer_connection_factory_proxy.h
+++ b/pc/peer_connection_factory_proxy.h
@@ -36,12 +36,10 @@
CreatePeerConnectionOrError,
const PeerConnectionInterface::RTCConfiguration&,
PeerConnectionDependencies)
-PROXY_CONSTMETHOD1(RtpCapabilities,
- GetRtpSenderCapabilities,
- cricket::MediaType)
+PROXY_CONSTMETHOD1(RtpCapabilities, GetRtpSenderCapabilities, webrtc::MediaType)
PROXY_CONSTMETHOD1(RtpCapabilities,
GetRtpReceiverCapabilities,
- cricket::MediaType)
+ webrtc::MediaType)
PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
CreateLocalMediaStream,
const std::string&)
diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc
index cad99fb..e8f6a8a 100644
--- a/pc/peer_connection_factory_unittest.cc
+++ b/pc/peer_connection_factory_unittest.cc
@@ -198,7 +198,7 @@
}
void VerifyAudioCodecCapability(const RtpCodecCapability& codec) {
- EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_AUDIO);
+ EXPECT_EQ(codec.kind, webrtc::MediaType::AUDIO);
EXPECT_FALSE(codec.name.empty());
EXPECT_GT(codec.clock_rate, 0);
EXPECT_GT(codec.num_channels, 0);
@@ -206,7 +206,7 @@
void VerifyVideoCodecCapability(const RtpCodecCapability& codec,
bool sender) {
- EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO);
+ EXPECT_EQ(codec.kind, webrtc::MediaType::VIDEO);
EXPECT_FALSE(codec.name.empty());
EXPECT_GT(codec.clock_rate, 0);
if (sender) {
@@ -345,7 +345,7 @@
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) {
RtpCapabilities audio_capabilities =
- factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO);
+ factory_->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO);
EXPECT_FALSE(audio_capabilities.codecs.empty());
for (const auto& codec : audio_capabilities.codecs) {
VerifyAudioCodecCapability(codec);
@@ -358,7 +358,7 @@
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) {
RtpCapabilities video_capabilities =
- factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
+ factory_->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO);
EXPECT_FALSE(video_capabilities.codecs.empty());
for (const auto& codec : video_capabilities.codecs) {
VerifyVideoCodecCapability(codec, true);
@@ -371,7 +371,7 @@
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) {
RtpCapabilities video_capabilities =
- factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
+ factory_->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO);
const auto it = std::find_if(
video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
[](const auto& c) { return c.name == cricket::kRtxCodecName; });
@@ -381,7 +381,7 @@
TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) {
auto factory = CreatePeerConnectionFactoryWithRtxDisabled();
RtpCapabilities video_capabilities =
- factory->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
+ factory->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO);
const auto it = std::find_if(
video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
[](const auto& c) { return c.name == cricket::kRtxCodecName; });
@@ -390,14 +390,14 @@
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) {
RtpCapabilities data_capabilities =
- factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_DATA);
+ factory_->GetRtpSenderCapabilities(webrtc::MediaType::DATA);
EXPECT_TRUE(data_capabilities.codecs.empty());
EXPECT_TRUE(data_capabilities.header_extensions.empty());
}
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) {
RtpCapabilities audio_capabilities =
- factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO);
+ factory_->GetRtpReceiverCapabilities(webrtc::MediaType::AUDIO);
EXPECT_FALSE(audio_capabilities.codecs.empty());
for (const auto& codec : audio_capabilities.codecs) {
VerifyAudioCodecCapability(codec);
@@ -410,7 +410,7 @@
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) {
RtpCapabilities video_capabilities =
- factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
+ factory_->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO);
EXPECT_FALSE(video_capabilities.codecs.empty());
for (const auto& codec : video_capabilities.codecs) {
VerifyVideoCodecCapability(codec, false);
@@ -423,7 +423,7 @@
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverRtxEnabledCapabilities) {
RtpCapabilities video_capabilities =
- factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
+ factory_->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO);
const auto it = std::find_if(
video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
[](const auto& c) { return c.name == cricket::kRtxCodecName; });
@@ -434,7 +434,7 @@
CheckRtpReceiverRtxDisabledCapabilities) {
auto factory = CreatePeerConnectionFactoryWithRtxDisabled();
RtpCapabilities video_capabilities =
- factory->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
+ factory->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO);
const auto it = std::find_if(
video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
[](const auto& c) { return c.name == cricket::kRtxCodecName; });
@@ -443,7 +443,7 @@
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) {
RtpCapabilities data_capabilities =
- factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_DATA);
+ factory_->GetRtpReceiverCapabilities(webrtc::MediaType::DATA);
EXPECT_TRUE(data_capabilities.codecs.empty());
EXPECT_TRUE(data_capabilities.header_extensions.empty());
}
diff --git a/pc/peer_connection_field_trial_tests.cc b/pc/peer_connection_field_trial_tests.cc
index 0ff31ef..c76dbda 100644
--- a/pc/peer_connection_field_trial_tests.cc
+++ b/pc/peer_connection_field_trial_tests.cc
@@ -16,7 +16,6 @@
#include <utility>
#include "absl/algorithm/container.h"
-#include "api/create_peerconnection_factory.h"
#include "api/enable_media_with_defaults.h"
#include "api/field_trials.h"
#include "api/field_trials_view.h"
@@ -105,7 +104,7 @@
"WebRTC-DependencyDescriptorAdvertised/Enabled/"));
WrapperPtr caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto offer = caller->CreateOffer();
auto contents1 = offer->description()->contents();
@@ -113,7 +112,7 @@
const MediaContentDescription* media_description1 =
contents1[0].media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, media_description1->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, media_description1->type());
const cricket::RtpHeaderExtensions& rtp_header_extensions1 =
media_description1->rtp_header_extensions();
@@ -139,7 +138,7 @@
WrapperPtr caller = CreatePeerConnection();
WrapperPtr callee = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto offer = caller->CreateOffer();
cricket::ContentInfos& contents1 = offer->description()->contents();
@@ -147,7 +146,7 @@
MediaContentDescription* media_description1 =
contents1[0].media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, media_description1->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, media_description1->type());
cricket::RtpHeaderExtensions rtp_header_extensions1 =
media_description1->rtp_header_extensions();
@@ -191,7 +190,7 @@
MediaContentDescription* media_description2 =
contents2[0].media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, media_description2->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, media_description2->type());
cricket::RtpHeaderExtensions rtp_header_extensions2 =
media_description2->rtp_header_extensions();
diff --git a/pc/peer_connection_header_extension_unittest.cc b/pc/peer_connection_header_extension_unittest.cc
index 620ee63..f83a7c8 100644
--- a/pc/peer_connection_header_extension_unittest.cc
+++ b/pc/peer_connection_header_extension_unittest.cc
@@ -51,7 +51,7 @@
class PeerConnectionHeaderExtensionTest
: public ::testing::TestWithParam<
- std::tuple<cricket::MediaType, SdpSemantics>> {
+ std::tuple<webrtc::MediaType, SdpSemantics>> {
protected:
PeerConnectionHeaderExtensionTest()
: socket_server_(rtc::CreateDefaultSocketServer()),
@@ -72,10 +72,10 @@
RtpTransceiverDirection::kSendRecv)}) {}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
std::optional<SdpSemantics> semantics) {
auto media_engine = std::make_unique<cricket::FakeMediaEngine>();
- if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO)
+ if (media_type == webrtc::MediaType::AUDIO)
media_engine->fake_voice_engine()->SetRtpHeaderExtensions(extensions_);
else
media_engine->fake_video_engine()->SetRtpHeaderExtensions(extensions_);
@@ -117,7 +117,7 @@
};
TEST_P(PeerConnectionHeaderExtensionTest, TransceiverOffersHeaderExtensions) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -130,7 +130,7 @@
TEST_P(PeerConnectionHeaderExtensionTest,
SenderReceiverCapabilitiesReturnNotStoppedExtensions) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
std::unique_ptr<PeerConnectionWrapper> wrapper =
@@ -150,7 +150,7 @@
}
TEST_P(PeerConnectionHeaderExtensionTest, OffersUnstoppedDefaultExtensions) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -169,7 +169,7 @@
}
TEST_P(PeerConnectionHeaderExtensionTest, OffersUnstoppedModifiedExtensions) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -193,7 +193,7 @@
}
TEST_P(PeerConnectionHeaderExtensionTest, AnswersUnstoppedModifiedExtensions) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -226,7 +226,7 @@
}
TEST_P(PeerConnectionHeaderExtensionTest, NegotiatedExtensionsAreAccessible) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -262,7 +262,7 @@
}
TEST_P(PeerConnectionHeaderExtensionTest, OfferedExtensionsArePerTransceiver) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -292,7 +292,7 @@
}
TEST_P(PeerConnectionHeaderExtensionTest, RemovalAfterRenegotiation) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -324,7 +324,7 @@
TEST_P(PeerConnectionHeaderExtensionTest,
StoppedByDefaultExtensionCanBeActivatedByRemoteSdp) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -359,7 +359,7 @@
TEST_P(PeerConnectionHeaderExtensionTest,
UnknownExtensionInRemoteOfferDoesNotShowUp) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -376,7 +376,7 @@
"AD:7E:77:43:2A:29:EC:93\r\n"
"a=ice-ufrag:6HHHdzzeIhkE0CKj\r\n"
"a=ice-pwd:XYDGVpfvklQIEnZ6YnyLsAew\r\n";
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
sdp +=
"m=audio 9 RTP/AVPF 111\r\n"
"a=rtpmap:111 fake_audio_codec/8000\r\n";
@@ -413,7 +413,7 @@
// of the API to only offer non-stopped extensions.
TEST_P(PeerConnectionHeaderExtensionTest,
SdpMungingAnswerWithoutApiUsageEnablesExtensions) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -430,7 +430,7 @@
"AD:7E:77:43:2A:29:EC:93\r\n"
"a=ice-ufrag:6HHHdzzeIhkE0CKj\r\n"
"a=ice-pwd:XYDGVpfvklQIEnZ6YnyLsAew\r\n";
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
sdp +=
"m=audio 9 RTP/AVPF 111\r\n"
"a=rtpmap:111 fake_audio_codec/8000\r\n";
@@ -470,7 +470,7 @@
TEST_P(PeerConnectionHeaderExtensionTest,
SdpMungingOfferWithoutApiUsageEnablesExtensions) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -500,7 +500,7 @@
}
TEST_P(PeerConnectionHeaderExtensionTest, EnablingExtensionsAfterRemoteOffer) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = GetParam();
if (semantics != SdpSemantics::kUnifiedPlan)
@@ -517,7 +517,7 @@
"AD:7E:77:43:2A:29:EC:93\r\n"
"a=ice-ufrag:6HHHdzzeIhkE0CKj\r\n"
"a=ice-pwd:XYDGVpfvklQIEnZ6YnyLsAew\r\n";
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
sdp +=
"m=audio 9 RTP/AVPF 111\r\n"
"a=rtpmap:111 fake_audio_codec/8000\r\n";
@@ -562,19 +562,17 @@
,
PeerConnectionHeaderExtensionTest,
Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
- Values(cricket::MediaType::MEDIA_TYPE_AUDIO,
- cricket::MediaType::MEDIA_TYPE_VIDEO)),
+ Values(webrtc::MediaType::AUDIO, webrtc::MediaType::VIDEO)),
[](const testing::TestParamInfo<
PeerConnectionHeaderExtensionTest::ParamType>& info) {
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
SdpSemantics semantics;
std::tie(media_type, semantics) = info.param;
return (rtc::StringBuilder("With")
<< (semantics == SdpSemantics::kPlanB_DEPRECATED ? "PlanB"
: "UnifiedPlan")
<< "And"
- << (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO ? "Voice"
- : "Video")
+ << (media_type == webrtc::MediaType::AUDIO ? "Voice" : "Video")
<< "Engine")
.str();
});
diff --git a/pc/peer_connection_ice_unittest.cc b/pc/peer_connection_ice_unittest.cc
index 371b366..107d176 100644
--- a/pc/peer_connection_ice_unittest.cc
+++ b/pc/peer_connection_ice_unittest.cc
@@ -273,7 +273,7 @@
pc_wrapper_ptr->pc());
PeerConnection* pc = static_cast<PeerConnection*>(pc_proxy->internal());
for (const auto& transceiver : pc->GetTransceiversInternal()) {
- if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (transceiver->media_type() == webrtc::MediaType::AUDIO) {
auto dtls_transport = pc->LookupDtlsTransportByMidInternal(
transceiver->internal()->channel()->mid());
return dtls_transport->ice_transport()->internal()->GetIceRole();
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index 8ad11b9..47df968 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -621,7 +621,7 @@
} else {
callee()->SetRemoteOfferHandler([this] {
callee()
- ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
+ ->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO)
->StopInternal();
});
}
@@ -657,7 +657,7 @@
// the offer, but by default it is send only.
auto transceivers = caller()->pc()->GetTransceivers();
ASSERT_EQ(2U, transceivers.size());
- ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
+ ASSERT_EQ(webrtc::MediaType::VIDEO,
transceivers[1]->receiver()->media_type());
transceivers[1]->sender()->SetTrack(
caller()->CreateLocalVideoTrack().get());
@@ -842,7 +842,7 @@
// rejected in the answer.
callee()->SetRemoteOfferHandler([this] {
callee()
- ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
+ ->GetFirstTransceiverOfType(webrtc::MediaType::AUDIO)
->StopInternal();
});
}
@@ -867,7 +867,7 @@
// The caller's transceiver should have stopped after receiving the answer,
// and thus no longer listed in transceivers.
EXPECT_EQ(nullptr,
- caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO));
+ caller()->GetFirstTransceiverOfType(webrtc::MediaType::AUDIO));
}
}
@@ -888,7 +888,7 @@
// rejected in the answer.
callee()->SetRemoteOfferHandler([this] {
callee()
- ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
+ ->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO)
->StopInternal();
});
}
@@ -913,7 +913,7 @@
// The caller's transceiver should have stopped after receiving the answer,
// and thus is no longer present.
EXPECT_EQ(nullptr,
- caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO));
+ caller()->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO));
}
}
@@ -989,7 +989,7 @@
});
} else {
caller()
- ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
+ ->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO)
->StopInternal();
}
caller()->CreateAndSetAndSignalOffer();
@@ -2332,7 +2332,7 @@
} else {
callee()->SetRemoteOfferHandler([this] {
callee()
- ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
+ ->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO)
->StopInternal();
});
}
@@ -2355,7 +2355,7 @@
} else {
// The caller's transceiver is stopped, so we need to add another track.
auto caller_transceiver =
- caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
+ caller()->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO);
EXPECT_EQ(nullptr, caller_transceiver.get());
caller()->AddVideoTrack();
}
@@ -2426,10 +2426,10 @@
MediaFlowsAfterEarlyWarmupWithAddTransceiver) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- auto audio_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto audio_result = caller()->pc()->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_EQ(RTCErrorType::NONE, audio_result.error().type());
auto caller_audio_sender = audio_result.MoveValue()->sender();
- auto video_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto video_result = caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
ASSERT_EQ(RTCErrorType::NONE, video_result.error().type());
auto caller_video_sender = video_result.MoveValue()->sender();
callee()->SetRemoteOfferHandler([this] {
@@ -2924,7 +2924,7 @@
ASSERT_TRUE(ExpectNewFrames(media_expectations));
ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
auto receiver = callee()->pc()->GetReceivers()[0];
- ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
+ ASSERT_EQ(receiver->media_type(), webrtc::MediaType::AUDIO);
auto sources = receiver->GetSources();
ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
@@ -2947,7 +2947,7 @@
ASSERT_TRUE(ExpectNewFrames(media_expectations));
ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
auto receiver = callee()->pc()->GetReceivers()[0];
- ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_VIDEO);
+ ASSERT_EQ(receiver->media_type(), webrtc::MediaType::VIDEO);
auto sources = receiver->GetSources();
ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
ASSERT_GT(sources.size(), 0u);
@@ -3665,7 +3665,7 @@
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
- caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller()->pc()->AddTransceiver(webrtc::MediaType::AUDIO);
caller()->CreateAndSetAndSignalOffer();
ASSERT_THAT(
@@ -3679,7 +3679,7 @@
while (current_size < 8) {
// Double the number of tracks
for (int i = 0; i < current_size; i++) {
- caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller()->pc()->AddTransceiver(webrtc::MediaType::AUDIO);
}
current_size = caller()->pc()->GetTransceivers().size();
RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
@@ -3703,7 +3703,7 @@
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
- caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
caller()->CreateAndSetAndSignalOffer();
ASSERT_THAT(
@@ -3719,7 +3719,7 @@
while (current_size < 8) {
// Double the number of tracks
for (int i = 0; i < current_size; i++) {
- caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
}
current_size = caller()->pc()->GetTransceivers().size();
RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
@@ -3765,7 +3765,7 @@
while (current_size < 16) {
// Double the number of tracks
for (int i = 0; i < current_size; i++) {
- caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
}
current_size = caller()->pc()->GetTransceivers().size();
RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
@@ -3789,7 +3789,7 @@
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
GetParametersHasEncodingsBeforeNegotiation) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
- auto result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto result = caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
auto transceiver = result.MoveValue();
auto parameters = transceiver->sender()->GetParameters();
EXPECT_EQ(parameters.encodings.size(), 1u);
@@ -3801,7 +3801,7 @@
RtpTransceiverInit init;
init.send_encodings.push_back({});
init.send_encodings[0].max_bitrate_bps = 12345;
- auto result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ auto result = caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init);
auto transceiver = result.MoveValue();
auto parameters = transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
@@ -3866,7 +3866,7 @@
// has the same track ID as the sending track.
auto receivers = callee()->pc()->GetReceivers();
ASSERT_EQ(1u, receivers.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, receivers[0]->media_type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, receivers[0]->media_type());
EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id());
MediaExpectations media_expectations;
@@ -3887,12 +3887,10 @@
// Verify that one audio and one video receiver have been created on the
// remote and that they have the same track IDs as the sending tracks.
- auto audio_receivers =
- callee()->GetReceiversOfType(cricket::MEDIA_TYPE_AUDIO);
+ auto audio_receivers = callee()->GetReceiversOfType(webrtc::MediaType::AUDIO);
ASSERT_EQ(1u, audio_receivers.size());
EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id());
- auto video_receivers =
- callee()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO);
+ auto video_receivers = callee()->GetReceiversOfType(webrtc::MediaType::VIDEO);
ASSERT_EQ(1u, video_receivers.size());
EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id());
@@ -3931,7 +3929,7 @@
IsRtcOk());
// Verify that only the audio track has been negotiated.
- EXPECT_EQ(0u, caller()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO).size());
+ EXPECT_EQ(0u, caller()->GetReceiversOfType(webrtc::MediaType::VIDEO).size());
// Might also check that the callee's NegotiationNeeded flag is set.
// Reverse roles.
diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc
index 4f66d0c..0cb713a 100644
--- a/pc/peer_connection_interface_unittest.cc
+++ b/pc/peer_connection_interface_unittest.cc
@@ -843,7 +843,7 @@
}
rtc::scoped_refptr<RtpReceiverInterface> GetFirstReceiverOfType(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
for (auto receiver : pc_->GetReceivers()) {
if (receiver->media_type() == media_type) {
return receiver;
@@ -1931,7 +1931,7 @@
// Test that we can get stats on a video track.
TEST_P(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
InitiateCall();
- auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
+ auto video_receiver = GetFirstReceiverOfType(webrtc::MediaType::VIDEO);
ASSERT_TRUE(video_receiver);
EXPECT_TRUE(DoGetStats(video_receiver->track().get()));
}
@@ -2457,8 +2457,8 @@
EXPECT_EQ(2u, pc_->GetTransceivers().size());
}
- auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
- auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
+ auto audio_receiver = GetFirstReceiverOfType(webrtc::MediaType::AUDIO);
+ auto video_receiver = GetFirstReceiverOfType(webrtc::MediaType::VIDEO);
if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
ASSERT_TRUE(audio_receiver);
ASSERT_TRUE(video_receiver);
@@ -2608,9 +2608,9 @@
// First create and set a remote offer, then reject its video content in our
// answer.
CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
- auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
+ auto audio_receiver = GetFirstReceiverOfType(webrtc::MediaType::AUDIO);
ASSERT_TRUE(audio_receiver);
- auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
+ auto video_receiver = GetFirstReceiverOfType(webrtc::MediaType::VIDEO);
ASSERT_TRUE(video_receiver);
rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
@@ -3719,7 +3719,7 @@
std::vector<rtc::scoped_refptr<RtpSenderInterface>> rtp_senders =
pc_->GetSenders();
ASSERT_EQ(rtp_senders.size(), 1u);
- ASSERT_EQ(rtp_senders[0]->media_type(), cricket::MEDIA_TYPE_VIDEO);
+ ASSERT_EQ(rtp_senders[0]->media_type(), webrtc::MediaType::VIDEO);
rtc::scoped_refptr<RtpSenderInterface> video_rtp_sender = rtp_senders[0];
RtpParameters parameters = video_rtp_sender->GetParameters();
ASSERT_NE(parameters.degradation_preference,
diff --git a/pc/peer_connection_internal.h b/pc/peer_connection_internal.h
index 3f38b58..e053ba9 100644
--- a/pc/peer_connection_internal.h
+++ b/pc/peer_connection_internal.h
@@ -119,7 +119,7 @@
// Internal implementation for AddTransceiver family of methods. If
// `fire_callback` is set, fires OnRenegotiationNeeded callback if successful.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
- AddTransceiver(cricket::MediaType media_type,
+ AddTransceiver(webrtc::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool fire_callback = true) = 0;
diff --git a/pc/peer_connection_jsep_unittest.cc b/pc/peer_connection_jsep_unittest.cc
index 7a4df54..c1f5b42 100644
--- a/pc/peer_connection_jsep_unittest.cc
+++ b/pc/peer_connection_jsep_unittest.cc
@@ -37,7 +37,6 @@
#include "media/base/stream_params.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/transport_info.h"
-#include "pc/channel_interface.h"
#include "pc/media_session.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/sdp_utils.h"
@@ -133,24 +132,24 @@
// section.
TEST_F(PeerConnectionJsepTest, AudioOnlyInitialOffer) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto offer = caller->CreateOffer();
auto contents = offer->description()->contents();
ASSERT_EQ(1u, contents.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, contents[0].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, contents[0].media_description()->type());
}
// Test than an initial offer with one video track generates one video media
// section
TEST_F(PeerConnectionJsepTest, VideoOnlyInitialOffer) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto offer = caller->CreateOffer();
auto contents = offer->description()->contents();
ASSERT_EQ(1u, contents.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, contents[0].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, contents[0].media_description()->type());
}
// Test that an initial offer with one data channel generates one data media
@@ -162,7 +161,7 @@
auto offer = caller->CreateOffer();
auto contents = offer->description()->contents();
ASSERT_EQ(1u, contents.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_DATA, contents[0].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::DATA, contents[0].media_description()->type());
}
// Test that creating multiple data channels only results in one data section
@@ -182,11 +181,11 @@
// JSEP section 5.2.1.
TEST_F(PeerConnectionJsepTest, MediaSectionsInInitialOfferOrderedCorrectly) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kSendOnly;
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO, init);
auto offer = caller->CreateOffer();
auto contents = offer->description()->contents();
@@ -194,19 +193,19 @@
const MediaContentDescription* media_description1 =
contents[0].media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, media_description1->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, media_description1->type());
EXPECT_EQ(RtpTransceiverDirection::kSendRecv,
media_description1->direction());
const MediaContentDescription* media_description2 =
contents[1].media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, media_description2->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, media_description2->type());
EXPECT_EQ(RtpTransceiverDirection::kSendRecv,
media_description2->direction());
const MediaContentDescription* media_description3 =
contents[2].media_description();
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, media_description3->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, media_description3->type());
EXPECT_EQ(RtpTransceiverDirection::kSendOnly,
media_description3->direction());
}
@@ -214,8 +213,8 @@
// Test that media sections in the initial offer have different mids.
TEST_F(PeerConnectionJsepTest, MediaSectionsInInitialOfferHaveDifferentMids) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto offer = caller->CreateOffer();
auto contents = offer->description()->contents();
@@ -226,7 +225,7 @@
TEST_F(PeerConnectionJsepTest,
StoppedTransceiverHasNoMediaSectionInInitialOffer) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
transceiver->StopInternal();
auto offer = caller->CreateOffer();
@@ -247,8 +246,8 @@
TEST_F(PeerConnectionJsepTest, SetLocalOfferSetsTransceiverMid) {
auto caller = CreatePeerConnection();
- auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto offer = caller->CreateOffer();
auto audio_mid = offer->description()->contents()[0].mid();
@@ -266,8 +265,8 @@
// transceivers, one for receiving audio and one for receiving video.
TEST_F(PeerConnectionJsepTest, SetRemoteOfferCreatesTransceivers) {
auto caller = CreatePeerConnection();
- auto caller_audio = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- auto caller_video = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto caller_audio = caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ auto caller_video = caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto callee = CreatePeerConnection();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -275,12 +274,12 @@
auto transceivers = callee->pc()->GetTransceivers();
ASSERT_EQ(2u, transceivers.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceivers[0]->media_type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, transceivers[0]->media_type());
EXPECT_EQ(caller_audio->mid(), transceivers[0]->mid());
EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceivers[0]->direction());
EXPECT_EQ(0u, transceivers[0]->sender()->stream_ids().size());
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceivers[1]->media_type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, transceivers[1]->media_type());
EXPECT_EQ(caller_video->mid(), transceivers[1]->mid());
EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceivers[1]->direction());
EXPECT_EQ(0u, transceivers[1]->sender()->stream_ids().size());
@@ -335,7 +334,7 @@
auto caller = CreatePeerConnection();
caller->AddAudioTrack("a");
auto callee = CreatePeerConnection();
- auto transceiver = callee->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = callee->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -415,9 +414,9 @@
// offered media in the same order and with the same mids.
TEST_F(PeerConnectionJsepTest, CreateAnswerHasSameMidsAsOffer) {
auto caller = CreatePeerConnection();
- auto first_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
- auto second_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- auto third_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto first_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
+ auto second_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ auto third_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
caller->CreateDataChannel("dc");
auto callee = CreatePeerConnection();
@@ -430,13 +429,13 @@
auto answer = callee->CreateAnswer();
auto contents = answer->description()->contents();
ASSERT_EQ(4u, contents.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, contents[0].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, contents[0].media_description()->type());
EXPECT_EQ(first_transceiver->mid(), contents[0].mid());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, contents[1].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, contents[1].media_description()->type());
EXPECT_EQ(second_transceiver->mid(), contents[1].mid());
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, contents[2].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, contents[2].media_description()->type());
EXPECT_EQ(third_transceiver->mid(), contents[2].mid());
- EXPECT_EQ(cricket::MEDIA_TYPE_DATA, contents[3].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::DATA, contents[3].media_description()->type());
EXPECT_EQ(offer_data->mid(), contents[3].mid());
}
@@ -464,7 +463,7 @@
auto caller = CreatePeerConnection();
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kSendOnly;
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO, init);
auto callee = CreatePeerConnection();
callee->AddAudioTrack("a");
@@ -485,7 +484,7 @@
// property of the transceivers mentioned in the session description.
TEST_F(PeerConnectionJsepTest, SetLocalAnswerUpdatesCurrentDirection) {
auto caller = CreatePeerConnection();
- auto caller_audio = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto caller_audio = caller->AddTransceiver(webrtc::MediaType::AUDIO);
caller_audio->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly);
auto callee = CreatePeerConnection();
callee->AddAudioTrack("a");
@@ -529,7 +528,7 @@
TEST_F(PeerConnectionJsepTest,
ChangeDirectionFromRecvOnlyToSendRecvDoesNotBreakVideoNegotiation) {
auto caller = CreatePeerConnection();
- auto caller_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto callee = CreatePeerConnection();
caller_transceiver->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly);
@@ -547,7 +546,7 @@
TEST_F(PeerConnectionJsepTest,
ChangeDirectionFromRecvOnlyToSendRecvDoesNotBreakAudioNegotiation) {
auto caller = CreatePeerConnection();
- auto caller_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
caller_transceiver->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly);
@@ -588,7 +587,7 @@
TEST_F(PeerConnectionJsepTest,
ReOfferMediaSectionForAssociatedStoppedTransceiverIsRejected) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -610,7 +609,7 @@
TEST_F(PeerConnectionJsepTest,
StoppingTransceiverInOfferStopsTransceiverOnRemoteSide) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -633,14 +632,14 @@
TEST_F(PeerConnectionJsepTest,
CreateOfferDoesNotRecycleMediaSectionIfFirstStopped) {
auto caller = CreatePeerConnection();
- auto first_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto first_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
ASSERT_TRUE(
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
- auto second_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto second_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
first_transceiver->StopInternal();
auto reoffer = caller->CreateOffer();
@@ -656,7 +655,7 @@
TEST_F(PeerConnectionJsepTest,
RecycleMediaSectionWhenStoppingTransceiverOnAnswerer) {
auto caller = CreatePeerConnection();
- auto first_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto first_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -718,7 +717,7 @@
TEST_F(PeerConnectionJsepTest, CreateOfferRecyclesWhenOfferingTwice) {
// Do a negotiation with a port 0 for the media section.
auto caller = CreatePeerConnection();
- auto first_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto first_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
first_transceiver->StopInternal();
@@ -765,15 +764,15 @@
class RecycleMediaSectionTest
: public PeerConnectionJsepTest,
public ::testing::WithParamInterface<
- std::tuple<cricket::MediaType, cricket::MediaType>> {
+ std::tuple<webrtc::MediaType, webrtc::MediaType>> {
protected:
RecycleMediaSectionTest() {
first_type_ = std::get<0>(GetParam());
second_type_ = std::get<1>(GetParam());
}
- cricket::MediaType first_type_;
- cricket::MediaType second_type_;
+ webrtc::MediaType first_type_;
+ webrtc::MediaType second_type_;
};
// Test that recycling works properly when a new transceiver recycles an m=
@@ -1118,15 +1117,15 @@
INSTANTIATE_TEST_SUITE_P(
PeerConnectionJsepTest,
RecycleMediaSectionTest,
- Combine(Values(cricket::MEDIA_TYPE_AUDIO, cricket::MEDIA_TYPE_VIDEO),
- Values(cricket::MEDIA_TYPE_AUDIO, cricket::MEDIA_TYPE_VIDEO)));
+ Combine(Values(webrtc::MediaType::AUDIO, webrtc::MediaType::VIDEO),
+ Values(webrtc::MediaType::AUDIO, webrtc::MediaType::VIDEO)));
// Test that a new data channel section will not reuse a recycleable audio or
// video media section. Additionally, tests that the new section is added to the
// end of the session description.
TEST_F(PeerConnectionJsepTest, DataChannelDoesNotRecycleMediaSection) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
@@ -1140,9 +1139,9 @@
auto offer = caller->CreateOffer();
auto offer_contents = offer->description()->contents();
ASSERT_EQ(2u, offer_contents.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO,
+ EXPECT_EQ(webrtc::MediaType::AUDIO,
offer_contents[0].media_description()->type());
- EXPECT_EQ(cricket::MEDIA_TYPE_DATA,
+ EXPECT_EQ(webrtc::MediaType::DATA,
offer_contents[1].media_description()->type());
ASSERT_TRUE(
@@ -1152,9 +1151,9 @@
auto answer = callee->CreateAnswer();
auto answer_contents = answer->description()->contents();
ASSERT_EQ(2u, answer_contents.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO,
+ EXPECT_EQ(webrtc::MediaType::AUDIO,
answer_contents[0].media_description()->type());
- EXPECT_EQ(cricket::MEDIA_TYPE_DATA,
+ EXPECT_EQ(webrtc::MediaType::DATA,
answer_contents[1].media_description()->type());
}
@@ -1173,8 +1172,8 @@
auto offer = caller->CreateOffer();
auto contents = offer->description()->contents();
ASSERT_EQ(2u, contents.size());
- EXPECT_EQ(cricket::MEDIA_TYPE_DATA, contents[0].media_description()->type());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, contents[1].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::DATA, contents[0].media_description()->type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, contents[1].media_description()->type());
}
// Tests for MID properties.
@@ -1345,7 +1344,7 @@
const std::string kTrackId = "audio_track";
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
transceiver->sender()->SetTrack(caller->CreateAudioTrack(kTrackId).get());
auto offer = caller->CreateOffer();
@@ -1363,11 +1362,11 @@
RtpTransceiverInit init_recvonly;
init_recvonly.direction = RtpTransceiverDirection::kRecvOnly;
- ASSERT_TRUE(caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init_recvonly));
+ ASSERT_TRUE(caller->AddTransceiver(webrtc::MediaType::AUDIO, init_recvonly));
RtpTransceiverInit init_inactive;
init_inactive.direction = RtpTransceiverDirection::kInactive;
- ASSERT_TRUE(caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init_inactive));
+ ASSERT_TRUE(caller->AddTransceiver(webrtc::MediaType::VIDEO, init_inactive));
auto offer = caller->CreateOffer();
auto contents = offer->description()->contents();
@@ -1387,13 +1386,13 @@
// no tracks to send in response.
RtpTransceiverInit init_recvonly;
init_recvonly.direction = RtpTransceiverDirection::kRecvOnly;
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init_recvonly);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO, init_recvonly);
// sendrecv transceiver will get negotiated to recvonly since the callee has
// no tracks to send in response.
RtpTransceiverInit init_sendrecv;
init_sendrecv.direction = RtpTransceiverDirection::kSendRecv;
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init_sendrecv);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO, init_sendrecv);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -1438,7 +1437,7 @@
// any MSID information for that section.
TEST_F(PeerConnectionJsepTest, RemoveMsidIfTransceiverStopped) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
@@ -1595,7 +1594,7 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
@@ -1907,7 +1906,7 @@
TEST_F(PeerConnectionJsepTest, RollbackRemovesTransceiver) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
ASSERT_EQ(callee->pc()->GetTransceivers().size(), 1u);
@@ -1925,7 +1924,7 @@
TEST_F(PeerConnectionJsepTest, RollbackKeepsTransceiverAndClearsMid) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
callee->AddAudioTrack("a");
@@ -1951,7 +1950,7 @@
TEST_F(PeerConnectionJsepTest,
RollbackKeepsTransceiverAfterAddTrackEvenWhenTrackIsNulled) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
callee->AddAudioTrack("a");
@@ -1971,7 +1970,7 @@
TEST_F(PeerConnectionJsepTest, RollbackRestoresMid) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
callee->AddAudioTrack("a");
auto offer = callee->CreateOffer();
@@ -1994,7 +1993,7 @@
init.send_encodings.push_back(encoding);
encoding.rid = "lo";
init.send_encodings.push_back(encoding);
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO, init);
auto encodings =
caller->pc()->GetTransceivers()[0]->sender()->init_send_encodings();
EXPECT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
@@ -2017,8 +2016,8 @@
init.send_encodings.push_back(encoding);
encoding.rid = "lo";
init.send_encodings.push_back(encoding);
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
- callee->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO, init);
+ callee->AddTransceiver(webrtc::MediaType::VIDEO);
callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal());
caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal());
auto params = caller->pc()->GetTransceivers()[0]->sender()->GetParameters();
@@ -2059,7 +2058,7 @@
EXPECT_EQ(callee->pc()->GetTransceivers().size(), 1u);
EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), mid);
EXPECT_EQ(callee->pc()->GetTransceivers()[0]->media_type(),
- cricket::MEDIA_TYPE_VIDEO);
+ webrtc::MediaType::VIDEO);
EXPECT_TRUE(callee->SetLocalDescription(std::move(offer)));
EXPECT_EQ(callee->observer()->remove_track_events_.size(),
callee->observer()->add_track_events_.size());
@@ -2095,9 +2094,9 @@
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
config.enable_implicit_rollback = true;
auto caller = CreatePeerConnection(config);
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto callee = CreatePeerConnection(config);
- callee->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ callee->AddTransceiver(webrtc::MediaType::VIDEO);
EXPECT_TRUE(callee->CreateOfferAndSetAsLocal());
auto initial_mid = callee->pc()->GetTransceivers()[0]->mid();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -2115,7 +2114,7 @@
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
auto caller = CreatePeerConnection(config);
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection(config);
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
EXPECT_TRUE(callee->CreateAnswerAndSetAsLocal());
@@ -2166,7 +2165,7 @@
TEST_F(PeerConnectionJsepTest, RollbackLocalDirectionChange) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
EXPECT_TRUE(
@@ -2189,7 +2188,7 @@
TEST_F(PeerConnectionJsepTest, RollbackRemoteDirectionChange) {
auto caller = CreatePeerConnection();
- auto caller_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
callee->AddAudioTrack("a");
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -2219,7 +2218,7 @@
TEST_F(PeerConnectionJsepTest,
RollbackRestoresFiredDirectionAndOnTrackCanFireAgain) {
auto caller = CreatePeerConnection();
- auto caller_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
callee->AddAudioTrack("a");
ASSERT_EQ(callee->pc()->GetTransceivers().size(), 1u);
@@ -2249,7 +2248,7 @@
TEST_F(PeerConnectionJsepTest,
RollbackFromInactiveToReceivingMakesOnTrackFire) {
auto caller = CreatePeerConnection();
- auto caller_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
// Perform full O/A so that transceiver is associated. Ontrack fires.
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
@@ -2272,9 +2271,9 @@
TEST_F(PeerConnectionJsepTest, RollbackAfterMultipleSLD) {
auto callee = CreatePeerConnection();
- callee->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ callee->AddTransceiver(webrtc::MediaType::AUDIO);
EXPECT_TRUE(callee->CreateOfferAndSetAsLocal());
- callee->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ callee->AddTransceiver(webrtc::MediaType::VIDEO);
EXPECT_TRUE(callee->CreateOfferAndSetAsLocal());
callee->observer()->clear_legacy_renegotiation_needed();
callee->observer()->clear_latest_negotiation_needed_event();
@@ -2288,9 +2287,9 @@
TEST_F(PeerConnectionJsepTest, NoRollbackNeeded) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
- callee->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ callee->AddTransceiver(webrtc::MediaType::AUDIO);
EXPECT_TRUE(caller->CreateOfferAndSetAsLocal());
EXPECT_TRUE(caller->CreateOfferAndSetAsLocal());
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
@@ -2323,7 +2322,7 @@
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
config.enable_implicit_rollback = true;
auto caller = CreatePeerConnection(config);
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto callee = CreatePeerConnection(config);
callee->CreateDataChannel("dummy");
EXPECT_TRUE(callee->CreateOfferAndSetAsLocal());
@@ -2340,7 +2339,7 @@
caller->CreateDataChannel("dummy");
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
EXPECT_TRUE(callee->SetRemoteDescription(callee->CreateRollback()));
- callee->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ callee->AddTransceiver(webrtc::MediaType::VIDEO);
EXPECT_TRUE(callee->CreateOfferAndSetAsLocal());
}
@@ -2351,7 +2350,7 @@
caller->CreateDataChannel("dummy");
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
EXPECT_TRUE(callee->SetRemoteDescription(callee->CreateRollback()));
- callee->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ callee->AddTransceiver(webrtc::MediaType::VIDEO);
callee->CreateDataChannel("dummy");
EXPECT_TRUE(callee->CreateOfferAndSetAsLocal());
}
@@ -2369,7 +2368,7 @@
TEST_F(PeerConnectionJsepTest, RollbackRemoteTransceiverThenAddDataChannel) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
EXPECT_TRUE(callee->SetRemoteDescription(callee->CreateRollback()));
callee->CreateDataChannel("dummy");
@@ -2380,19 +2379,19 @@
RollbackRemoteTransceiverThenAddDataChannelAndTransceiver) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
EXPECT_TRUE(callee->SetRemoteDescription(callee->CreateRollback()));
callee->CreateDataChannel("dummy");
- callee->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ callee->AddTransceiver(webrtc::MediaType::VIDEO);
EXPECT_TRUE(callee->CreateOfferAndSetAsLocal());
}
TEST_F(PeerConnectionJsepTest, BundleOnlySectionDoesNotNeedRtcpMux) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto offer = caller->CreateOffer();
// Remove rtcp-mux and set bundle-only on the second content.
offer->description()->contents()[1].media_description()->set_rtcp_mux(false);
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index 686cf8a..cbcc1c0 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -239,7 +239,7 @@
RtpTransceiverDirection GetMediaContentDirection(
const SessionDescriptionInterface* sdesc,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
auto* content = GetFirstMediaContent(sdesc->description(), media_type);
RTC_DCHECK(content);
return content->media_description()->direction();
@@ -394,7 +394,7 @@
options.num_simulcast_layers = 3;
auto offer = caller->CreateOffer(options);
auto* description =
- GetFirstMediaContent(offer->description(), cricket::MEDIA_TYPE_VIDEO)
+ GetFirstMediaContent(offer->description(), webrtc::MediaType::VIDEO)
->media_description();
ASSERT_EQ(1u, description->streams().size());
ASSERT_TRUE(description->streams()[0].get_ssrc_group("SIM"));
@@ -404,7 +404,7 @@
caller->SetLocalDescription(std::move(offer));
auto senders = caller->pc()->GetSenders();
ASSERT_EQ(1u, senders.size());
- EXPECT_EQ(cricket::MediaType::MEDIA_TYPE_VIDEO, senders[0]->media_type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, senders[0]->media_type());
EXPECT_EQ(3u, senders[0]->GetParameters().encodings.size());
}
@@ -421,7 +421,7 @@
options.num_simulcast_layers = 3;
auto answer = callee->CreateAnswer(options);
auto* description =
- GetFirstMediaContent(answer->description(), cricket::MEDIA_TYPE_VIDEO)
+ GetFirstMediaContent(answer->description(), webrtc::MediaType::VIDEO)
->media_description();
ASSERT_EQ(1u, description->streams().size());
ASSERT_TRUE(description->streams()[0].get_ssrc_group("SIM"));
@@ -431,7 +431,7 @@
callee->SetLocalDescription(std::move(answer));
auto senders = callee->pc()->GetSenders();
ASSERT_EQ(1u, senders.size());
- EXPECT_EQ(cricket::MediaType::MEDIA_TYPE_VIDEO, senders[0]->media_type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, senders[0]->media_type());
EXPECT_EQ(3u, senders[0]->GetParameters().encodings.size());
}
@@ -699,7 +699,7 @@
auto offer = caller->CreateOffer(options);
auto* content =
- GetFirstMediaContent(offer->description(), cricket::MEDIA_TYPE_AUDIO);
+ GetFirstMediaContent(offer->description(), webrtc::MediaType::AUDIO);
if (expected_direction_ == RtpTransceiverDirection::kInactive) {
EXPECT_FALSE(content);
} else {
@@ -785,7 +785,7 @@
auto expected_direction =
RtpTransceiverDirectionFromSendRecv(negotiate_send, negotiate_recv);
EXPECT_EQ(expected_direction,
- GetMediaContentDirection(answer.get(), cricket::MEDIA_TYPE_AUDIO));
+ GetMediaContentDirection(answer.get(), webrtc::MediaType::AUDIO));
}
// Tests that the media section is rejected if and only if the callee has no
@@ -846,9 +846,9 @@
auto offer = caller->CreateOffer(options);
EXPECT_EQ(RtpTransceiverDirection::kRecvOnly,
- GetMediaContentDirection(offer.get(), cricket::MEDIA_TYPE_AUDIO));
+ GetMediaContentDirection(offer.get(), webrtc::MediaType::AUDIO));
EXPECT_EQ(RtpTransceiverDirection::kSendOnly,
- GetMediaContentDirection(offer.get(), cricket::MEDIA_TYPE_VIDEO));
+ GetMediaContentDirection(offer.get(), webrtc::MediaType::VIDEO));
}
TEST_P(PeerConnectionMediaTest, AnswerHasDifferentDirectionsForAudioVideo) {
@@ -870,9 +870,9 @@
auto answer = callee->CreateAnswer(options);
EXPECT_EQ(RtpTransceiverDirection::kRecvOnly,
- GetMediaContentDirection(answer.get(), cricket::MEDIA_TYPE_AUDIO));
+ GetMediaContentDirection(answer.get(), webrtc::MediaType::AUDIO));
EXPECT_EQ(RtpTransceiverDirection::kSendOnly,
- GetMediaContentDirection(answer.get(), cricket::MEDIA_TYPE_VIDEO));
+ GetMediaContentDirection(answer.get(), webrtc::MediaType::VIDEO));
}
void AddComfortNoiseCodecsToSend(cricket::FakeMediaEngine* media_engine) {
@@ -1134,7 +1134,7 @@
}
void RenameContent(SessionDescription* desc,
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::string& new_name) {
auto* content = GetFirstMediaContent(desc, media_type);
RTC_DCHECK(content);
@@ -1162,8 +1162,8 @@
auto callee = CreatePeerConnectionWithAudioVideo();
auto offer = caller->CreateOffer();
- RenameContent(offer->description(), cricket::MEDIA_TYPE_AUDIO, kAudioMid);
- RenameContent(offer->description(), cricket::MEDIA_TYPE_VIDEO, kVideoMid);
+ RenameContent(offer->description(), webrtc::MediaType::AUDIO, kAudioMid);
+ RenameContent(offer->description(), webrtc::MediaType::VIDEO, kVideoMid);
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
auto answer = callee->CreateAnswer();
@@ -1181,8 +1181,8 @@
auto callee = CreatePeerConnectionWithAudioVideo();
auto offer = caller->CreateOffer();
- RenameContent(offer->description(), cricket::MEDIA_TYPE_AUDIO, kAudioMid);
- RenameContent(offer->description(), cricket::MEDIA_TYPE_VIDEO, kVideoMid);
+ RenameContent(offer->description(), webrtc::MediaType::AUDIO, kAudioMid);
+ RenameContent(offer->description(), webrtc::MediaType::VIDEO, kVideoMid);
ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
@@ -1198,8 +1198,8 @@
auto callee = CreatePeerConnectionWithAudioVideo();
auto offer = caller->CreateOffer();
- RenameContent(offer->description(), cricket::MEDIA_TYPE_AUDIO, "same");
- RenameContent(offer->description(), cricket::MEDIA_TYPE_VIDEO, "same");
+ RenameContent(offer->description(), webrtc::MediaType::AUDIO, "same");
+ RenameContent(offer->description(), webrtc::MediaType::VIDEO, "same");
std::string error;
EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer), &error));
@@ -1500,8 +1500,8 @@
auto caller = CreatePeerConnectionWithAudio(std::move(fake_engine));
auto transceiver = caller->pc()->GetTransceivers().front();
- auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_AUDIO);
+ auto capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO);
std::vector<RtpCodecCapability> codecs;
absl::c_copy_if(capabilities.codecs, std::back_inserter(codecs),
@@ -1519,14 +1519,12 @@
auto caller = CreatePeerConnectionWithAudio();
auto transceiver = caller->pc()->GetTransceivers().front();
- auto video_codecs =
- caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO)
- .codecs;
- auto codecs =
- caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_AUDIO)
- .codecs;
+ auto video_codecs = caller->pc_factory()
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
+ .codecs;
+ auto codecs = caller->pc_factory()
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
+ .codecs;
codecs.insert(codecs.end(), video_codecs.begin(), video_codecs.end());
auto result = transceiver->SetCodecPreferences(codecs);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
@@ -1548,10 +1546,9 @@
auto transceiver =
RtpTransceiverInternal(caller->pc()->GetTransceivers().front());
- auto codecs =
- caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_AUDIO)
- .codecs;
+ auto codecs = caller->pc_factory()
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
+ .codecs;
auto codecs_only_rtx_red_fec = codecs;
auto it = std::remove_if(codecs_only_rtx_red_fec.begin(),
codecs_only_rtx_red_fec.end(),
@@ -1571,7 +1568,7 @@
auto sender_audio_codecs =
caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
.codecs;
auto audio_transceiver = caller->pc()->GetTransceivers().front();
@@ -1590,7 +1587,7 @@
auto sender_audio_codecs =
caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
.codecs;
std::vector<RtpCodecCapability> empty_codecs = {};
@@ -1609,14 +1606,12 @@
auto caller = CreatePeerConnectionWithVideo();
auto transceiver = caller->pc()->GetTransceivers().front();
- auto audio_codecs =
- caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_AUDIO)
- .codecs;
- auto codecs =
- caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO)
- .codecs;
+ auto audio_codecs = caller->pc_factory()
+ ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO)
+ .codecs;
+ auto codecs = caller->pc_factory()
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
+ .codecs;
codecs.insert(codecs.end(), audio_codecs.begin(), audio_codecs.end());
auto result = transceiver->SetCodecPreferences(codecs);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
@@ -1637,10 +1632,9 @@
auto caller = CreatePeerConnectionWithVideo(std::move(fake_engine));
auto transceiver = caller->pc()->GetTransceivers().front();
- auto codecs =
- caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO)
- .codecs;
+ auto codecs = caller->pc_factory()
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
+ .codecs;
auto codecs_only_rtx_red_fec = codecs;
auto it = std::remove_if(codecs_only_rtx_red_fec.begin(),
codecs_only_rtx_red_fec.end(),
@@ -1660,7 +1654,7 @@
auto sender_video_codecs =
caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
auto video_transceiver = caller->pc()->GetTransceivers().front();
@@ -1679,7 +1673,7 @@
auto sender_video_codecs =
caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
std::vector<RtpCodecCapability> empty_codecs = {};
@@ -1700,7 +1694,7 @@
auto sender_video_codecs =
caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
auto video_transceiver = caller->pc()->GetTransceivers().front();
@@ -1737,7 +1731,7 @@
auto sender_video_codecs =
caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
auto video_transceiver = caller->pc()->GetTransceivers().front();
@@ -1785,7 +1779,7 @@
auto callee = CreatePeerConnection(std::move(callee_fake_engine));
auto video_codecs = caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
auto send_transceiver = caller->pc()->GetTransceivers().front();
@@ -1849,7 +1843,7 @@
auto callee = CreatePeerConnection(std::move(callee_fake_engine));
auto video_codecs = caller->pc_factory()
- ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
+ ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO)
.codecs;
auto send_transceiver = caller->pc()->GetTransceivers().front();
@@ -1897,8 +1891,8 @@
EXPECT_TRUE(HasAnyComfortNoiseCodecs(offer->description()));
auto transceiver = caller->pc()->GetTransceivers().front();
- auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_AUDIO);
+ auto capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO);
EXPECT_TRUE(transceiver->SetCodecPreferences(capabilities.codecs).ok());
options.voice_activity_detection = false;
@@ -1929,13 +1923,13 @@
ASSERT_EQ(2u, transceivers.size());
auto audio_transceiver = caller->pc()->GetTransceivers()[0];
- auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_AUDIO);
+ auto capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO);
EXPECT_TRUE(audio_transceiver->SetCodecPreferences(capabilities.codecs).ok());
auto video_transceiver = caller->pc()->GetTransceivers()[1];
- capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_VIDEO);
+ capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO);
EXPECT_TRUE(video_transceiver->SetCodecPreferences(capabilities.codecs).ok());
RTCOfferAnswerOptions options;
@@ -1974,13 +1968,13 @@
ASSERT_EQ(2u, transceivers.size());
auto audio_transceiver = caller->pc()->GetTransceivers()[0];
- auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_AUDIO);
+ auto capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO);
EXPECT_TRUE(audio_transceiver->SetCodecPreferences(capabilities.codecs).ok());
auto video_transceiver = caller->pc()->GetTransceivers()[1];
- capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_VIDEO);
+ capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO);
EXPECT_TRUE(video_transceiver->SetCodecPreferences(capabilities.codecs).ok());
auto answer = caller->CreateAnswer(options);
@@ -2020,13 +2014,13 @@
ASSERT_EQ(2u, transceivers.size());
auto audio_transceiver = caller->pc()->GetTransceivers()[0];
- auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_AUDIO);
+ auto capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO);
EXPECT_TRUE(audio_transceiver->SetCodecPreferences(capabilities.codecs).ok());
auto video_transceiver = caller->pc()->GetTransceivers()[1];
- capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_VIDEO);
+ capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO);
EXPECT_TRUE(video_transceiver->SetCodecPreferences(capabilities.codecs).ok());
auto reoffer = caller->CreateOffer(options);
@@ -2060,7 +2054,7 @@
RtpTransceiverDirection::kSendOnly);
ASSERT_TRUE(error.ok());
auto capabilities = caller->pc_factory()->GetRtpReceiverCapabilities(
- cricket::MediaType::MEDIA_TYPE_AUDIO);
+ webrtc::MediaType::AUDIO);
EXPECT_TRUE(audio_transceiver->SetCodecPreferences(capabilities.codecs).ok());
RTCOfferAnswerOptions options;
EXPECT_TRUE(caller->SetLocalDescription(caller->CreateOffer(options)));
@@ -2089,8 +2083,8 @@
EXPECT_TRUE(video_transceiver
->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly)
.ok());
- auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
- cricket::MediaType::MEDIA_TYPE_VIDEO);
+ auto capabilities =
+ caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO);
auto it =
std::remove_if(capabilities.codecs.begin(), capabilities.codecs.end(),
[](const RtpCodecCapability& codec) {
diff --git a/pc/peer_connection_proxy.h b/pc/peer_connection_proxy.h
index 03f8a80..15f9d7d2 100644
--- a/pc/peer_connection_proxy.h
+++ b/pc/peer_connection_proxy.h
@@ -53,10 +53,10 @@
const RtpTransceiverInit&)
PROXY_METHOD1(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
AddTransceiver,
- cricket::MediaType)
+ webrtc::MediaType)
PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
AddTransceiver,
- cricket::MediaType,
+ webrtc::MediaType,
const RtpTransceiverInit&)
PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
CreateSender,
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index 4b17364..dbf4ac6 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -346,11 +346,11 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
RtpTransceiverInit video_transceiver_init;
video_transceiver_init.stream_ids = {kStreamId1, kStreamId2};
auto video_transceiver =
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, video_transceiver_init);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO, video_transceiver_init);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
@@ -400,7 +400,7 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
EXPECT_TRUE(
transceiver->SetDirectionWithError(RtpTransceiverDirection::kInactive)
.ok());
@@ -431,7 +431,7 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
@@ -502,7 +502,7 @@
TEST_F(PeerConnectionRtpTestUnifiedPlan,
ChangeDirectionInAnswerResultsInRemoveTrackEvent) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto callee = CreatePeerConnection();
callee->AddAudioTrack("audio_track", {});
@@ -803,8 +803,8 @@
// Caller offers to receive audio and video.
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kRecvOnly;
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO, init);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO, init);
// Callee wants to send audio and video tracks.
callee->AddTrack(callee->CreateAudioTrack("audio_track"), {});
@@ -957,7 +957,7 @@
AddTransceiverHasCorrectInitProperties) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
EXPECT_EQ(std::nullopt, transceiver->mid());
EXPECT_FALSE(transceiver->stopped());
EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
@@ -970,14 +970,14 @@
AddAudioTransceiverCreatesAudioSenderAndReceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->media_type());
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ EXPECT_EQ(webrtc::MediaType::AUDIO, transceiver->media_type());
ASSERT_TRUE(transceiver->sender());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->sender()->media_type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, transceiver->sender()->media_type());
ASSERT_TRUE(transceiver->receiver());
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->receiver()->media_type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, transceiver->receiver()->media_type());
auto track = transceiver->receiver()->track();
ASSERT_TRUE(track);
@@ -991,14 +991,14 @@
AddAudioTransceiverCreatesVideoSenderAndReceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
+ EXPECT_EQ(webrtc::MediaType::VIDEO, transceiver->media_type());
ASSERT_TRUE(transceiver->sender());
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->sender()->media_type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, transceiver->sender()->media_type());
ASSERT_TRUE(transceiver->receiver());
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->receiver()->media_type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, transceiver->receiver()->media_type());
auto track = transceiver->receiver()->track();
ASSERT_TRUE(track);
@@ -1012,7 +1012,7 @@
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverShowsInLists) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
EXPECT_EQ(
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>{transceiver},
caller->pc()->GetTransceivers());
@@ -1034,7 +1034,7 @@
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kSendOnly;
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO, init);
EXPECT_EQ(RtpTransceiverDirection::kSendOnly, transceiver->direction());
}
@@ -1089,7 +1089,7 @@
AddTransceiverWithInvalidKindReturnsError) {
auto caller = CreatePeerConnection();
- auto result = caller->pc()->AddTransceiver(cricket::MEDIA_TYPE_DATA);
+ auto result = caller->pc()->AddTransceiver(webrtc::MediaType::DATA);
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
}
@@ -1111,7 +1111,7 @@
auto sender = caller->AddTrack(audio_track);
ASSERT_TRUE(sender);
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, sender->media_type());
+ EXPECT_EQ(webrtc::MediaType::AUDIO, sender->media_type());
EXPECT_EQ(audio_track, sender->track());
}
@@ -1124,7 +1124,7 @@
auto sender = caller->AddTrack(video_track);
ASSERT_TRUE(sender);
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
+ EXPECT_EQ(webrtc::MediaType::VIDEO, sender->media_type());
EXPECT_EQ(video_track, sender->track());
}
@@ -1148,7 +1148,7 @@
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesTransceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto audio_track = caller->CreateAudioTrack("a");
auto sender = caller->AddTrack(audio_track);
ASSERT_TRUE(sender);
@@ -1164,7 +1164,7 @@
AddTrackWithSendEncodingDoesNotReuseTransceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto audio_track = caller->CreateAudioTrack("a");
RtpEncodingParameters encoding;
auto sender = caller->AddTrack(audio_track, {}, {encoding});
@@ -1198,8 +1198,8 @@
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesTransceiverOfType) {
auto caller = CreatePeerConnection();
- auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto sender = caller->AddVideoTrack("v");
ASSERT_EQ(2u, caller->pc()->GetTransceivers().size());
@@ -1214,7 +1214,7 @@
AddTrackDoesNotReuseTransceiverOfWrongType) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto sender = caller->AddVideoTrack("v");
auto transceivers = caller->pc()->GetTransceivers();
@@ -1229,8 +1229,8 @@
AddTrackReusesFirstMatchingTransceiver) {
auto caller = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
auto sender = caller->AddAudioTrack("a");
auto transceivers = caller->pc()->GetTransceivers();
@@ -1248,7 +1248,7 @@
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kInactive;
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO, init);
EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());
@@ -1271,7 +1271,7 @@
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kRecvOnly;
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO, init);
EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());
@@ -1492,8 +1492,8 @@
RtpTransceiverInit recvonly;
recvonly.direction = RtpTransceiverDirection::kRecvOnly;
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, recvonly);
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, recvonly);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO, recvonly);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO, recvonly);
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
@@ -1542,7 +1542,7 @@
EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());
@@ -1561,7 +1561,7 @@
NoRenegotiationNeededAfterTransceiverSetSameDirection) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
caller->observer()->clear_legacy_renegotiation_needed();
caller->observer()->clear_latest_negotiation_needed_event();
@@ -1576,7 +1576,7 @@
NoRenegotiationNeededAfterSetDirectionOnStoppedTransceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
transceiver->StopInternal();
caller->observer()->clear_legacy_renegotiation_needed();
@@ -1591,7 +1591,7 @@
CheckStoppedCurrentDirectionOnStoppedTransceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
transceiver->StopInternal();
EXPECT_TRUE(transceiver->stopping());
@@ -1605,7 +1605,7 @@
CheckForInvalidStateOnStoppingTransceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
transceiver->StopStandard();
EXPECT_TRUE(transceiver->stopping());
@@ -1621,7 +1621,7 @@
CheckForInvalidStateOnStoppedTransceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
transceiver->StopInternal();
EXPECT_TRUE(transceiver->stopping());
@@ -1637,7 +1637,7 @@
CheckForTypeErrorForStoppedOnTransceiver) {
auto caller = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
EXPECT_EQ(
RTCErrorType::INVALID_PARAMETER,
transceiver->SetDirectionWithError(RtpTransceiverDirection::kStopped)
@@ -1650,7 +1650,7 @@
SetLocalDescriptionWithStoppedMediaSection) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
callee->pc()->GetTransceivers()[0]->StopStandard();
ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
@@ -1663,7 +1663,7 @@
StopAndNegotiateCausesTransceiverToDisappear) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
callee->pc()->GetTransceivers()[0]->StopStandard();
ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
@@ -1729,7 +1729,7 @@
init.send_encodings[0].ssrc = 1;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init)
+ ->AddTransceiver(webrtc::MediaType::AUDIO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1748,7 +1748,7 @@
init.send_encodings[0].scale_resolution_down_by = 0.5;
EXPECT_EQ(RTCErrorType::INVALID_RANGE,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init)
+ ->AddTransceiver(webrtc::MediaType::VIDEO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1756,7 +1756,7 @@
init.send_encodings[0].bitrate_priority = 0;
EXPECT_EQ(RTCErrorType::INVALID_RANGE,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init)
+ ->AddTransceiver(webrtc::MediaType::VIDEO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1765,7 +1765,7 @@
init.send_encodings[0].max_bitrate_bps = 100000;
EXPECT_EQ(RTCErrorType::INVALID_RANGE,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init)
+ ->AddTransceiver(webrtc::MediaType::VIDEO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1773,7 +1773,7 @@
init.send_encodings[0].num_temporal_layers = 0;
EXPECT_EQ(RTCErrorType::INVALID_RANGE,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init)
+ ->AddTransceiver(webrtc::MediaType::VIDEO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1781,7 +1781,7 @@
init.send_encodings[0].num_temporal_layers = 5;
EXPECT_EQ(RTCErrorType::INVALID_RANGE,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init)
+ ->AddTransceiver(webrtc::MediaType::VIDEO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1791,7 +1791,7 @@
cricket::CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters();
EXPECT_EQ(RTCErrorType::NONE,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init)
+ ->AddTransceiver(webrtc::MediaType::VIDEO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1801,7 +1801,7 @@
cricket::CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters();
EXPECT_EQ(RTCErrorType::NONE,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init)
+ ->AddTransceiver(webrtc::MediaType::VIDEO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1811,7 +1811,7 @@
cricket::CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters();
EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
caller->pc()
- ->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init)
+ ->AddTransceiver(webrtc::MediaType::VIDEO, init)
.error()
.type());
init.send_encodings = default_send_encodings;
@@ -1827,7 +1827,7 @@
init.send_encodings[0].active = false;
init.send_encodings[0].max_bitrate_bps = 180000;
- auto result = caller->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ auto result = caller->pc()->AddTransceiver(webrtc::MediaType::AUDIO, init);
ASSERT_TRUE(result.ok());
auto init_send_encodings = result.value()->sender()->init_send_encodings();
diff --git a/pc/peer_connection_signaling_unittest.cc b/pc/peer_connection_signaling_unittest.cc
index fd2ab40..1162c40 100644
--- a/pc/peer_connection_signaling_unittest.cc
+++ b/pc/peer_connection_signaling_unittest.cc
@@ -1260,7 +1260,7 @@
auto caller = CreatePeerConnection();
EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
auto transceiver =
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, RtpTransceiverInit());
+ caller->AddTransceiver(webrtc::MediaType::AUDIO, RtpTransceiverInit());
EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());
EXPECT_TRUE(caller->pc()->ShouldFireNegotiationNeededEvent(
caller->observer()->latest_negotiation_needed_event()));
@@ -1271,7 +1271,7 @@
auto caller = CreatePeerConnection();
EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
auto transceiver =
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, RtpTransceiverInit());
+ caller->AddTransceiver(webrtc::MediaType::AUDIO, RtpTransceiverInit());
EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());
auto observer = rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
@@ -1302,7 +1302,7 @@
EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
auto transceiver =
- callee->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, RtpTransceiverInit());
+ callee->AddTransceiver(webrtc::MediaType::AUDIO, RtpTransceiverInit());
EXPECT_TRUE(callee->observer()->has_negotiation_needed_event());
// Change signaling state (to "have-remote-offer") by setting a remote offer.
@@ -1379,7 +1379,7 @@
// and triggered surprising behavior.
auto caller = CreatePeerConnection();
auto transceiver =
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, RtpTransceiverInit());
+ caller->AddTransceiver(webrtc::MediaType::AUDIO, RtpTransceiverInit());
auto offer = caller->CreateOffer(RTCOfferAnswerOptions());
std::string offer_sdp;
diff --git a/pc/peer_connection_simulcast_unittest.cc b/pc/peer_connection_simulcast_unittest.cc
index c912269..3fc80a4 100644
--- a/pc/peer_connection_simulcast_unittest.cc
+++ b/pc/peer_connection_simulcast_unittest.cc
@@ -9,23 +9,16 @@
*/
#include <algorithm>
+#include <cstring>
#include <iterator>
-#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/strings/match.h"
-#include "absl/strings/string_view.h"
-#include "api/audio/audio_device.h"
-#include "api/audio/audio_mixer.h"
-#include "api/audio/audio_processing.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/audio_codecs/opus_audio_decoder_factory.h"
-#include "api/audio_codecs/opus_audio_encoder_factory.h"
#include "api/create_peerconnection_factory.h"
#include "api/jsep.h"
#include "api/media_types.h"
@@ -36,7 +29,6 @@
#include "api/rtp_transceiver_direction.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
-#include "api/uma_metrics.h"
#include "api/video/video_codec_constants.h"
#include "api/video_codecs/video_decoder_factory_template.h"
#include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h"
@@ -48,23 +40,16 @@
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
-#include "media/base/media_constants.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
-#include "pc/channel_interface.h"
#include "pc/peer_connection_wrapper.h"
-#include "pc/sdp_utils.h"
#include "pc/session_description.h"
#include "pc/simulcast_description.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "pc/test/simulcast_layer_util.h"
#include "rtc_base/checks.h"
-#include "rtc_base/gunit.h"
-#include "rtc_base/strings/string_builder.h"
#include "rtc_base/thread.h"
-#include "rtc_base/unique_id_generator.h"
-#include "system_wrappers/include/metrics.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -153,7 +138,7 @@
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
PeerConnectionWrapper* pc,
const std::vector<SimulcastLayer>& layers,
- cricket::MediaType media_type = cricket::MEDIA_TYPE_VIDEO) {
+ webrtc::MediaType media_type = webrtc::MediaType::VIDEO) {
auto init = CreateTransceiverInit(layers);
return pc->AddTransceiver(media_type, init);
}
@@ -211,7 +196,7 @@
for (RtpEncodingParameters& parameters : init.send_encodings) {
parameters.rid = "";
}
- auto transceiver = pc->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ auto transceiver = pc->AddTransceiver(webrtc::MediaType::VIDEO, init);
auto parameters = transceiver->sender()->GetParameters();
ASSERT_EQ(3u, parameters.encodings.size());
EXPECT_THAT(parameters.encodings,
@@ -227,7 +212,7 @@
auto layers = CreateLayers({"f", "h", "remove"}, true);
auto init = CreateTransceiverInit(layers);
init.send_encodings[2].rid = "";
- auto error = pc->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ auto error = pc->AddTransceiver(webrtc::MediaType::VIDEO, init);
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.error().type());
}
@@ -237,7 +222,7 @@
auto pc = pc_wrapper->pc();
auto layers = CreateLayers({"f", "h", "~q"}, true);
auto init = CreateTransceiverInit(layers);
- auto error = pc->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ auto error = pc->AddTransceiver(webrtc::MediaType::VIDEO, init);
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.error().type());
}
@@ -530,7 +515,7 @@
auto remote = CreatePeerConnectionWrapper();
auto layers = CreateLayers({"1", "2", "3", "4"}, true);
auto transceiver =
- AddTransceiver(local.get(), layers, cricket::MEDIA_TYPE_AUDIO);
+ AddTransceiver(local.get(), layers, webrtc::MediaType::AUDIO);
// Should only have the first layer.
auto parameters = transceiver->sender()->GetParameters();
EXPECT_EQ(1u, parameters.encodings.size());
diff --git a/pc/peer_connection_svc_integrationtest.cc b/pc/peer_connection_svc_integrationtest.cc
index 5366fc5..fead8e1 100644
--- a/pc/peer_connection_svc_integrationtest.cc
+++ b/pc/peer_connection_svc_integrationtest.cc
@@ -46,7 +46,7 @@
absl::string_view codec_name) {
RtpCapabilities capabilities =
caller()->pc_factory()->GetRtpReceiverCapabilities(
- cricket::MEDIA_TYPE_VIDEO);
+ webrtc::MediaType::VIDEO);
std::vector<RtpCodecCapability> codecs;
for (const RtpCodecCapability& codec_capability : capabilities.codecs) {
if (codec_capability.name == codec_name)
@@ -101,7 +101,7 @@
RtpCapabilities capabilities =
caller()->pc_factory()->GetRtpReceiverCapabilities(
- cricket::MEDIA_TYPE_VIDEO);
+ webrtc::MediaType::VIDEO);
std::vector<RtpCodecCapability> vp8_codec;
for (const RtpCodecCapability& codec_capability : capabilities.codecs) {
if (codec_capability.name == cricket::kVp8CodecName)
@@ -244,7 +244,7 @@
RtpCapabilities capabilities =
caller()->pc_factory()->GetRtpReceiverCapabilities(
- cricket::MEDIA_TYPE_VIDEO);
+ webrtc::MediaType::VIDEO);
std::vector<RtpCodecCapability> send_codecs = capabilities.codecs;
// Only keep VP9 in the caller
send_codecs.erase(std::partition(send_codecs.begin(), send_codecs.end(),
diff --git a/pc/peer_connection_wrapper.cc b/pc/peer_connection_wrapper.cc
index 10bce74..20491fe 100644
--- a/pc/peer_connection_wrapper.cc
+++ b/pc/peer_connection_wrapper.cc
@@ -277,7 +277,7 @@
}
rtc::scoped_refptr<RtpTransceiverInterface>
-PeerConnectionWrapper::AddTransceiver(cricket::MediaType media_type) {
+PeerConnectionWrapper::AddTransceiver(webrtc::MediaType media_type) {
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
pc()->AddTransceiver(media_type);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
@@ -285,7 +285,7 @@
}
rtc::scoped_refptr<RtpTransceiverInterface>
-PeerConnectionWrapper::AddTransceiver(cricket::MediaType media_type,
+PeerConnectionWrapper::AddTransceiver(webrtc::MediaType media_type,
const RtpTransceiverInit& init) {
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
pc()->AddTransceiver(media_type, init);
diff --git a/pc/peer_connection_wrapper.h b/pc/peer_connection_wrapper.h
index a72f2a0..49b2da7 100644
--- a/pc/peer_connection_wrapper.h
+++ b/pc/peer_connection_wrapper.h
@@ -132,9 +132,9 @@
// AddTransceiver method. They return the result of calling AddTransceiver
// with the given arguments, DCHECKing if there is an error.
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const RtpTransceiverInit& init);
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track);
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 3ef11d6..c55aca7 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -132,51 +132,51 @@
}
std::string RTCInboundRtpStreamStatsIDFromSSRC(const std::string& transport_id,
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
uint32_t ssrc) {
char buf[1024];
SimpleStringBuilder sb(buf);
sb << 'I' << transport_id
- << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V') << ssrc;
+ << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V') << ssrc;
return sb.str();
}
std::string RTCOutboundRtpStreamStatsIDFromSSRC(const std::string& transport_id,
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
uint32_t ssrc) {
char buf[1024];
SimpleStringBuilder sb(buf);
sb << 'O' << transport_id
- << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V') << ssrc;
+ << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V') << ssrc;
return sb.str();
}
std::string RTCRemoteInboundRtpStreamStatsIdFromSourceSsrc(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
uint32_t source_ssrc) {
char buf[1024];
SimpleStringBuilder sb(buf);
- sb << "RI" << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V')
+ sb << "RI" << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V')
<< source_ssrc;
return sb.str();
}
std::string RTCRemoteOutboundRTPStreamStatsIDFromSSRC(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
uint32_t source_ssrc) {
char buf[1024];
SimpleStringBuilder sb(buf);
- sb << "RO" << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V')
+ sb << "RO" << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V')
<< source_ssrc;
return sb.str();
}
std::string RTCMediaSourceStatsIDFromKindAndAttachment(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
int attachment_id) {
char buf[1024];
SimpleStringBuilder sb(buf);
- sb << 'S' << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V')
+ sb << 'S' << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V')
<< attachment_id;
return sb.str();
}
@@ -453,7 +453,7 @@
RTCStatsReport* report) {
auto inbound_audio = std::make_unique<RTCInboundRtpStreamStats>(
/*id=*/RTCInboundRtpStreamStatsIDFromSSRC(
- transport_id, cricket::MEDIA_TYPE_AUDIO, voice_receiver_info.ssrc()),
+ transport_id, webrtc::MediaType::AUDIO, voice_receiver_info.ssrc()),
timestamp);
SetInboundRTPStreamStatsFromMediaReceiverInfo(voice_receiver_info,
inbound_audio.get());
@@ -536,7 +536,7 @@
CreateRemoteOutboundMediaStreamStats(
const cricket::MediaReceiverInfo& media_receiver_info,
const std::string& mid,
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const RTCInboundRtpStreamStats& inbound_audio_stats,
const std::string& transport_id,
const bool stats_timestamp_with_environment_clock) {
@@ -561,7 +561,7 @@
// Populate.
// - RTCRtpStreamStats.
stats->ssrc = media_receiver_info.ssrc();
- stats->kind = cricket::MediaTypeToString(media_type);
+ stats->kind = webrtc::MediaTypeToString(media_type);
stats->transport_id = transport_id;
if (inbound_audio_stats.codec_id.has_value()) {
stats->codec_id = *inbound_audio_stats.codec_id;
@@ -599,8 +599,8 @@
Timestamp timestamp,
RTCStatsReport* report) {
auto inbound_video = std::make_unique<RTCInboundRtpStreamStats>(
- RTCInboundRtpStreamStatsIDFromSSRC(
- transport_id, cricket::MEDIA_TYPE_VIDEO, video_receiver_info.ssrc()),
+ RTCInboundRtpStreamStatsIDFromSSRC(transport_id, webrtc::MediaType::VIDEO,
+ video_receiver_info.ssrc()),
timestamp);
SetInboundRTPStreamStatsFromMediaReceiverInfo(video_receiver_info,
inbound_video.get());
@@ -749,7 +749,7 @@
RTCStatsReport* report) {
auto outbound_audio = std::make_unique<RTCOutboundRtpStreamStats>(
RTCOutboundRtpStreamStatsIDFromSSRC(
- transport_id, cricket::MEDIA_TYPE_AUDIO, voice_sender_info.ssrc()),
+ transport_id, webrtc::MediaType::AUDIO, voice_sender_info.ssrc()),
timestamp);
SetOutboundRTPStreamStatsFromMediaSenderInfo(voice_sender_info,
outbound_audio.get());
@@ -784,7 +784,7 @@
RTCStatsReport* report) {
auto outbound_video = std::make_unique<RTCOutboundRtpStreamStats>(
RTCOutboundRtpStreamStatsIDFromSSRC(
- transport_id, cricket::MEDIA_TYPE_VIDEO, video_sender_info.ssrc()),
+ transport_id, webrtc::MediaType::VIDEO, video_sender_info.ssrc()),
timestamp);
SetOutboundRTPStreamStatsFromMediaSenderInfo(video_sender_info,
outbound_video.get());
@@ -874,7 +874,7 @@
ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
const std::string& transport_id,
const ReportBlockData& report_block,
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::map<std::string, RTCOutboundRtpStreamStats*>& outbound_rtps,
const RTCStatsReport& report,
const bool stats_timestamp_with_environment_clock) {
@@ -890,7 +890,7 @@
arrival_timestamp);
remote_inbound->ssrc = report_block.source_ssrc();
remote_inbound->kind =
- media_type == cricket::MEDIA_TYPE_AUDIO ? "audio" : "video";
+ media_type == webrtc::MediaType::AUDIO ? "audio" : "video";
remote_inbound->packets_lost = report_block.cumulative_lost();
remote_inbound->fraction_lost = report_block.fraction_lost();
if (report_block.num_rtts() > 0) {
@@ -1608,7 +1608,7 @@
static_cast<AudioTrackInterface*>(track.get());
auto audio_source_stats = std::make_unique<RTCAudioSourceStats>(
RTCMediaSourceStatsIDFromKindAndAttachment(
- cricket::MEDIA_TYPE_AUDIO, sender_internal->AttachmentId()),
+ webrtc::MediaType::AUDIO, sender_internal->AttachmentId()),
timestamp);
// TODO(https://crbug.com/webrtc/10771): We shouldn't need to have an
// SSRC assigned (there shouldn't need to exist a send-stream, created
@@ -1646,7 +1646,7 @@
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
auto video_source_stats = std::make_unique<RTCVideoSourceStats>(
RTCMediaSourceStatsIDFromKindAndAttachment(
- cricket::MEDIA_TYPE_VIDEO, sender_internal->AttachmentId()),
+ webrtc::MediaType::VIDEO, sender_internal->AttachmentId()),
timestamp);
auto* video_track = static_cast<VideoTrackInterface*>(track.get());
auto* video_source = video_track->GetSource();
@@ -1710,9 +1710,9 @@
Thread::ScopedDisallowBlockingCalls no_blocking_calls;
for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos) {
- if (stats.media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (stats.media_type == webrtc::MediaType::AUDIO) {
ProduceAudioRTPStreamStats_n(timestamp, stats, report);
- } else if (stats.media_type == cricket::MEDIA_TYPE_VIDEO) {
+ } else if (stats.media_type == webrtc::MediaType::VIDEO) {
ProduceVideoRTPStreamStats_n(timestamp, stats, report);
} else {
RTC_DCHECK_NOTREACHED();
@@ -1751,7 +1751,7 @@
if (audio_track) {
inbound_audio->track_identifier = audio_track->id();
}
- if (audio_device_stats_ && stats.media_type == cricket::MEDIA_TYPE_AUDIO &&
+ if (audio_device_stats_ && stats.media_type == webrtc::MediaType::AUDIO &&
stats.current_direction &&
(*stats.current_direction == RtpTransceiverDirection::kSendRecv ||
*stats.current_direction == RtpTransceiverDirection::kRecvOnly)) {
@@ -1765,7 +1765,7 @@
}
// Remote-outbound.
auto remote_outbound_audio = CreateRemoteOutboundMediaStreamStats(
- voice_receiver_info, mid, cricket::MEDIA_TYPE_AUDIO, *inbound_audio_ptr,
+ voice_receiver_info, mid, webrtc::MediaType::AUDIO, *inbound_audio_ptr,
transport_id, stats_timestamp_with_environment_clock_);
// Add stats.
if (remote_outbound_audio) {
@@ -1797,7 +1797,7 @@
stats.track_media_info_map.GetAttachmentIdByTrack(audio_track.get())
.value();
outbound_audio->media_source_id =
- RTCMediaSourceStatsIDFromKindAndAttachment(cricket::MEDIA_TYPE_AUDIO,
+ RTCMediaSourceStatsIDFromKindAndAttachment(webrtc::MediaType::AUDIO,
attachment_id);
}
auto audio_outbound_pair =
@@ -1818,7 +1818,7 @@
stats.track_media_info_map.voice_media_info()->senders) {
for (const auto& report_block_data : voice_sender_info.report_block_datas) {
report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
- transport_id, report_block_data, cricket::MEDIA_TYPE_AUDIO,
+ transport_id, report_block_data, webrtc::MediaType::AUDIO,
audio_outbound_rtps, *report,
stats_timestamp_with_environment_clock_));
}
@@ -1860,7 +1860,7 @@
}
// Remote-outbound.
auto remote_outbound_video = CreateRemoteOutboundMediaStreamStats(
- video_receiver_info, mid, cricket::MEDIA_TYPE_VIDEO, *inbound_video_ptr,
+ video_receiver_info, mid, webrtc::MediaType::VIDEO, *inbound_video_ptr,
transport_id, stats_timestamp_with_environment_clock_);
// Add stats.
if (remote_outbound_video) {
@@ -1892,7 +1892,7 @@
stats.track_media_info_map.GetAttachmentIdByTrack(video_track.get())
.value();
outbound_video->media_source_id =
- RTCMediaSourceStatsIDFromKindAndAttachment(cricket::MEDIA_TYPE_VIDEO,
+ RTCMediaSourceStatsIDFromKindAndAttachment(webrtc::MediaType::VIDEO,
attachment_id);
}
auto video_outbound_pair =
@@ -1913,7 +1913,7 @@
stats.track_media_info_map.video_media_info()->senders) {
for (const auto& report_block_data : video_sender_info.report_block_datas) {
report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
- transport_id, report_block_data, cricket::MEDIA_TYPE_VIDEO,
+ transport_id, report_block_data, webrtc::MediaType::VIDEO,
video_outbound_rtps, *report,
stats_timestamp_with_environment_clock_));
}
@@ -2104,7 +2104,7 @@
for (const auto& transceiver_proxy : transceivers) {
RtpTransceiver* transceiver = transceiver_proxy->internal();
- cricket::MediaType media_type = transceiver->media_type();
+ webrtc::MediaType media_type = transceiver->media_type();
// Prepare stats entry. The TrackMediaInfoMap will be filled in after the
// stats have been fetched on the worker thread.
@@ -2122,7 +2122,7 @@
stats.mid = channel->mid();
stats.transport_name = std::string(channel->transport_name());
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
auto voice_send_channel = channel->voice_media_send_channel();
RTC_DCHECK(voice_send_stats.find(voice_send_channel) ==
voice_send_stats.end());
@@ -2134,7 +2134,7 @@
voice_receive_stats.end());
voice_receive_stats.insert(std::make_pair(
voice_receive_channel, cricket::VoiceMediaReceiveInfo()));
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ } else if (media_type == webrtc::MediaType::VIDEO) {
auto video_send_channel = channel->video_media_send_channel();
RTC_DCHECK(video_send_stats.find(video_send_channel) ==
video_send_stats.end());
@@ -2189,14 +2189,14 @@
std::optional<cricket::VideoMediaInfo> video_media_info;
auto channel = transceiver->channel();
if (channel) {
- cricket::MediaType media_type = transceiver->media_type();
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ webrtc::MediaType media_type = transceiver->media_type();
+ if (media_type == webrtc::MediaType::AUDIO) {
auto voice_send_channel = channel->voice_media_send_channel();
auto voice_receive_channel = channel->voice_media_receive_channel();
voice_media_info = cricket::VoiceMediaInfo(
std::move(voice_send_stats[voice_send_channel]),
std::move(voice_receive_stats[voice_receive_channel]));
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ } else if (media_type == webrtc::MediaType::VIDEO) {
auto video_send_channel = channel->video_media_send_channel();
auto video_receive_channel = channel->video_media_receive_channel();
video_media_info = cricket::VideoMediaInfo(
@@ -2217,7 +2217,7 @@
stats.track_media_info_map.Initialize(std::move(voice_media_info),
std::move(video_media_info),
senders, receivers);
- if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (transceiver->media_type() == webrtc::MediaType::AUDIO) {
has_audio_receiver |= !receivers.empty();
}
}
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index b3264de..3fbe7f7 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -172,7 +172,7 @@
// then `mid` and `transport_name` will be null.
struct RtpTransceiverStatsInfo {
rtc::scoped_refptr<RtpTransceiver> transceiver;
- cricket::MediaType media_type;
+ webrtc::MediaType media_type;
std::optional<std::string> mid;
std::optional<std::string> transport_name;
TrackMediaInfoMap track_media_info_map;
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 0f843d8..f291b7c 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -56,6 +56,8 @@
#include "api/video_codecs/scalability_mode.h"
#include "call/call.h"
#include "common_video/include/quality_limitation_reason.h"
+#include "json/reader.h"
+#include "json/value.h"
#include "media/base/media_channel.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
@@ -87,7 +89,6 @@
#include "rtc_base/ssl_identity.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/string_encode.h"
-#include "rtc_base/strings/json.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
@@ -313,30 +314,30 @@
};
rtc::scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::string& track_id,
MediaStreamTrackInterface::TrackState track_state,
bool create_fake_audio_processor = false) {
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
return FakeAudioTrackForStats::Create(track_id, track_state,
create_fake_audio_processor);
} else {
- RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK_EQ(media_type, webrtc::MediaType::VIDEO);
return FakeVideoTrackForStats::Create(track_id, track_state, nullptr);
}
}
rtc::scoped_refptr<MockRtpSenderInternal> CreateMockSender(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
uint32_t ssrc,
int attachment_id,
std::vector<std::string> local_stream_ids) {
RTC_DCHECK(!track ||
(track->kind() == MediaStreamTrackInterface::kAudioKind &&
- media_type == cricket::MEDIA_TYPE_AUDIO) ||
+ media_type == webrtc::MediaType::AUDIO) ||
(track->kind() == MediaStreamTrackInterface::kVideoKind &&
- media_type == cricket::MEDIA_TYPE_VIDEO));
+ media_type == webrtc::MediaType::VIDEO));
auto sender = rtc::make_ref_counted<MockRtpSenderInternal>();
EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track));
EXPECT_CALL(*sender, ssrc()).WillRepeatedly(Return(ssrc));
@@ -372,8 +373,8 @@
EXPECT_CALL(*receiver, media_type())
.WillRepeatedly(
Return(track->kind() == MediaStreamTrackInterface::kAudioKind
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO));
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO));
EXPECT_CALL(*receiver, GetParameters()).WillRepeatedly(Invoke([ssrc]() {
RtpParameters params;
params.encodings.push_back(RtpEncodingParameters());
@@ -426,7 +427,7 @@
}
rtc::scoped_refptr<MockRtpSenderInternal> SetupLocalTrackAndSender(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::string& track_id,
uint32_t ssrc,
bool add_stream,
@@ -438,7 +439,7 @@
}
rtc::scoped_refptr<MediaStreamTrackInterface> track;
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
track = CreateFakeTrack(media_type, track_id,
MediaStreamTrackInterface::kLive);
if (add_stream) {
@@ -464,7 +465,7 @@
}
rtc::scoped_refptr<MockRtpReceiverInternal> SetupRemoteTrackAndReceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::string& track_id,
const std::string& stream_id,
uint32_t ssrc) {
@@ -473,7 +474,7 @@
pc_->mutable_remote_streams()->AddStream(remote_stream);
rtc::scoped_refptr<MediaStreamTrackInterface> track;
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
track = CreateFakeTrack(media_type, track_id,
MediaStreamTrackInterface::kLive);
remote_stream->AddTrack(rtc::scoped_refptr<AudioTrackInterface>(
@@ -529,7 +530,7 @@
voice_media_info.senders.push_back(voice_sender_info);
rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockSender(
- cricket::MEDIA_TYPE_AUDIO,
+ webrtc::MediaType::AUDIO,
rtc::scoped_refptr<MediaStreamTrackInterface>(local_audio_track),
voice_sender_info.local_stats[0].ssrc,
voice_sender_info.local_stats[0].ssrc + 10, local_stream_ids);
@@ -568,7 +569,7 @@
video_media_info.senders.push_back(video_sender_info);
video_media_info.aggregated_senders.push_back(video_sender_info);
rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockSender(
- cricket::MEDIA_TYPE_VIDEO,
+ webrtc::MediaType::VIDEO,
rtc::scoped_refptr<MediaStreamTrackInterface>(local_video_track),
video_sender_info.local_stats[0].ssrc,
video_sender_info.local_stats[0].ssrc + 10, local_stream_ids);
@@ -724,10 +725,10 @@
pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
// outbound-rtp's sender
graph.sender = stats_->SetupLocalTrackAndSender(
- cricket::MEDIA_TYPE_VIDEO, "LocalVideoTrackID", 3, false, 50);
+ webrtc::MediaType::VIDEO, "LocalVideoTrackID", 3, false, 50);
// inbound-rtp's receiver
graph.receiver = stats_->SetupRemoteTrackAndReceiver(
- cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 4);
+ webrtc::MediaType::VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 4);
// peer-connection
graph.peer_connection_id = "P";
// media-source (kind: video)
@@ -825,10 +826,10 @@
pc_->AddVoiceChannel("VoiceMid", "TransportName", media_info);
// outbound-rtp's sender
graph.sender = stats_->SetupLocalTrackAndSender(
- cricket::MEDIA_TYPE_AUDIO, "LocalAudioTrackID", kLocalSsrc, false, 50);
+ webrtc::MediaType::AUDIO, "LocalAudioTrackID", kLocalSsrc, false, 50);
// inbound-rtp's receiver
graph.receiver = stats_->SetupRemoteTrackAndReceiver(
- cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID", "RemoteStreamId",
+ webrtc::MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId",
kRemoteSsrc);
// peer-connection
graph.peer_connection_id = "P";
@@ -1082,7 +1083,7 @@
RtpCodecParameters inbound_audio_codec;
inbound_audio_codec.payload_type = 1;
- inbound_audio_codec.kind = cricket::MEDIA_TYPE_AUDIO;
+ inbound_audio_codec.kind = webrtc::MediaType::AUDIO;
inbound_audio_codec.name = "opus";
inbound_audio_codec.clock_rate = 1337;
inbound_audio_codec.num_channels = 1;
@@ -1092,7 +1093,7 @@
RtpCodecParameters outbound_audio_codec;
outbound_audio_codec.payload_type = 2;
- outbound_audio_codec.kind = cricket::MEDIA_TYPE_AUDIO;
+ outbound_audio_codec.kind = webrtc::MediaType::AUDIO;
outbound_audio_codec.name = "isac";
outbound_audio_codec.clock_rate = 1338;
outbound_audio_codec.num_channels = 2;
@@ -1104,7 +1105,7 @@
RtpCodecParameters inbound_video_codec;
inbound_video_codec.payload_type = 3;
- inbound_video_codec.kind = cricket::MEDIA_TYPE_VIDEO;
+ inbound_video_codec.kind = webrtc::MediaType::VIDEO;
inbound_video_codec.name = "H264";
inbound_video_codec.clock_rate = 1339;
inbound_video_codec.parameters = {{"level-asymmetry-allowed", "1"},
@@ -1115,7 +1116,7 @@
RtpCodecParameters outbound_video_codec;
outbound_video_codec.payload_type = 4;
- outbound_video_codec.kind = cricket::MEDIA_TYPE_VIDEO;
+ outbound_video_codec.kind = webrtc::MediaType::VIDEO;
outbound_video_codec.name = "VP8";
outbound_video_codec.clock_rate = 1340;
video_media_info.send_codecs.insert(
@@ -1227,14 +1228,14 @@
// PT=10
RtpCodecParameters outbound_codec_pt10;
outbound_codec_pt10.payload_type = 10;
- outbound_codec_pt10.kind = cricket::MEDIA_TYPE_VIDEO;
+ outbound_codec_pt10.kind = webrtc::MediaType::VIDEO;
outbound_codec_pt10.name = "VP8";
outbound_codec_pt10.clock_rate = 9000;
// PT=11
RtpCodecParameters outbound_codec_pt11;
outbound_codec_pt11.payload_type = 11;
- outbound_codec_pt11.kind = cricket::MEDIA_TYPE_VIDEO;
+ outbound_codec_pt11.kind = webrtc::MediaType::VIDEO;
outbound_codec_pt11.name = "VP8";
outbound_codec_pt11.clock_rate = 9000;
@@ -1292,7 +1293,7 @@
// PT=111, useinbandfec=0
RtpCodecParameters inbound_codec_pt111_nofec;
inbound_codec_pt111_nofec.payload_type = 111;
- inbound_codec_pt111_nofec.kind = cricket::MEDIA_TYPE_AUDIO;
+ inbound_codec_pt111_nofec.kind = webrtc::MediaType::AUDIO;
inbound_codec_pt111_nofec.name = "opus";
inbound_codec_pt111_nofec.clock_rate = 48000;
inbound_codec_pt111_nofec.parameters.insert(
@@ -1301,7 +1302,7 @@
// PT=111, useinbandfec=1
RtpCodecParameters inbound_codec_pt111_fec;
inbound_codec_pt111_fec.payload_type = 111;
- inbound_codec_pt111_fec.kind = cricket::MEDIA_TYPE_AUDIO;
+ inbound_codec_pt111_fec.kind = webrtc::MediaType::AUDIO;
inbound_codec_pt111_fec.name = "opus";
inbound_codec_pt111_fec.clock_rate = 48000;
inbound_codec_pt111_fec.parameters.insert(
@@ -1349,7 +1350,7 @@
// PT=112, useinbandfec=1
RtpCodecParameters inbound_codec_pt112_fec;
inbound_codec_pt112_fec.payload_type = 112;
- inbound_codec_pt112_fec.kind = cricket::MEDIA_TYPE_AUDIO;
+ inbound_codec_pt112_fec.kind = webrtc::MediaType::AUDIO;
inbound_codec_pt112_fec.name = "opus";
inbound_codec_pt112_fec.clock_rate = 48000;
inbound_codec_pt112_fec.parameters.insert(
@@ -2199,7 +2200,7 @@
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
- codec_parameters.kind = cricket::MEDIA_TYPE_AUDIO;
+ codec_parameters.kind = webrtc::MediaType::AUDIO;
codec_parameters.name = "dummy";
codec_parameters.clock_rate = 0;
voice_media_info.receive_codecs.insert(
@@ -2208,7 +2209,7 @@
auto voice_media_channels =
pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
stats_->SetupRemoteTrackAndReceiver(
- cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
+ webrtc::MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
// Needed for playoutId to be populated.
pc_->SetAudioDeviceStats(AudioDeviceModule::Stats());
@@ -2289,7 +2290,7 @@
pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
stats_->SetupRemoteTrackAndReceiver(
- cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
+ webrtc::MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
// Needed for playoutId to be populated.
pc_->SetAudioDeviceStats(AudioDeviceModule::Stats());
@@ -2377,7 +2378,7 @@
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
- codec_parameters.kind = cricket::MEDIA_TYPE_VIDEO;
+ codec_parameters.kind = webrtc::MediaType::VIDEO;
codec_parameters.name = "dummy";
codec_parameters.clock_rate = 0;
video_media_info.receive_codecs.insert(
@@ -2386,7 +2387,7 @@
auto video_media_channels =
pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
stats_->SetupRemoteTrackAndReceiver(
- cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1);
+ webrtc::MediaType::VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1);
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
@@ -2487,7 +2488,7 @@
pc_->AddVoiceChannel("AudioMid", "TransportName", {});
stats_->SetupRemoteTrackAndReceiver(
- cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
+ webrtc::MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
auto stats_of_track_type = report->GetStatsOfType<RTCAudioPlayoutStats>();
@@ -2532,7 +2533,7 @@
pc_->AddVideoChannel("Mid0", "Transport0", video_media_info);
stats_->SetupRemoteTrackAndReceiver(
- cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1);
+ webrtc::MediaType::VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1);
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
@@ -2561,14 +2562,14 @@
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
- codec_parameters.kind = cricket::MEDIA_TYPE_AUDIO;
+ codec_parameters.kind = webrtc::MediaType::AUDIO;
codec_parameters.name = "dummy";
codec_parameters.clock_rate = 0;
voice_media_info.send_codecs.insert(
std::make_pair(codec_parameters.payload_type, codec_parameters));
pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
- stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
+ stats_->SetupLocalTrackAndSender(webrtc::MediaType::AUDIO,
"LocalAudioTrackID", 1, true,
/*attachment_id=*/50);
@@ -2650,7 +2651,7 @@
video_media_info.aggregated_senders.push_back(video_media_info.senders[0]);
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
- codec_parameters.kind = cricket::MEDIA_TYPE_AUDIO;
+ codec_parameters.kind = webrtc::MediaType::AUDIO;
codec_parameters.name = "dummy";
codec_parameters.clock_rate = 0;
video_media_info.send_codecs.insert(
@@ -2664,7 +2665,7 @@
auto video_media_channels =
pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
- stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_VIDEO,
+ stats_->SetupLocalTrackAndSender(webrtc::MediaType::VIDEO,
"LocalVideoTrackID", 1, true,
/*attachment_id=*/50);
@@ -3009,7 +3010,7 @@
RtpCodecParameters codec_parameters;
codec_parameters.payload_type = 42;
- codec_parameters.kind = cricket::MEDIA_TYPE_AUDIO;
+ codec_parameters.kind = webrtc::MediaType::AUDIO;
codec_parameters.name = "dummy";
codec_parameters.clock_rate = 0;
voice_media_info.send_codecs.insert(
@@ -3017,7 +3018,7 @@
// Emulates the case where AddTrack is used without an associated MediaStream
pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
- stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
+ stats_->SetupLocalTrackAndSender(webrtc::MediaType::AUDIO,
"LocalAudioTrackID", 1, false,
/*attachment_id=*/50);
@@ -3063,7 +3064,7 @@
voice_media_info.senders[0].apm_statistics.echo_return_loss_enhancement =
52.0;
pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
- stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
+ stats_->SetupLocalTrackAndSender(webrtc::MediaType::AUDIO,
"LocalAudioTrackID", kSsrc, false,
kAttachmentId);
@@ -3107,7 +3108,7 @@
auto video_track = FakeVideoTrackForStats::Create(
"LocalVideoTrackID", MediaStreamTrackInterface::kLive, video_source);
rtc::scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
- cricket::MEDIA_TYPE_VIDEO, video_track, kSsrc, kAttachmentId, {});
+ webrtc::MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {});
EXPECT_CALL(*sender, Stop());
EXPECT_CALL(*sender, SetMediaChannel(_));
EXPECT_CALL(*sender, SetSendCodecs(_));
@@ -3152,7 +3153,7 @@
auto video_track = FakeVideoTrackForStats::Create(
"LocalVideoTrackID", MediaStreamTrackInterface::kLive, video_source);
rtc::scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
- cricket::MEDIA_TYPE_VIDEO, video_track, kNoSsrc, kAttachmentId, {});
+ webrtc::MediaType::VIDEO, video_track, kNoSsrc, kAttachmentId, {});
EXPECT_CALL(*sender, Stop());
EXPECT_CALL(*sender, SetMediaChannel(_));
EXPECT_CALL(*sender, SetSendCodecs(_));
@@ -3183,7 +3184,7 @@
"LocalVideoTrackID", MediaStreamTrackInterface::kLive,
/*source=*/nullptr);
rtc::scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
- cricket::MEDIA_TYPE_VIDEO, video_track, kSsrc, kAttachmentId, {});
+ webrtc::MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {});
EXPECT_CALL(*sender, Stop());
EXPECT_CALL(*sender, SetMediaChannel(_));
EXPECT_CALL(*sender, SetSendCodecs(_));
@@ -3207,7 +3208,7 @@
voice_media_info.senders[0].local_stats[0].ssrc = kSsrc;
pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
rtc::scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
- cricket::MEDIA_TYPE_AUDIO, /*track=*/nullptr, kSsrc, kAttachmentId, {});
+ webrtc::MediaType::AUDIO, /*track=*/nullptr, kSsrc, kAttachmentId, {});
EXPECT_CALL(*sender, Stop());
EXPECT_CALL(*sender, SetMediaChannel(_));
EXPECT_CALL(*sender, SetSendCodecs(_));
@@ -3217,21 +3218,21 @@
EXPECT_FALSE(report->Get("SA42"));
}
-// Parameterized tests on cricket::MediaType (audio or video).
+// Parameterized tests on webrtc::MediaType (audio or video).
class RTCStatsCollectorTestWithParamKind
: public RTCStatsCollectorTest,
- public ::testing::WithParamInterface<cricket::MediaType> {
+ public ::testing::WithParamInterface<webrtc::MediaType> {
public:
RTCStatsCollectorTestWithParamKind() : media_type_(GetParam()) {
- RTC_DCHECK(media_type_ == cricket::MEDIA_TYPE_AUDIO ||
- media_type_ == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(media_type_ == webrtc::MediaType::AUDIO ||
+ media_type_ == webrtc::MediaType::VIDEO);
}
std::string MediaTypeCharStr() const {
switch (media_type_) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
return "A";
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
return "V";
default:
RTC_DCHECK_NOTREACHED();
@@ -3241,9 +3242,9 @@
std::string MediaTypeKind() const {
switch (media_type_) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
return "audio";
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
return "video";
default:
RTC_DCHECK_NOTREACHED();
@@ -3258,7 +3259,7 @@
const std::vector<ReportBlockData>& report_block_datas,
std::optional<RtpCodecParameters> codec) {
switch (media_type_) {
- case cricket::MEDIA_TYPE_AUDIO: {
+ case webrtc::MediaType::AUDIO: {
cricket::VoiceMediaInfo voice_media_info;
for (const auto& report_block_data : report_block_datas) {
cricket::VoiceSenderInfo sender;
@@ -3275,7 +3276,7 @@
pc_->AddVoiceChannel("mid", transport_name, voice_media_info);
return;
}
- case cricket::MEDIA_TYPE_VIDEO: {
+ case webrtc::MediaType::VIDEO: {
cricket::VideoMediaInfo video_media_info;
for (const auto& report_block_data : report_block_datas) {
cricket::VideoSenderInfo sender;
@@ -3293,14 +3294,14 @@
pc_->AddVideoChannel("mid", transport_name, video_media_info);
return;
}
- case cricket::MEDIA_TYPE_DATA:
+ case webrtc::MediaType::DATA:
default:
RTC_DCHECK_NOTREACHED();
}
}
protected:
- cricket::MediaType media_type_;
+ webrtc::MediaType media_type_;
};
// Verifies RTCRemoteInboundRtpStreamStats members that don't require
@@ -3515,8 +3516,8 @@
INSTANTIATE_TEST_SUITE_P(All,
RTCStatsCollectorTestWithParamKind,
- ::testing::Values(cricket::MEDIA_TYPE_AUDIO, // "/0"
- cricket::MEDIA_TYPE_VIDEO)); // "/1"
+ ::testing::Values(webrtc::MediaType::AUDIO, // "/0"
+ webrtc::MediaType::VIDEO)); // "/1"
// Checks that no remote outbound stats are collected if not available in
// `VoiceMediaInfo`.
@@ -3567,7 +3568,7 @@
pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
rtc::scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
- cricket::MEDIA_TYPE_VIDEO, /*track=*/nullptr, kSsrc, kAttachmentId, {});
+ webrtc::MediaType::VIDEO, /*track=*/nullptr, kSsrc, kAttachmentId, {});
EXPECT_CALL(*sender, Stop());
EXPECT_CALL(*sender, SetMediaChannel(_));
EXPECT_CALL(*sender, SetSendCodecs(_));
@@ -3586,7 +3587,7 @@
// Local audio track
rtc::scoped_refptr<MediaStreamTrackInterface> local_audio_track =
- CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "LocalAudioTrackID",
+ CreateFakeTrack(webrtc::MediaType::AUDIO, "LocalAudioTrackID",
MediaStreamTrackInterface::kEnded,
/*create_fake_audio_processor=*/true);
local_stream->AddTrack(rtc::scoped_refptr<AudioTrackInterface>(
@@ -3686,11 +3687,10 @@
// Before SetLocalDescription() senders don't have an SSRC.
// To simulate this case we create a mock sender with SSRC=0.
TEST_F(RTCStatsCollectorTest, RtpIsMissingWhileSsrcIsZero) {
- rtc::scoped_refptr<MediaStreamTrackInterface> track =
- CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "audioTrack",
- MediaStreamTrackInterface::kLive);
+ rtc::scoped_refptr<MediaStreamTrackInterface> track = CreateFakeTrack(
+ webrtc::MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive);
rtc::scoped_refptr<MockRtpSenderInternal> sender =
- CreateMockSender(cricket::MEDIA_TYPE_AUDIO, track, 0, 49, {});
+ CreateMockSender(webrtc::MediaType::AUDIO, track, 0, 49, {});
EXPECT_CALL(*sender, Stop());
EXPECT_CALL(*sender, SetSendCodecs(_));
pc_->AddSender(sender);
@@ -3704,11 +3704,10 @@
// We may also be in a case where the SSRC has been assigned but no
// `voice_sender_info` stats exist yet.
TEST_F(RTCStatsCollectorTest, DoNotCrashIfSsrcIsKnownButInfosAreStillMissing) {
- rtc::scoped_refptr<MediaStreamTrackInterface> track =
- CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "audioTrack",
- MediaStreamTrackInterface::kLive);
+ rtc::scoped_refptr<MediaStreamTrackInterface> track = CreateFakeTrack(
+ webrtc::MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive);
rtc::scoped_refptr<MockRtpSenderInternal> sender =
- CreateMockSender(cricket::MEDIA_TYPE_AUDIO, track, 4711, 49, {});
+ CreateMockSender(webrtc::MediaType::AUDIO, track, 4711, 49, {});
EXPECT_CALL(*sender, Stop());
EXPECT_CALL(*sender, SetSendCodecs(_));
pc_->AddSender(sender);
diff --git a/pc/rtp_parameters_conversion.cc b/pc/rtp_parameters_conversion.cc
index 415eac6..a2ba2ef 100644
--- a/pc/rtp_parameters_conversion.cc
+++ b/pc/rtp_parameters_conversion.cc
@@ -79,8 +79,8 @@
RtpCodecCapability codec;
codec.name = cricket_codec.name;
codec.kind = cricket_codec.type == cricket::Codec::Type::kAudio
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO;
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO;
codec.clock_rate.emplace(cricket_codec.clockrate);
codec.preferred_payload_type.emplace(cricket_codec.id);
for (const cricket::FeedbackParam& cricket_feedback :
diff --git a/pc/rtp_parameters_conversion_unittest.cc b/pc/rtp_parameters_conversion_unittest.cc
index 337313e..7c87f41 100644
--- a/pc/rtp_parameters_conversion_unittest.cc
+++ b/pc/rtp_parameters_conversion_unittest.cc
@@ -86,7 +86,7 @@
RtpCodecCapability codec = ToRtpCodecCapability(cricket_codec);
EXPECT_EQ("foo", codec.name);
- EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, codec.kind);
+ EXPECT_EQ(webrtc::MediaType::AUDIO, codec.kind);
EXPECT_EQ(50, codec.preferred_payload_type);
EXPECT_EQ(22222, codec.clock_rate);
EXPECT_EQ(4, codec.num_channels);
@@ -108,7 +108,7 @@
RtpCodecCapability codec = ToRtpCodecCapability(cricket_codec);
EXPECT_EQ("VID", codec.name);
- EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, codec.kind);
+ EXPECT_EQ(webrtc::MediaType::VIDEO, codec.kind);
EXPECT_EQ(101, codec.preferred_payload_type);
EXPECT_EQ(80000, codec.clock_rate);
ASSERT_EQ(2u, codec.parameters.size());
diff --git a/pc/rtp_receiver_proxy.h b/pc/rtp_receiver_proxy.h
index feb569b..55c8d9a 100644
--- a/pc/rtp_receiver_proxy.h
+++ b/pc/rtp_receiver_proxy.h
@@ -38,7 +38,7 @@
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
streams)
-BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
+BYPASS_PROXY_CONSTMETHOD0(webrtc::MediaType, media_type)
BYPASS_PROXY_CONSTMETHOD0(std::string, id)
PROXY_SECONDARY_CONSTMETHOD0(RtpParameters, GetParameters)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index 9de5129..5a1cb7f 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -367,8 +367,8 @@
// ObserverInterface implementation.
void OnChanged() override;
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_AUDIO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::AUDIO;
}
std::string track_kind() const override {
return MediaStreamTrackInterface::kAudioKind;
@@ -429,8 +429,8 @@
// ObserverInterface implementation
void OnChanged() override;
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_VIDEO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::VIDEO;
}
std::string track_kind() const override {
return MediaStreamTrackInterface::kVideoKind;
diff --git a/pc/rtp_sender_proxy.h b/pc/rtp_sender_proxy.h
index 69e2e86..ec63186 100644
--- a/pc/rtp_sender_proxy.h
+++ b/pc/rtp_sender_proxy.h
@@ -29,7 +29,7 @@
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
PROXY_CONSTMETHOD0(uint32_t, ssrc)
-BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
+BYPASS_PROXY_CONSTMETHOD0(webrtc::MediaType, media_type)
BYPASS_PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
PROXY_CONSTMETHOD0(std::vector<RtpEncodingParameters>, init_send_encodings)
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index f18d61b..4c725d4 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -115,7 +115,7 @@
} // namespace
-RtpTransceiver::RtpTransceiver(cricket::MediaType media_type,
+RtpTransceiver::RtpTransceiver(webrtc::MediaType media_type,
ConnectionContext* context,
cricket::CodecLookupHelper* codec_lookup_helper)
: thread_(GetCurrentTaskQueueOrThread()),
@@ -123,8 +123,8 @@
media_type_(media_type),
context_(context),
codec_lookup_helper_(codec_lookup_helper) {
- RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO);
RTC_DCHECK(context_);
RTC_DCHECK(codec_lookup_helper_);
}
@@ -146,11 +146,11 @@
std::move(header_extensions_to_negotiate)),
on_negotiation_needed_(std::move(on_negotiation_needed)) {
RTC_DCHECK(context_);
- RTC_DCHECK(media_type_ == cricket::MEDIA_TYPE_AUDIO ||
- media_type_ == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(media_type_ == webrtc::MediaType::AUDIO ||
+ media_type_ == webrtc::MediaType::VIDEO);
RTC_DCHECK_EQ(sender->media_type(), receiver->media_type());
sender->internal()->SetSendCodecs(
- sender->media_type() == cricket::MEDIA_TYPE_VIDEO
+ sender->media_type() == webrtc::MediaType::VIDEO
? codec_vendor().video_send_codecs().codecs()
: codec_vendor().audio_send_codecs().codecs());
senders_.push_back(sender);
@@ -218,7 +218,7 @@
}
std::unique_ptr<cricket::ChannelInterface> new_channel;
- if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type() == webrtc::MediaType::AUDIO) {
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
// the worker thread. We shouldn't be using the `call_ptr_` hack here but
// simply be on the worker thread and use `call_` (update upstream code).
@@ -254,7 +254,7 @@
context()->ssrc_generator());
});
} else {
- RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, media_type());
+ RTC_DCHECK_EQ(webrtc::MediaType::VIDEO, media_type());
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
// the worker thread. We shouldn't be using the `call_ptr_` hack here but
@@ -405,7 +405,7 @@
RTC_DCHECK(!absl::c_linear_search(senders_, sender));
std::vector<cricket::Codec> send_codecs =
- media_type() == cricket::MEDIA_TYPE_VIDEO
+ media_type() == webrtc::MediaType::VIDEO
? codec_vendor().video_send_codecs().codecs()
: codec_vendor().audio_send_codecs().codecs();
sender->internal()->SetSendCodecs(send_codecs);
@@ -472,7 +472,7 @@
return rtc::scoped_refptr<RtpReceiverInternal>(receivers_[0]->internal());
}
-cricket::MediaType RtpTransceiver::media_type() const {
+webrtc::MediaType RtpTransceiver::media_type() const {
return media_type_;
}
@@ -682,10 +682,10 @@
const std::vector<RtpCodecCapability>& codecs) {
// Get codec capabilities from media engine.
std::vector<cricket::Codec> send_codecs, recv_codecs;
- if (media_type_ == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type_ == webrtc::MediaType::AUDIO) {
send_codecs = codec_vendor().audio_send_codecs().codecs();
recv_codecs = codec_vendor().audio_recv_codecs().codecs();
- } else if (media_type_ == cricket::MEDIA_TYPE_VIDEO) {
+ } else if (media_type_ == webrtc::MediaType::VIDEO) {
send_codecs = codec_vendor().video_send_codecs().codecs();
recv_codecs = codec_vendor().video_recv_codecs().codecs();
}
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index c8a16ff..71db88c 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -92,7 +92,7 @@
// channel set.
// `media_type` specifies the type of RtpTransceiver (and, by transitivity,
// the type of senders, receivers, and channel). Can either by audio or video.
- RtpTransceiver(cricket::MediaType media_type,
+ RtpTransceiver(webrtc::MediaType media_type,
ConnectionContext* context,
cricket::CodecLookupHelper* codec_lookup_helper);
// Construct a Unified Plan-style RtpTransceiver with the given sender and
@@ -264,7 +264,7 @@
void StopTransceiverProcedure();
// RtpTransceiverInterface implementation.
- cricket::MediaType media_type() const override;
+ webrtc::MediaType media_type() const override;
std::optional<std::string> mid() const override;
rtc::scoped_refptr<RtpSenderInterface> sender() const override;
rtc::scoped_refptr<RtpReceiverInterface> receiver() const override;
@@ -332,7 +332,7 @@
// Enforce that this object is created, used and destroyed on one thread.
TaskQueueBase* const thread_;
const bool unified_plan_;
- const cricket::MediaType media_type_;
+ const webrtc::MediaType media_type_;
rtc::scoped_refptr<PendingTaskSafetyFlag> signaling_thread_safety_;
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
senders_;
@@ -376,7 +376,7 @@
BEGIN_PRIMARY_PROXY_MAP(RtpTransceiver)
PROXY_PRIMARY_THREAD_DESTRUCTOR()
-BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
+BYPASS_PROXY_CONSTMETHOD0(webrtc::MediaType, media_type)
PROXY_CONSTMETHOD0(std::optional<std::string>, mid)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver)
diff --git a/pc/rtp_transceiver_unittest.cc b/pc/rtp_transceiver_unittest.cc
index 6993d56..0ee66b9 100644
--- a/pc/rtp_transceiver_unittest.cc
+++ b/pc/rtp_transceiver_unittest.cc
@@ -108,10 +108,10 @@
TEST_F(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) {
const std::string content_name("my_mid");
auto transceiver = rtc::make_ref_counted<RtpTransceiver>(
- cricket::MediaType::MEDIA_TYPE_AUDIO, context(), codec_lookup_helper());
+ webrtc::MediaType::AUDIO, context(), codec_lookup_helper());
auto channel1 = std::make_unique<NiceMock<MockChannelInterface>>();
EXPECT_CALL(*channel1, media_type())
- .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
+ .WillRepeatedly(Return(webrtc::MediaType::AUDIO));
EXPECT_CALL(*channel1, mid()).WillRepeatedly(ReturnRef(content_name));
EXPECT_CALL(*channel1, SetFirstPacketReceivedCallback(_));
EXPECT_CALL(*channel1, SetRtpTransport(_)).WillRepeatedly(Return(true));
@@ -128,7 +128,7 @@
auto channel2 = std::make_unique<NiceMock<MockChannelInterface>>();
EXPECT_CALL(*channel2, media_type())
- .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
+ .WillRepeatedly(Return(webrtc::MediaType::AUDIO));
// Clear the current channel - required to allow SetChannel()
EXPECT_CALL(*channel1_ptr, SetFirstPacketReceivedCallback(_));
@@ -144,10 +144,10 @@
TEST_F(RtpTransceiverTest, CanUnsetChannelOnStoppedTransceiver) {
const std::string content_name("my_mid");
auto transceiver = rtc::make_ref_counted<RtpTransceiver>(
- cricket::MediaType::MEDIA_TYPE_VIDEO, context(), codec_lookup_helper());
+ webrtc::MediaType::VIDEO, context(), codec_lookup_helper());
auto channel = std::make_unique<NiceMock<MockChannelInterface>>();
EXPECT_CALL(*channel, media_type())
- .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_VIDEO));
+ .WillRepeatedly(Return(webrtc::MediaType::VIDEO));
EXPECT_CALL(*channel, mid()).WillRepeatedly(ReturnRef(content_name));
EXPECT_CALL(*channel, SetFirstPacketReceivedCallback(_))
.WillRepeatedly(testing::Return());
@@ -172,7 +172,7 @@
class RtpTransceiverUnifiedPlanTest : public RtpTransceiverTest {
public:
static rtc::scoped_refptr<MockRtpReceiverInternal> MockReceiver(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
auto receiver = rtc::make_ref_counted<NiceMock<MockRtpReceiverInternal>>();
EXPECT_CALL(*receiver.get(), media_type())
.WillRepeatedly(Return(media_type));
@@ -180,7 +180,7 @@
}
static rtc::scoped_refptr<MockRtpSenderInternal> MockSender(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
auto sender = rtc::make_ref_counted<NiceMock<MockRtpSenderInternal>>();
EXPECT_CALL(*sender.get(), media_type()).WillRepeatedly(Return(media_type));
return sender;
@@ -206,9 +206,9 @@
// Basic tests for Stop()
TEST_F(RtpTransceiverUnifiedPlanTest, StopSetsDirection) {
rtc::scoped_refptr<MockRtpReceiverInternal> receiver =
- MockReceiver(cricket::MediaType::MEDIA_TYPE_AUDIO);
+ MockReceiver(webrtc::MediaType::AUDIO);
rtc::scoped_refptr<MockRtpSenderInternal> sender =
- MockSender(cricket::MediaType::MEDIA_TYPE_AUDIO);
+ MockSender(webrtc::MediaType::AUDIO);
rtc::scoped_refptr<RtpTransceiver> transceiver =
CreateTransceiver(sender, receiver);
@@ -233,9 +233,9 @@
: public RtpTransceiverUnifiedPlanTest {
public:
RtpTransceiverFilteredCodecPreferencesTest()
- : transceiver_(CreateTransceiver(
- MockSender(cricket::MediaType::MEDIA_TYPE_VIDEO),
- MockReceiver(cricket::MediaType::MEDIA_TYPE_VIDEO))) {}
+ : transceiver_(
+ CreateTransceiver(MockSender(webrtc::MediaType::VIDEO),
+ MockReceiver(webrtc::MediaType::VIDEO))) {}
struct H264CodecCapabilities {
cricket::Codec cricket_sendrecv_codec;
@@ -253,9 +253,8 @@
// at transceiver create time.
void RecreateTransceiver() {
fake_codec_lookup_helper()->Reset();
- transceiver_ =
- CreateTransceiver(MockSender(cricket::MediaType::MEDIA_TYPE_VIDEO),
- MockReceiver(cricket::MediaType::MEDIA_TYPE_VIDEO));
+ transceiver_ = CreateTransceiver(MockSender(webrtc::MediaType::VIDEO),
+ MockReceiver(webrtc::MediaType::VIDEO));
}
// For H264, the profile and level IDs are entangled. This function uses
@@ -599,9 +598,9 @@
}
rtc::scoped_refptr<MockRtpReceiverInternal> receiver_ =
- MockReceiver(cricket::MediaType::MEDIA_TYPE_AUDIO);
+ MockReceiver(webrtc::MediaType::AUDIO);
rtc::scoped_refptr<MockRtpSenderInternal> sender_ =
- MockSender(cricket::MediaType::MEDIA_TYPE_AUDIO);
+ MockSender(webrtc::MediaType::AUDIO);
std::vector<RtpHeaderExtensionCapability> extensions_;
rtc::scoped_refptr<RtpTransceiver> transceiver_;
@@ -747,7 +746,7 @@
auto mock_channel_ptr = mock_channel.get();
EXPECT_CALL(*mock_channel, SetFirstPacketReceivedCallback(_));
EXPECT_CALL(*mock_channel, media_type())
- .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
+ .WillRepeatedly(Return(webrtc::MediaType::AUDIO));
EXPECT_CALL(*mock_channel, voice_media_send_channel())
.WillRepeatedly(Return(nullptr));
EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name));
@@ -780,7 +779,7 @@
auto mock_channel_ptr = mock_channel.get();
EXPECT_CALL(*mock_channel, SetFirstPacketReceivedCallback(_));
EXPECT_CALL(*mock_channel, media_type())
- .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
+ .WillRepeatedly(Return(webrtc::MediaType::AUDIO));
EXPECT_CALL(*mock_channel, voice_media_send_channel())
.WillRepeatedly(Return(nullptr));
EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name));
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index a823e18..9b4a0e1 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -173,10 +173,10 @@
if (adjusted_stream_ids.empty()) {
adjusted_stream_ids.push_back(CreateRandomUuid());
}
- cricket::MediaType media_type =
+ webrtc::MediaType media_type =
(track->kind() == MediaStreamTrackInterface::kAudioKind
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO);
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO);
auto new_sender = CreateSender(
media_type, track->id(), track, adjusted_stream_ids,
init_send_encodings
@@ -214,7 +214,7 @@
FindFirstTransceiverForAddedTrack(track, init_send_encodings);
if (transceiver) {
RTC_LOG(LS_INFO) << "Reusing an existing "
- << cricket::MediaTypeToString(transceiver->media_type())
+ << webrtc::MediaTypeToString(transceiver->media_type())
<< " transceiver for AddTrack.";
if (transceiver->stopping()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
@@ -232,11 +232,11 @@
transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
transceiver->internal()->set_reused_for_addtrack(true);
} else {
- cricket::MediaType media_type =
+ webrtc::MediaType media_type =
(track->kind() == MediaStreamTrackInterface::kAudioKind
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO);
- RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO);
+ RTC_LOG(LS_INFO) << "Adding " << webrtc::MediaTypeToString(media_type)
<< " transceiver in response to a call to AddTrack.";
std::string sender_id = track->id();
// Avoid creating a sender with an existing ID by generating a random ID.
@@ -260,14 +260,14 @@
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
RtpTransmissionManager::CreateSender(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::string& id,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids,
const std::vector<RtpEncodingParameters>& send_encodings) {
RTC_DCHECK_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
RTC_DCHECK(!track ||
(track->kind() == MediaStreamTrackInterface::kAudioKind));
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
@@ -275,7 +275,7 @@
AudioRtpSender::Create(env_, worker_thread(), id, legacy_stats_, this));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
- RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK_EQ(media_type, webrtc::MediaType::VIDEO);
RTC_DCHECK(!track ||
(track->kind() == MediaStreamTrackInterface::kVideoKind));
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
@@ -291,12 +291,12 @@
}
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
-RtpTransmissionManager::CreateReceiver(cricket::MediaType media_type,
+RtpTransmissionManager::CreateReceiver(webrtc::MediaType media_type,
const std::string& receiver_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver;
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), worker_thread(),
rtc::make_ref_counted<AudioRtpReceiver>(worker_thread(), receiver_id,
@@ -304,7 +304,7 @@
IsUnifiedPlan()));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
- RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK_EQ(media_type, webrtc::MediaType::VIDEO);
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), worker_thread(),
rtc::make_ref_counted<VideoRtpReceiver>(worker_thread(), receiver_id,
@@ -329,7 +329,7 @@
signaling_thread(),
rtc::make_ref_counted<RtpTransceiver>(
sender, receiver, context_, codec_lookup_helper_,
- sender->media_type() == cricket::MEDIA_TYPE_AUDIO
+ sender->media_type() == webrtc::MediaType::AUDIO
? media_engine()->voice().GetRtpHeaderExtensions()
: media_engine()->video().GetRtpHeaderExtensions(),
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr()]() {
@@ -352,8 +352,7 @@
}
for (auto transceiver : transceivers()->List()) {
if (!transceiver->sender()->track() &&
- cricket::MediaTypeToString(transceiver->media_type()) ==
- track->kind() &&
+ webrtc::MediaTypeToString(transceiver->media_type()) == track->kind() &&
!transceiver->internal()->has_ever_been_used_to_send() &&
!transceiver->stopped()) {
return transceiver;
@@ -402,7 +401,7 @@
// audio/video transceiver.
RTC_DCHECK(!IsUnifiedPlan());
for (auto transceiver : transceivers_.List()) {
- if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (transceiver->media_type() == webrtc::MediaType::AUDIO) {
return transceiver;
}
}
@@ -417,7 +416,7 @@
// audio/video transceiver.
RTC_DCHECK(!IsUnifiedPlan());
for (auto transceiver : transceivers_.List()) {
- if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
+ if (transceiver->media_type() == webrtc::MediaType::VIDEO) {
return transceiver;
}
}
@@ -439,7 +438,7 @@
}
// Normal case; we've never seen this track before.
- auto new_sender = CreateSender(cricket::MEDIA_TYPE_AUDIO, track->id(),
+ auto new_sender = CreateSender(webrtc::MediaType::AUDIO, track->id(),
rtc::scoped_refptr<AudioTrackInterface>(track),
{stream->id()}, {{}});
new_sender->internal()->SetMediaChannel(voice_media_send_channel());
@@ -486,7 +485,7 @@
}
// Normal case; we've never seen this track before.
- auto new_sender = CreateSender(cricket::MEDIA_TYPE_VIDEO, track->id(),
+ auto new_sender = CreateSender(webrtc::MediaType::VIDEO, track->id(),
rtc::scoped_refptr<VideoTrackInterface>(track),
{stream->id()}, {{}});
new_sender->internal()->SetMediaChannel(video_media_send_channel());
@@ -570,7 +569,7 @@
<< remote_sender_info.sender_id << " doesn't exist.";
return nullptr;
}
- if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (receiver->media_type() == webrtc::MediaType::AUDIO) {
GetAudioTransceiver()->internal()->RemoveReceiver(receiver.get());
} else {
GetVideoTransceiver()->internal()->RemoveReceiver(receiver.get());
@@ -581,15 +580,15 @@
void RtpTransmissionManager::OnRemoteSenderAdded(
const RtpSenderInfo& sender_info,
MediaStreamInterface* stream,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
- RTC_LOG(LS_INFO) << "Creating " << cricket::MediaTypeToString(media_type)
+ RTC_LOG(LS_INFO) << "Creating " << webrtc::MediaTypeToString(media_type)
<< " receiver for track_id=" << sender_info.sender_id
<< " and stream_id=" << sender_info.stream_id;
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
CreateAudioReceiver(stream, sender_info);
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ } else if (media_type == webrtc::MediaType::VIDEO) {
CreateVideoReceiver(stream, sender_info);
} else {
RTC_DCHECK_NOTREACHED() << "Invalid media type";
@@ -599,14 +598,14 @@
void RtpTransmissionManager::OnRemoteSenderRemoved(
const RtpSenderInfo& sender_info,
MediaStreamInterface* stream,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
- RTC_LOG(LS_INFO) << "Removing " << cricket::MediaTypeToString(media_type)
+ RTC_LOG(LS_INFO) << "Removing " << webrtc::MediaTypeToString(media_type)
<< " receiver for track_id=" << sender_info.sender_id
<< " and stream_id=" << sender_info.stream_id;
rtc::scoped_refptr<RtpReceiverInterface> receiver;
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
// will be notified which will end the AudioRtpReceiver::track().
receiver = RemoveAndStopReceiver(sender_info);
@@ -615,7 +614,7 @@
if (audio_track) {
stream->RemoveTrack(audio_track);
}
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ } else if (media_type == webrtc::MediaType::VIDEO) {
// Stopping or destroying a VideoRtpReceiver will end the
// VideoRtpReceiver::track().
receiver = RemoveAndStopReceiver(sender_info);
@@ -637,7 +636,7 @@
void RtpTransmissionManager::OnLocalSenderAdded(
const RtpSenderInfo& sender_info,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsUnifiedPlan());
auto sender = FindSenderById(sender_info.sender_id);
@@ -660,7 +659,7 @@
void RtpTransmissionManager::OnLocalSenderRemoved(
const RtpSenderInfo& sender_info,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
auto sender = FindSenderById(sender_info.sender_id);
if (!sender) {
@@ -682,20 +681,19 @@
}
std::vector<RtpSenderInfo>* RtpTransmissionManager::GetRemoteSenderInfos(
- cricket::MediaType media_type) {
- RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO);
- return (media_type == cricket::MEDIA_TYPE_AUDIO)
- ? &remote_audio_sender_infos_
- : &remote_video_sender_infos_;
+ webrtc::MediaType media_type) {
+ RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO);
+ return (media_type == webrtc::MediaType::AUDIO) ? &remote_audio_sender_infos_
+ : &remote_video_sender_infos_;
}
std::vector<RtpSenderInfo>* RtpTransmissionManager::GetLocalSenderInfos(
- cricket::MediaType media_type) {
- RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO);
- return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_sender_infos_
- : &local_video_sender_infos_;
+ webrtc::MediaType media_type) {
+ RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO);
+ return (media_type == webrtc::MediaType::AUDIO) ? &local_audio_sender_infos_
+ : &local_video_sender_infos_;
}
const RtpSenderInfo* RtpTransmissionManager::FindSenderInfo(
diff --git a/pc/rtp_transmission_manager.h b/pc/rtp_transmission_manager.h
index 36bd3fd..a6b2669 100644
--- a/pc/rtp_transmission_manager.h
+++ b/pc/rtp_transmission_manager.h
@@ -97,7 +97,7 @@
// Create a new RTP sender. Does not associate with a transceiver.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
- CreateSender(cricket::MediaType media_type,
+ CreateSender(webrtc::MediaType media_type,
const std::string& id,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids,
@@ -105,7 +105,7 @@
// Create a new RTP receiver. Does not associate with a transceiver.
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
- CreateReceiver(cricket::MediaType media_type, const std::string& receiver_id);
+ CreateReceiver(webrtc::MediaType media_type, const std::string& receiver_id);
// Create a new RtpTransceiver of the given type and add it to the list of
// registered transceivers.
@@ -155,14 +155,14 @@
// implementation and triggers CreateAudioReceiver or CreateVideoReceiver.
void OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
MediaStreamInterface* stream,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
// Triggered when a remote sender has been removed from a remote session
// description. It removes the remote sender with id `sender_id` from a remote
// MediaStream and triggers DestroyAudioReceiver or DestroyVideoReceiver.
void OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
MediaStreamInterface* stream,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
// Triggered when a local sender has been seen for the first time in a local
// session description.
@@ -170,7 +170,7 @@
// streams in the local SessionDescription can be mapped to a MediaStreamTrack
// in a MediaStream in `local_streams_`
void OnLocalSenderAdded(const RtpSenderInfo& sender_info,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
// Triggered when a local sender has been removed from a local session
// description.
@@ -178,12 +178,11 @@
// has been removed from the local SessionDescription and the stream can be
// mapped to a MediaStreamTrack in a MediaStream in `local_streams_`.
void OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
std::vector<RtpSenderInfo>* GetRemoteSenderInfos(
- cricket::MediaType media_type);
- std::vector<RtpSenderInfo>* GetLocalSenderInfos(
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
+ std::vector<RtpSenderInfo>* GetLocalSenderInfos(webrtc::MediaType media_type);
const RtpSenderInfo* FindSenderInfo(const std::vector<RtpSenderInfo>& infos,
const std::string& stream_id,
const std::string& sender_id) const;
diff --git a/pc/sdp_munging_detector.cc b/pc/sdp_munging_detector.cc
index 8506d0b..ef841b1 100644
--- a/pc/sdp_munging_detector.cc
+++ b/pc/sdp_munging_detector.cc
@@ -399,14 +399,14 @@
continue;
}
// Validate video and audio contents.
- cricket::MediaType media_type = last_created_media_description->type();
- if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ webrtc::MediaType media_type = last_created_media_description->type();
+ if (media_type == webrtc::MediaType::VIDEO) {
type = DetermineVideoSdpMungingType(last_created_media_description,
media_description_to_set);
if (type != SdpMungingType::kNoModification) {
return type;
}
- } else if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ } else if (media_type == webrtc::MediaType::AUDIO) {
type = DetermineAudioSdpMungingType(last_created_media_description,
media_description_to_set);
if (type != SdpMungingType::kNoModification) {
@@ -426,7 +426,7 @@
// Codec position swapped.
for (size_t j = i + 1; j < last_created_codecs.size(); j++) {
if (last_created_codecs[i] == codecs_to_set[j]) {
- return media_type == cricket::MEDIA_TYPE_AUDIO
+ return media_type == webrtc::MediaType::AUDIO
? SdpMungingType::kAudioCodecsReordered
: SdpMungingType::kVideoCodecsReordered;
}
@@ -437,13 +437,13 @@
return SdpMungingType::kPayloadTypes;
}
if (last_created_codecs[i].params != codecs_to_set[i].params) {
- return media_type == cricket::MEDIA_TYPE_AUDIO
+ return media_type == webrtc::MediaType::AUDIO
? SdpMungingType::kAudioCodecsFmtp
: SdpMungingType::kVideoCodecsFmtp;
}
if (last_created_codecs[i].feedback_params !=
codecs_to_set[i].feedback_params) {
- return media_type == cricket::MEDIA_TYPE_AUDIO
+ return media_type == webrtc::MediaType::AUDIO
? SdpMungingType::kAudioCodecsRtcpFb
: SdpMungingType::kVideoCodecsRtcpFb;
}
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index d16a71f..8f836da 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -458,8 +458,8 @@
continue;
}
const auto type = media_description->type();
- if (type == cricket::MEDIA_TYPE_AUDIO ||
- type == cricket::MEDIA_TYPE_VIDEO) {
+ if (type == webrtc::MediaType::AUDIO ||
+ type == webrtc::MediaType::VIDEO) {
for (const auto& c : media_description->codecs()) {
auto error = FindDuplicateCodecParameters(
c.ToCodecParameters(), payload_to_codec_parameters);
@@ -591,8 +591,7 @@
continue;
}
const auto type = media_description->type();
- if (type == cricket::MEDIA_TYPE_AUDIO ||
- type == cricket::MEDIA_TYPE_VIDEO) {
+ if (type == webrtc::MediaType::AUDIO || type == webrtc::MediaType::VIDEO) {
for (const auto& codec : media_description->codecs()) {
if (!PayloadType::IsValid(codec.id, media_description->rtcp_mux())) {
LOG_AND_RETURN_ERROR(
@@ -719,15 +718,15 @@
// The SDP parser used to populate these values by default for the 'content
// name' if an a=mid line was absent.
-absl::string_view GetDefaultMidForPlanB(cricket::MediaType media_type) {
+absl::string_view GetDefaultMidForPlanB(webrtc::MediaType media_type) {
switch (media_type) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
return cricket::CN_AUDIO;
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
return cricket::CN_VIDEO;
- case cricket::MEDIA_TYPE_DATA:
+ case webrtc::MediaType::DATA:
return cricket::CN_DATA;
- case cricket::MEDIA_TYPE_UNSUPPORTED:
+ case webrtc::MediaType::UNSUPPORTED:
return "not supported";
default:
// Fall through to RTC_CHECK_NOTREACHED
@@ -745,13 +744,13 @@
cricket::MediaDescriptionOptions* video_media_description_options,
int num_sim_layers) {
for (const auto& sender : senders) {
- if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (sender->media_type() == webrtc::MediaType::AUDIO) {
if (audio_media_description_options) {
audio_media_description_options->AddAudioSender(
sender->id(), sender->internal()->stream_ids());
}
} else {
- RTC_DCHECK(sender->media_type() == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(sender->media_type() == webrtc::MediaType::VIDEO);
if (video_media_description_options) {
video_media_description_options->AddVideoSender(
sender->id(), sender->internal()->stream_ids(), {},
@@ -1947,7 +1946,7 @@
GetFirstAudioContent(local_description()->description());
if (audio_content) {
if (audio_content->rejected) {
- RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
+ RemoveSenders(webrtc::MediaType::AUDIO);
} else {
const MediaContentDescription* audio_desc =
audio_content->media_description();
@@ -1959,7 +1958,7 @@
GetFirstVideoContent(local_description()->description());
if (video_content) {
if (video_content->rejected) {
- RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
+ RemoveSenders(webrtc::MediaType::VIDEO);
} else {
const MediaContentDescription* video_desc =
video_content->media_description();
@@ -2334,7 +2333,7 @@
// and MediaStreams.
if (audio_content) {
if (audio_content->rejected) {
- RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
+ RemoveSenders(webrtc::MediaType::AUDIO);
} else {
bool default_audio_track_needed =
!remote_peer_supports_msid_ &&
@@ -2349,7 +2348,7 @@
// and MediaStreams.
if (video_content) {
if (video_content->rejected) {
- RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
+ RemoveSenders(webrtc::MediaType::VIDEO);
} else {
bool default_video_track_needed =
!remote_peer_supports_msid_ &&
@@ -3784,8 +3783,8 @@
// media section.
for (const ContentInfo& content : sdesc->description()->contents()) {
const MediaContentDescription& desc = *content.media_description();
- if ((desc.type() == cricket::MEDIA_TYPE_AUDIO ||
- desc.type() == cricket::MEDIA_TYPE_VIDEO) &&
+ if ((desc.type() == webrtc::MediaType::AUDIO ||
+ desc.type() == webrtc::MediaType::VIDEO) &&
desc.streams().size() > 1u) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
@@ -3832,13 +3831,13 @@
const ContentInfos& new_contents = new_session.description()->contents();
for (size_t i = 0; i < new_contents.size(); ++i) {
const ContentInfo& new_content = new_contents[i];
- cricket::MediaType media_type = new_content.media_description()->type();
+ webrtc::MediaType media_type = new_content.media_description()->type();
mid_generator_.AddKnownId(new_content.mid());
auto it = bundle_groups_by_mid.find(new_content.mid());
const ContentGroup* bundle_group =
it != bundle_groups_by_mid.end() ? it->second : nullptr;
- if (media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO) {
const ContentInfo* old_local_content = nullptr;
if (old_local_description &&
i < old_local_description->description()->contents().size()) {
@@ -3899,7 +3898,7 @@
if (!error.ok()) {
return error;
}
- } else if (media_type == cricket::MEDIA_TYPE_DATA) {
+ } else if (media_type == webrtc::MediaType::DATA) {
const auto data_mid = pc_->sctp_mid();
if (data_mid && new_content.mid() != data_mid.value()) {
// Ignore all but the first data section.
@@ -3912,7 +3911,7 @@
if (!error.ok()) {
return error;
}
- } else if (media_type == cricket::MEDIA_TYPE_UNSUPPORTED) {
+ } else if (media_type == webrtc::MediaType::UNSUPPORTED) {
RTC_LOG(LS_INFO) << "Ignoring unsupported media type";
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
@@ -3978,7 +3977,7 @@
// recvonly direction.
if (!transceiver) {
RTC_LOG(LS_INFO) << "Adding "
- << cricket::MediaTypeToString(media_desc->type())
+ << webrtc::MediaTypeToString(media_desc->type())
<< " transceiver for MID=" << content.mid()
<< " at i=" << mline_index
<< " in response to the remote description.";
@@ -4178,7 +4177,7 @@
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
SdpOfferAnswerHandler::FindAvailableTransceiverToReceive(
- cricket::MediaType media_type) const {
+ webrtc::MediaType media_type) const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
// From JSEP section 5.10 (Applying a Remote Description):
@@ -4309,7 +4308,7 @@
// Add audio/video/data m= sections to the end if needed.
if (!audio_index && offer_new_audio_description) {
cricket::MediaDescriptionOptions options(
- cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
+ webrtc::MediaType::AUDIO, cricket::CN_AUDIO,
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), false);
options.header_extensions =
media_engine()->voice().GetRtpHeaderExtensions();
@@ -4318,7 +4317,7 @@
}
if (!video_index && offer_new_video_description) {
cricket::MediaDescriptionOptions options(
- cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
+ webrtc::MediaType::VIDEO, cricket::CN_VIDEO,
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), false);
options.header_extensions =
media_engine()->video().GetRtpHeaderExtensions();
@@ -4381,11 +4380,11 @@
(current_remote_content && current_remote_content->rejected);
const std::string& mid =
(local_content ? local_content->mid() : remote_content->mid());
- cricket::MediaType media_type =
+ webrtc::MediaType media_type =
(local_content ? local_content->media_description()->type()
: remote_content->media_description()->type());
- if (media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO) {
// A media section is considered eligible for recycling if it is marked as
// rejected in either the current local or current remote description.
auto transceiver = transceivers()->FindByMid(mid);
@@ -4420,14 +4419,14 @@
transceiver->internal()->set_mline_index(i);
}
}
- } else if (media_type == cricket::MEDIA_TYPE_UNSUPPORTED) {
+ } else if (media_type == webrtc::MediaType::UNSUPPORTED) {
RTC_DCHECK(local_content->rejected);
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(media_type, mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
} else {
- RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
+ RTC_CHECK_EQ(webrtc::MediaType::DATA, media_type);
if (had_been_rejected) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForRejectedData(mid));
@@ -4485,7 +4484,7 @@
for (size_t i = 0; i < session_options->media_description_options.size();
i++) {
auto media_description = session_options->media_description_options[i];
- if (media_description.type == cricket::MEDIA_TYPE_DATA &&
+ if (media_description.type == webrtc::MediaType::DATA &&
media_description.stopped) {
session_options->media_description_options[i] =
GetMediaDescriptionOptionsForActiveData(media_description.mid);
@@ -4594,9 +4593,9 @@
RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
for (const ContentInfo& content :
remote_description()->description()->contents()) {
- cricket::MediaType media_type = content.media_description()->type();
- if (media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO) {
+ webrtc::MediaType media_type = content.media_description()->type();
+ if (media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO) {
auto transceiver = transceivers()->FindByMid(content.mid());
if (transceiver) {
session_options->media_description_options.push_back(
@@ -4611,14 +4610,14 @@
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
}
- } else if (media_type == cricket::MEDIA_TYPE_UNSUPPORTED) {
+ } else if (media_type == webrtc::MediaType::UNSUPPORTED) {
RTC_DCHECK(content.rejected);
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(media_type, content.mid(),
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
} else {
- RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
+ RTC_CHECK_EQ(webrtc::MediaType::DATA, media_type);
// Reject all data sections if data channels are disabled.
// Reject a data section if it has already been rejected.
// Reject all data sections except for the first one.
@@ -4673,9 +4672,9 @@
if (options.offer_to_receive_audio == 0) {
RemoveRecvDirectionFromReceivingTransceiversOfType(
- cricket::MEDIA_TYPE_AUDIO);
+ webrtc::MediaType::AUDIO);
} else if (options.offer_to_receive_audio == 1) {
- AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_AUDIO);
+ AddUpToOneReceivingTransceiverOfType(webrtc::MediaType::AUDIO);
} else if (options.offer_to_receive_audio > 1) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"offer_to_receive_audio > 1 is not supported.");
@@ -4683,9 +4682,9 @@
if (options.offer_to_receive_video == 0) {
RemoveRecvDirectionFromReceivingTransceiversOfType(
- cricket::MEDIA_TYPE_VIDEO);
+ webrtc::MediaType::VIDEO);
} else if (options.offer_to_receive_video == 1) {
- AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
+ AddUpToOneReceivingTransceiverOfType(webrtc::MediaType::VIDEO);
} else if (options.offer_to_receive_video > 1) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"offer_to_receive_video > 1 is not supported.");
@@ -4695,12 +4694,12 @@
}
void SdpOfferAnswerHandler::RemoveRecvDirectionFromReceivingTransceiversOfType(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) {
RtpTransceiverDirection new_direction =
RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
if (new_direction != transceiver->direction()) {
- RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
+ RTC_LOG(LS_INFO) << "Changing " << webrtc::MediaTypeToString(media_type)
<< " transceiver (MID="
<< transceiver->mid().value_or("<not set>") << ") from "
<< RtpTransceiverDirectionToString(
@@ -4714,11 +4713,11 @@
}
void SdpOfferAnswerHandler::AddUpToOneReceivingTransceiverOfType(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (GetReceivingTransceiversOfType(media_type).empty()) {
RTC_LOG(LS_INFO)
- << "Adding one recvonly " << cricket::MediaTypeToString(media_type)
+ << "Adding one recvonly " << webrtc::MediaTypeToString(media_type)
<< " transceiver since CreateOffer specified offer_to_receive=1";
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kRecvOnly;
@@ -4729,7 +4728,7 @@
std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
SdpOfferAnswerHandler::GetReceivingTransceiversOfType(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
receiving_transceivers;
@@ -4774,7 +4773,7 @@
}
}
-void SdpOfferAnswerHandler::RemoveSenders(cricket::MediaType media_type) {
+void SdpOfferAnswerHandler::RemoveSenders(webrtc::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
UpdateLocalSenders(std::vector<cricket::StreamParams>(), media_type);
UpdateRemoteSendersList(std::vector<cricket::StreamParams>(), false,
@@ -4783,7 +4782,7 @@
void SdpOfferAnswerHandler::UpdateLocalSenders(
const std::vector<cricket::StreamParams>& streams,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateLocalSenders");
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<RtpSenderInfo>* current_senders =
@@ -4825,7 +4824,7 @@
void SdpOfferAnswerHandler::UpdateRemoteSendersList(
const cricket::StreamParamsVec& streams,
bool default_sender_needed,
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
StreamCollection* new_streams) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateRemoteSendersList");
RTC_DCHECK_RUN_ON(signaling_thread());
@@ -4912,7 +4911,7 @@
remote_streams_->AddStream(default_stream);
new_streams->AddStream(default_stream);
}
- std::string default_sender_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
+ std::string default_sender_id = (media_type == webrtc::MediaType::AUDIO)
? kDefaultAudioSenderId
: kDefaultVideoSenderId;
const RtpSenderInfo* default_sender_info = rtp_manager()->FindSenderInfo(
@@ -5352,12 +5351,12 @@
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
for (const auto& transceiver : list) {
- if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
+ if (transceiver->media_type() == webrtc::MediaType::VIDEO) {
transceiver->internal()->ClearChannel();
}
}
for (const auto& transceiver : list) {
- if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (transceiver->media_type() == webrtc::MediaType::AUDIO) {
transceiver->internal()->ClearChannel();
}
}
@@ -5379,12 +5378,12 @@
if (*audio_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
- cricket::MEDIA_TYPE_AUDIO, content.mid(),
+ webrtc::MediaType::AUDIO, content.mid(),
RtpTransceiverDirection::kInactive, /*stopped=*/true));
} else {
bool stopped = (audio_direction == RtpTransceiverDirection::kInactive);
session_options->media_description_options.push_back(
- cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO,
+ cricket::MediaDescriptionOptions(webrtc::MediaType::AUDIO,
content.mid(), audio_direction,
stopped));
*audio_index = session_options->media_description_options.size() - 1;
@@ -5396,12 +5395,12 @@
if (*video_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
- cricket::MEDIA_TYPE_VIDEO, content.mid(),
+ webrtc::MediaType::VIDEO, content.mid(),
RtpTransceiverDirection::kInactive, /*stopped=*/true));
} else {
bool stopped = (video_direction == RtpTransceiverDirection::kInactive);
session_options->media_description_options.push_back(
- cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO,
+ cricket::MediaDescriptionOptions(webrtc::MediaType::VIDEO,
content.mid(), video_direction,
stopped));
*video_index = session_options->media_description_options.size() - 1;
@@ -5410,7 +5409,7 @@
media_engine()->video().GetRtpHeaderExtensions();
} else if (IsUnsupportedContent(&content)) {
session_options->media_description_options.push_back(
- cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_UNSUPPORTED,
+ cricket::MediaDescriptionOptions(webrtc::MediaType::UNSUPPORTED,
content.mid(),
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
@@ -5435,7 +5434,7 @@
RTC_DCHECK_RUN_ON(signaling_thread());
// Direction for data sections is meaningless, but legacy endpoints might
// expect sendrecv.
- cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
+ cricket::MediaDescriptionOptions options(webrtc::MediaType::DATA, mid,
RtpTransceiverDirection::kSendRecv,
/*stopped=*/false);
return options;
@@ -5445,7 +5444,7 @@
SdpOfferAnswerHandler::GetMediaDescriptionOptionsForRejectedData(
const std::string& mid) const {
RTC_DCHECK_RUN_ON(signaling_thread());
- cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
+ cricket::MediaDescriptionOptions options(webrtc::MediaType::DATA, mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true);
return options;
@@ -5498,15 +5497,15 @@
// Ignore transceivers that are not receiving.
continue;
}
- const cricket::MediaType media_type =
+ const webrtc::MediaType media_type =
content_info.media_description()->type();
- if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MediaType::MEDIA_TYPE_VIDEO) {
- if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO &&
+ if (media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO) {
+ if (media_type == webrtc::MediaType::AUDIO &&
!mid_header_extension_missing_audio) {
mid_header_extension_missing_audio =
!ContentHasHeaderExtension(content_info, RtpExtension::kMidUri);
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO &&
+ } else if (media_type == webrtc::MediaType::VIDEO &&
!mid_header_extension_missing_video) {
mid_header_extension_missing_video =
!ContentHasHeaderExtension(content_info, RtpExtension::kMidUri);
@@ -5514,16 +5513,16 @@
const MediaContentDescription* media_desc =
content_info.media_description();
for (const cricket::Codec& codec : media_desc->codecs()) {
- if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
if (payload_types->audio_payload_types.count(codec.id)) {
// Two m= sections are using the same payload type, thus demuxing
// by payload type is not possible.
- if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
payload_types->pt_demuxing_possible_audio = false;
}
}
payload_types->audio_payload_types.insert(codec.id);
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ } else if (media_type == webrtc::MediaType::VIDEO) {
if (payload_types->video_payload_types.count(codec.id)) {
// Two m= sections are using the same payload type, thus demuxing
// by payload type is not possible.
@@ -5565,9 +5564,9 @@
continue;
}
- const cricket::MediaType media_type = channel->media_type();
- if (media_type != cricket::MediaType::MEDIA_TYPE_AUDIO &&
- media_type != cricket::MediaType::MEDIA_TYPE_VIDEO) {
+ const webrtc::MediaType media_type = channel->media_type();
+ if (media_type != webrtc::MediaType::AUDIO &&
+ media_type != webrtc::MediaType::VIDEO) {
continue;
}
@@ -5581,7 +5580,7 @@
const ContentGroup* bundle_group =
bundle_it != bundle_groups_by_mid.end() ? bundle_it->second : nullptr;
bool pt_demux_enabled = RtpTransceiverDirectionHasRecv(local_direction);
- if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
pt_demux_enabled &=
!bundle_group ||
(bundled_pt_demux_allowed_audio &&
@@ -5590,7 +5589,7 @@
pt_demuxing_has_been_used_audio_ = true;
}
} else {
- RTC_DCHECK_EQ(media_type, cricket::MediaType::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK_EQ(media_type, webrtc::MediaType::VIDEO);
pt_demux_enabled &=
!bundle_group ||
(bundled_pt_demux_allowed_video &&
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index b7387da..a9c68e8 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -384,7 +384,7 @@
// Returns an RtpTransceiver, if available, that can be used to receive the
// given media type according to JSEP rules.
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- FindAvailableTransceiverToReceive(cricket::MediaType media_type) const;
+ FindAvailableTransceiverToReceive(webrtc::MediaType media_type) const;
// Returns a MediaSessionOptions struct with options decided by `options`,
// the local MediaStreams and DataChannels.
@@ -430,12 +430,12 @@
RTCError HandleLegacyOfferOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& options);
void RemoveRecvDirectionFromReceivingTransceiversOfType(
- cricket::MediaType media_type) RTC_RUN_ON(signaling_thread());
- void AddUpToOneReceivingTransceiverOfType(cricket::MediaType media_type);
+ webrtc::MediaType media_type) RTC_RUN_ON(signaling_thread());
+ void AddUpToOneReceivingTransceiverOfType(webrtc::MediaType media_type);
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
- GetReceivingTransceiversOfType(cricket::MediaType media_type)
+ GetReceivingTransceiversOfType(webrtc::MediaType media_type)
RTC_RUN_ON(signaling_thread());
// Runs the algorithm specified in
@@ -458,14 +458,14 @@
// Remove all local and remote senders of type `media_type`.
// Called when a media type is rejected (m-line set to port 0).
- void RemoveSenders(cricket::MediaType media_type);
+ void RemoveSenders(webrtc::MediaType media_type);
// Loops through the vector of `streams` and finds added and removed
// StreamParams since last time this method was called.
// For each new or removed StreamParam, OnLocalSenderSeen or
// OnLocalSenderRemoved is invoked.
void UpdateLocalSenders(const std::vector<cricket::StreamParams>& streams,
- cricket::MediaType media_type);
+ webrtc::MediaType media_type);
// Makes sure a MediaStreamTrack is created for each StreamParam in `streams`,
// and existing MediaStreamTracks are removed if there is no corresponding
@@ -476,7 +476,7 @@
void UpdateRemoteSendersList(
const std::vector<cricket::StreamParams>& streams,
bool default_track_needed,
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
StreamCollection* new_streams);
// Enables media channels to allow sending of media.
diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc
index efbda16..36d545c 100644
--- a/pc/sdp_offer_answer_unittest.cc
+++ b/pc/sdp_offer_answer_unittest.cc
@@ -139,7 +139,7 @@
}
std::optional<RtpCodecCapability> FindFirstSendCodecWithName(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::string& name) const {
std::vector<RtpCodecCapability> codecs =
pc_factory_->GetRtpSenderCapabilities(media_type).codecs;
@@ -163,7 +163,7 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
+ auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
// Verify that caller->observer->OnTrack() has been called with a
@@ -607,7 +607,7 @@
rid2.rid = "2";
init.send_encodings.push_back(rid2);
- auto transceiver = pc->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ auto transceiver = pc->AddTransceiver(webrtc::MediaType::VIDEO, init);
EXPECT_TRUE(pc->CreateOfferAndSetAsLocal());
auto mid = pc->pc()->local_description()->description()->contents()[0].mid();
@@ -659,11 +659,11 @@
FieldTrials::CreateNoGlobal("WebRTC-MixedCodecSimulcast/Enabled/"));
std::optional<RtpCodecCapability> vp8_codec_capability =
- FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
cricket::kVp8CodecName);
ASSERT_TRUE(vp8_codec_capability);
std::optional<RtpCodecCapability> vp9_codec_capability =
- FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
+ FindFirstSendCodecWithName(webrtc::MediaType::VIDEO,
cricket::kVp9CodecName);
ASSERT_TRUE(vp9_codec_capability);
@@ -677,7 +677,7 @@
rid2.codec = *vp9_codec_capability;
init.send_encodings.push_back(rid2);
- auto transceiver = pc->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+ auto transceiver = pc->AddTransceiver(webrtc::MediaType::VIDEO, init);
auto offer = pc->CreateOffer();
auto& offer_contents = offer->description()->contents();
auto send_codecs = offer_contents[0].media_description()->codecs();
@@ -1620,8 +1620,8 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
auto receivers = callee->pc()->GetReceivers();
@@ -1643,8 +1643,8 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
auto offer = caller->CreateOfferAndSetAsLocal();
ASSERT_NE(offer, nullptr);
@@ -1678,9 +1678,9 @@
// 1. Restrict codecs and set a local description and remote description.
// with a different payload type.
- auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO);
std::vector<RtpCodecCapability> codec_caps =
- pc_factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO).codecs;
+ pc_factory_->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO).codecs;
codec_caps.erase(std::remove_if(codec_caps.begin(), codec_caps.end(),
[](const RtpCodecCapability& codec) {
return !absl::EqualsIgnoreCase(codec.name,
@@ -1710,7 +1710,7 @@
// 3. sCP to reenable that codec. Payload type is not matched at this point.
codec_caps =
- pc_factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO).codecs;
+ pc_factory_->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO).codecs;
codec_caps.erase(
std::remove_if(codec_caps.begin(), codec_caps.end(),
[](const RtpCodecCapability& codec) {
@@ -1754,8 +1754,8 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
- caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ caller->AddTransceiver(webrtc::MediaType::AUDIO);
+ caller->AddTransceiver(webrtc::MediaType::VIDEO);
// Negotiate, gather candidates, then exchange ICE candidates.
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
diff --git a/pc/session_description.h b/pc/session_description.h
index 56a58a9..e0faf9e 100644
--- a/pc/session_description.h
+++ b/pc/session_description.h
@@ -59,7 +59,7 @@
MediaContentDescription() = default;
virtual ~MediaContentDescription() = default;
- virtual cricket::MediaType type() const = 0;
+ virtual webrtc::MediaType type() const = 0;
// Try to cast this media description to an AudioContentDescription. Returns
// nullptr if the cast fails.
@@ -298,7 +298,7 @@
RTC_DCHECK(cricket::IsRtpProtocol(protocol));
protocol_ = std::string(protocol);
}
- cricket::MediaType type() const override { return cricket::MEDIA_TYPE_AUDIO; }
+ webrtc::MediaType type() const override { return webrtc::MediaType::AUDIO; }
AudioContentDescription* as_audio() override { return this; }
const AudioContentDescription* as_audio() const override { return this; }
@@ -314,7 +314,7 @@
RTC_DCHECK(cricket::IsRtpProtocol(protocol));
protocol_ = std::string(protocol);
}
- cricket::MediaType type() const override { return cricket::MEDIA_TYPE_VIDEO; }
+ webrtc::MediaType type() const override { return webrtc::MediaType::VIDEO; }
VideoContentDescription* as_video() override { return this; }
const VideoContentDescription* as_video() const override { return this; }
@@ -332,7 +332,7 @@
use_sctpmap_(o.use_sctpmap_),
port_(o.port_),
max_message_size_(o.max_message_size_) {}
- cricket::MediaType type() const override { return cricket::MEDIA_TYPE_DATA; }
+ webrtc::MediaType type() const override { return webrtc::MediaType::DATA; }
SctpDataContentDescription* as_sctp() override { return this; }
const SctpDataContentDescription* as_sctp() const override { return this; }
@@ -366,8 +366,8 @@
public:
explicit UnsupportedContentDescription(absl::string_view media_type)
: media_type_(media_type) {}
- cricket::MediaType type() const override {
- return cricket::MEDIA_TYPE_UNSUPPORTED;
+ webrtc::MediaType type() const override {
+ return webrtc::MediaType::UNSUPPORTED;
}
UnsupportedContentDescription* as_unsupported() override { return this; }
diff --git a/pc/test/fake_peer_connection_base.h b/pc/test/fake_peer_connection_base.h
index aad1310..fbefff0 100644
--- a/pc/test/fake_peer_connection_base.h
+++ b/pc/test/fake_peer_connection_base.h
@@ -116,12 +116,12 @@
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
- cricket::MediaType media_type) override {
+ webrtc::MediaType media_type) override {
return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const RtpTransceiverInit& init) override {
return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
}
@@ -382,7 +382,7 @@
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool fire_callback = true) override {
diff --git a/pc/test/fake_peer_connection_for_stats.h b/pc/test/fake_peer_connection_for_stats.h
index 1a07bcb..402bb05 100644
--- a/pc/test/fake_peer_connection_for_stats.h
+++ b/pc/test/fake_peer_connection_for_stats.h
@@ -336,12 +336,11 @@
std::move(voice_media_receive_channel), mid, kDefaultSrtpRequired,
CryptoOptions(), context_->ssrc_generator(), transport_name);
auto transceiver =
- GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
- ->internal();
+ GetOrCreateFirstTransceiverOfType(webrtc::MediaType::AUDIO)->internal();
if (transceiver->channel()) {
// This transceiver already has a channel, create a new one.
transceiver =
- CreateTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)->internal();
+ CreateTransceiverOfType(webrtc::MediaType::AUDIO)->internal();
}
RTC_DCHECK(!transceiver->channel());
transceiver->SetChannel(std::move(voice_channel),
@@ -370,12 +369,11 @@
std::move(video_media_receive_channel), mid, kDefaultSrtpRequired,
CryptoOptions(), context_->ssrc_generator(), transport_name);
auto transceiver =
- GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
- ->internal();
+ GetOrCreateFirstTransceiverOfType(webrtc::MediaType::VIDEO)->internal();
if (transceiver->channel()) {
// This transceiver already has a channel, create a new one.
transceiver =
- CreateTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->internal();
+ CreateTransceiverOfType(webrtc::MediaType::VIDEO)->internal();
}
RTC_DCHECK(!transceiver->channel());
transceiver->SetChannel(std::move(video_channel),
@@ -551,7 +549,7 @@
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- GetOrCreateFirstTransceiverOfType(cricket::MediaType media_type) {
+ GetOrCreateFirstTransceiverOfType(webrtc::MediaType media_type) {
for (auto transceiver : transceivers_) {
if (transceiver->internal()->media_type() == media_type) {
return transceiver;
@@ -561,7 +559,7 @@
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- CreateTransceiverOfType(cricket::MediaType media_type) {
+ CreateTransceiverOfType(webrtc::MediaType media_type) {
auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread_,
rtc::make_ref_counted<RtpTransceiver>(media_type, context_.get(),
diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index 619b54d..ad0341a 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -188,10 +188,10 @@
class MockRtpReceiverObserver : public RtpReceiverObserverInterface {
public:
- explicit MockRtpReceiverObserver(cricket::MediaType media_type)
+ explicit MockRtpReceiverObserver(webrtc::MediaType media_type)
: expected_media_type_(media_type) {}
- void OnFirstPacketReceived(cricket::MediaType media_type) override {
+ void OnFirstPacketReceived(webrtc::MediaType media_type) override {
ASSERT_EQ(expected_media_type_, media_type);
first_packet_received_ = true;
}
@@ -202,15 +202,15 @@
private:
bool first_packet_received_ = false;
- cricket::MediaType expected_media_type_;
+ webrtc::MediaType expected_media_type_;
};
class MockRtpSenderObserver : public RtpSenderObserverInterface {
public:
- explicit MockRtpSenderObserver(cricket::MediaType media_type)
+ explicit MockRtpSenderObserver(webrtc::MediaType media_type)
: expected_media_type_(media_type) {}
- void OnFirstPacketSent(cricket::MediaType media_type) override {
+ void OnFirstPacketSent(webrtc::MediaType media_type) override {
ASSERT_EQ(expected_media_type_, media_type);
first_packet_sent_ = true;
}
@@ -221,7 +221,7 @@
private:
bool first_packet_sent_ = false;
- cricket::MediaType expected_media_type_;
+ webrtc::MediaType expected_media_type_;
};
// Helper class that wraps a peer connection, observes it, and can accept
@@ -397,7 +397,7 @@
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
for (const auto& receiver : pc()->GetReceivers()) {
if (receiver->media_type() == media_type) {
@@ -408,7 +408,7 @@
}
rtc::scoped_refptr<RtpTransceiverInterface> GetFirstTransceiverOfType(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
for (auto transceiver : pc()->GetTransceivers()) {
if (transceiver->receiver()->media_type() == media_type) {
return transceiver;
@@ -714,7 +714,7 @@
void NegotiateCorruptionDetectionHeader() {
for (const auto& transceiver : pc()->GetTransceivers()) {
- if (transceiver->media_type() != cricket::MEDIA_TYPE_VIDEO) {
+ if (transceiver->media_type() != webrtc::MediaType::VIDEO) {
continue;
}
auto extensions = transceiver->GetHeaderExtensionsToNegotiate();
@@ -925,7 +925,7 @@
// Note - we don't check for direction here. This function is called
// before direction is set, and in that case, we should not remove
// the renderer.
- if (transceiver->receiver()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
+ if (transceiver->receiver()->media_type() == webrtc::MediaType::VIDEO) {
active_renderers.insert(transceiver->receiver()->track()->id());
}
}
@@ -1027,7 +1027,7 @@
void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override {
- if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
+ if (receiver->media_type() == webrtc::MediaType::VIDEO) {
rtc::scoped_refptr<VideoTrackInterface> video_track(
static_cast<VideoTrackInterface*>(receiver->track().get()));
ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
@@ -1038,7 +1038,7 @@
}
void OnRemoveTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
- if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
+ if (receiver->media_type() == webrtc::MediaType::VIDEO) {
auto it = fake_video_renderers_.find(receiver->track()->id());
if (it != fake_video_renderers_.end()) {
fake_video_renderers_.erase(it);
diff --git a/pc/test/mock_channel_interface.h b/pc/test/mock_channel_interface.h
index 30bd909..f1f176d 100644
--- a/pc/test/mock_channel_interface.h
+++ b/pc/test/mock_channel_interface.h
@@ -31,7 +31,7 @@
// implementation of BaseChannel.
class MockChannelInterface : public cricket::ChannelInterface {
public:
- MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
+ MOCK_METHOD(webrtc::MediaType, media_type, (), (const, override));
MOCK_METHOD(cricket::VideoChannel*, AsVideoChannel, (), (override));
MOCK_METHOD(cricket::VoiceChannel*, AsVoiceChannel, (), (override));
MOCK_METHOD(cricket::MediaSendChannelInterface*,
diff --git a/pc/test/mock_peer_connection_internal.h b/pc/test/mock_peer_connection_internal.h
index b554c54..a9e4f84 100644
--- a/pc/test/mock_peer_connection_internal.h
+++ b/pc/test/mock_peer_connection_internal.h
@@ -106,11 +106,11 @@
(override));
MOCK_METHOD(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
AddTransceiver,
- (cricket::MediaType),
+ (webrtc::MediaType),
(override));
MOCK_METHOD(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
AddTransceiver,
- (cricket::MediaType, const RtpTransceiverInit&),
+ (webrtc::MediaType, const RtpTransceiverInit&),
(override));
MOCK_METHOD(rtc::scoped_refptr<RtpSenderInterface>,
CreateSender,
@@ -304,7 +304,7 @@
(override));
MOCK_METHOD(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
AddTransceiver,
- (cricket::MediaType,
+ (webrtc::MediaType,
rtc::scoped_refptr<MediaStreamTrackInterface>,
const RtpTransceiverInit&,
bool),
diff --git a/pc/test/mock_rtp_receiver_internal.h b/pc/test/mock_rtp_receiver_internal.h
index 90ccd94..00fa3f4 100644
--- a/pc/test/mock_rtp_receiver_internal.h
+++ b/pc/test/mock_rtp_receiver_internal.h
@@ -47,7 +47,7 @@
streams,
(),
(const, override));
- MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
+ MOCK_METHOD(webrtc::MediaType, media_type, (), (const, override));
MOCK_METHOD(std::string, id, (), (const, override));
MOCK_METHOD(RtpParameters, GetParameters, (), (const, override));
MOCK_METHOD(void, SetObserver, (RtpReceiverObserverInterface*), (override));
diff --git a/pc/test/mock_rtp_sender_internal.h b/pc/test/mock_rtp_sender_internal.h
index b272bd7..4722a69 100644
--- a/pc/test/mock_rtp_sender_internal.h
+++ b/pc/test/mock_rtp_sender_internal.h
@@ -48,7 +48,7 @@
dtls_transport,
(),
(const, override));
- MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
+ MOCK_METHOD(webrtc::MediaType, media_type, (), (const, override));
MOCK_METHOD(std::string, id, (), (const, override));
MOCK_METHOD(std::vector<std::string>, stream_ids, (), (const, override));
MOCK_METHOD(std::vector<RtpEncodingParameters>,
diff --git a/pc/test/mock_voice_media_receive_channel_interface.h b/pc/test/mock_voice_media_receive_channel_interface.h
index 76e3b39..31f8dee 100644
--- a/pc/test/mock_voice_media_receive_channel_interface.h
+++ b/pc/test/mock_voice_media_receive_channel_interface.h
@@ -88,7 +88,7 @@
AsVoiceReceiveChannel,
(),
(override));
- MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
+ MOCK_METHOD(webrtc::MediaType, media_type, (), (const, override));
MOCK_METHOD(bool, AddRecvStream, (const StreamParams& sp), (override));
MOCK_METHOD(bool, RemoveRecvStream, (uint32_t ssrc), (override));
MOCK_METHOD(void, ResetUnsignaledRecvStream, (), (override));
diff --git a/pc/test/peer_connection_test_wrapper.cc b/pc/test/peer_connection_test_wrapper.cc
index 8434b2d..03c4ce3 100644
--- a/pc/test/peer_connection_test_wrapper.cc
+++ b/pc/test/peer_connection_test_wrapper.cc
@@ -247,7 +247,7 @@
std::optional<webrtc::RtpCodecCapability>
PeerConnectionTestWrapper::FindFirstSendCodecWithName(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::string& name) const {
std::vector<webrtc::RtpCodecCapability> codecs =
peer_connection_factory_->GetRtpSenderCapabilities(media_type).codecs;
diff --git a/pc/test/peer_connection_test_wrapper.h b/pc/test/peer_connection_test_wrapper.h
index ecd6bdb..43b1c0d 100644
--- a/pc/test/peer_connection_test_wrapper.h
+++ b/pc/test/peer_connection_test_wrapper.h
@@ -79,7 +79,7 @@
const webrtc::DataChannelInit& init);
std::optional<webrtc::RtpCodecCapability> FindFirstSendCodecWithName(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const std::string& name) const;
void WaitForNegotiation();
diff --git a/pc/track_media_info_map.cc b/pc/track_media_info_map.cc
index 3ad4bbc..5508939 100644
--- a/pc/track_media_info_map.cc
+++ b/pc/track_media_info_map.cc
@@ -11,13 +11,20 @@
#include "pc/track_media_info_map.h"
#include <cstdint>
+#include <map>
+#include <optional>
#include <set>
-#include <type_traits>
#include <utility>
+#include "api/array_view.h"
+#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
+#include "api/scoped_refptr.h"
+#include "media/base/media_channel.h"
#include "media/base/stream_params.h"
+#include "pc/rtp_receiver.h"
+#include "pc/rtp_sender.h"
#include "rtc_base/checks.h"
#include "rtc_base/thread.h"
@@ -51,7 +58,7 @@
RTC_DCHECK(remote_audio_track_by_ssrc->empty());
RTC_DCHECK(remote_video_track_by_ssrc->empty());
for (const auto& rtp_sender : rtp_senders) {
- cricket::MediaType media_type = rtp_sender->media_type();
+ webrtc::MediaType media_type = rtp_sender->media_type();
MediaStreamTrackInterface* track = rtp_sender->track().get();
if (!track) {
continue;
@@ -59,7 +66,7 @@
// TODO(deadbeef): `ssrc` should be removed in favor of `GetParameters`.
uint32_t ssrc = rtp_sender->ssrc();
if (ssrc != 0) {
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
RTC_DCHECK(local_audio_track_by_ssrc->find(ssrc) ==
local_audio_track_by_ssrc->end());
(*local_audio_track_by_ssrc)[ssrc] =
@@ -73,21 +80,21 @@
}
}
for (const auto& rtp_receiver : rtp_receivers) {
- cricket::MediaType media_type = rtp_receiver->media_type();
+ webrtc::MediaType media_type = rtp_receiver->media_type();
MediaStreamTrackInterface* track = rtp_receiver->track().get();
RTC_DCHECK(track);
RtpParameters params = rtp_receiver->GetParameters();
for (const RtpEncodingParameters& encoding : params.encodings) {
if (!encoding.ssrc) {
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
*unsignaled_audio_track = static_cast<AudioTrackInterface*>(track);
} else {
- RTC_DCHECK(media_type == cricket::MEDIA_TYPE_VIDEO);
+ RTC_DCHECK(media_type == webrtc::MediaType::VIDEO);
*unsignaled_video_track = static_cast<VideoTrackInterface*>(track);
}
continue;
}
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
RTC_DCHECK(remote_audio_track_by_ssrc->find(*encoding.ssrc) ==
remote_audio_track_by_ssrc->end());
(*remote_audio_track_by_ssrc)[*encoding.ssrc] =
diff --git a/pc/track_media_info_map_unittest.cc b/pc/track_media_info_map_unittest.cc
index f62043a..8c4f10a 100644
--- a/pc/track_media_info_map_unittest.cc
+++ b/pc/track_media_info_map_unittest.cc
@@ -14,21 +14,25 @@
#include <cstdint>
#include <initializer_list>
+#include <optional>
#include <string>
-#include <type_traits>
#include <utility>
#include <vector>
+#include "api/make_ref_counted.h"
+#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
+#include "api/scoped_refptr.h"
#include "api/test/mock_video_track.h"
#include "media/base/media_channel.h"
#include "pc/audio_track.h"
+#include "pc/rtp_receiver.h"
+#include "pc/rtp_sender.h"
#include "pc/test/fake_video_track_source.h"
#include "pc/test/mock_rtp_receiver_internal.h"
#include "pc/test/mock_rtp_sender_internal.h"
#include "pc/video_track.h"
-#include "rtc_base/checks.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -51,7 +55,7 @@
}
rtc::scoped_refptr<MockRtpSenderInternal> CreateMockRtpSender(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
std::initializer_list<uint32_t> ssrcs,
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
uint32_t first_ssrc;
@@ -73,7 +77,7 @@
}
rtc::scoped_refptr<MockRtpReceiverInternal> CreateMockRtpReceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
std::initializer_list<uint32_t> ssrcs,
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
auto receiver = rtc::make_ref_counted<MockRtpReceiverInternal>();
@@ -119,8 +123,8 @@
MediaStreamTrackInterface* local_track) {
rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockRtpSender(
local_track->kind() == MediaStreamTrackInterface::kAudioKind
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO,
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO,
ssrcs, rtc::scoped_refptr<MediaStreamTrackInterface>(local_track));
rtp_senders_.push_back(rtp_sender);
@@ -148,8 +152,8 @@
MediaStreamTrackInterface* remote_track) {
auto rtp_receiver = CreateMockRtpReceiver(
remote_track->kind() == MediaStreamTrackInterface::kAudioKind
- ? cricket::MEDIA_TYPE_AUDIO
- : cricket::MEDIA_TYPE_VIDEO,
+ ? webrtc::MediaType::AUDIO
+ : webrtc::MediaType::VIDEO,
ssrcs, rtc::scoped_refptr<MediaStreamTrackInterface>(remote_track));
rtp_receivers_.push_back(rtp_receiver);
diff --git a/pc/typed_codec_vendor.cc b/pc/typed_codec_vendor.cc
index 74741ae..dee14fc 100644
--- a/pc/typed_codec_vendor.cc
+++ b/pc/typed_codec_vendor.cc
@@ -24,6 +24,7 @@
#include "media/base/codec_list.h"
#include "media/base/media_constants.h"
#include "media/base/media_engine.h"
+#include "rtc_base/logging.h"
namespace cricket {
@@ -102,7 +103,7 @@
const webrtc::FieldTrialsView& trials) {
if (trials.IsEnabled("WebRTC-PayloadTypesInTransport")) {
// Get the capabilities from the factory and compute the codecs.
- if (type == MEDIA_TYPE_AUDIO) {
+ if (type == webrtc::MediaType::AUDIO) {
if (is_sender) {
if (media_engine->voice().encoder_factory()) {
codecs_ = CodecList::CreateFromTrustedData(CollectAudioCodecs(
@@ -133,7 +134,7 @@
}
} else {
// Use current mechanisms for getting codecs from media engine.
- if (type == MEDIA_TYPE_AUDIO) {
+ if (type == webrtc::MediaType::AUDIO) {
if (is_sender) {
codecs_ = CodecList::CreateFromTrustedData(
media_engine->voice().LegacySendCodecs());
diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h
index caef459..02c78db 100644
--- a/pc/video_rtp_receiver.h
+++ b/pc/video_rtp_receiver.h
@@ -68,8 +68,8 @@
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override;
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_VIDEO;
+ webrtc::MediaType media_type() const override {
+ return webrtc::MediaType::VIDEO;
}
std::string id() const override { return id_; }
diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc
index da92a13..14511ae 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -266,22 +266,22 @@
static void BuildMediaDescription(const ContentInfo* content_info,
const TransportInfo* transport_info,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
const std::vector<Candidate>& candidates,
int msid_signaling,
std::string* message);
-static void BuildMediaLine(const cricket::MediaType media_type,
+static void BuildMediaLine(const webrtc::MediaType media_type,
const ContentInfo* content_info,
const MediaContentDescription* media_desc,
std::string* message);
static void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
int msid_signaling,
std::string* message);
static void BuildRtpHeaderExtensions(const RtpHeaderExtensions& extensions,
std::string* message);
static void BuildRtpmap(const MediaContentDescription* media_desc,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
std::string* message);
static void BuildCandidate(const std::vector<Candidate>& candidates,
bool include_ufrag,
@@ -312,7 +312,7 @@
SdpParseError* error);
static bool ParseContent(
absl::string_view message,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
int mline_index,
absl::string_view protocol,
const std::vector<int>& payload_types,
@@ -335,12 +335,12 @@
SsrcGroupVec* ssrc_groups,
SdpParseError* error);
static bool ParseRtpmapAttribute(absl::string_view line,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
const std::vector<int>& payload_types,
MediaContentDescription* media_desc,
SdpParseError* error);
static bool ParseFmtpAttributes(absl::string_view line,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
MediaContentDescription* media_desc,
SdpParseError* error);
static bool ParseFmtpParam(absl::string_view line,
@@ -348,11 +348,11 @@
std::string* value,
SdpParseError* error);
static bool ParsePacketizationAttribute(absl::string_view line,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
MediaContentDescription* media_desc,
SdpParseError* error);
static bool ParseRtcpFbAttribute(absl::string_view line,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
MediaContentDescription* media_desc,
SdpParseError* error);
static bool ParseIceOptions(absl::string_view line,
@@ -1399,7 +1399,7 @@
}
}
-void BuildMediaLine(const cricket::MediaType media_type,
+void BuildMediaLine(const webrtc::MediaType media_type,
const ContentInfo* content_info,
const MediaContentDescription* media_desc,
std::string* message) {
@@ -1410,15 +1410,15 @@
// fmt is a list of payload type numbers that MAY be used in the session.
std::string type;
std::string fmt;
- if (media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO) {
- type = media_type == cricket::MEDIA_TYPE_AUDIO ? kSdpMediaTypeAudio
- : kSdpMediaTypeVideo;
+ if (media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO) {
+ type = media_type == webrtc::MediaType::AUDIO ? kSdpMediaTypeAudio
+ : kSdpMediaTypeVideo;
for (const cricket::Codec& codec : media_desc->codecs()) {
fmt.append(" ");
fmt.append(rtc::ToString(codec.id));
}
- } else if (media_type == cricket::MEDIA_TYPE_DATA) {
+ } else if (media_type == webrtc::MediaType::DATA) {
type = kSdpMediaTypeData;
const SctpDataContentDescription* sctp_data_desc = media_desc->as_sctp();
if (sctp_data_desc) {
@@ -1432,7 +1432,7 @@
} else {
RTC_DCHECK_NOTREACHED() << "Data description without SCTP";
}
- } else if (media_type == cricket::MEDIA_TYPE_UNSUPPORTED) {
+ } else if (media_type == webrtc::MediaType::UNSUPPORTED) {
const UnsupportedContentDescription* unsupported_desc =
media_desc->as_unsupported();
type = unsupported_desc->media_type();
@@ -1470,7 +1470,7 @@
void BuildMediaDescription(const ContentInfo* content_info,
const TransportInfo* transport_info,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
const std::vector<Candidate>& candidates,
int msid_signaling,
std::string* message) {
@@ -1562,7 +1562,7 @@
}
void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
int msid_signaling,
std::string* message) {
SimulcastSdpSerializer serializer;
@@ -1875,12 +1875,12 @@
}
void BuildRtpmap(const MediaContentDescription* media_desc,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
std::string* message) {
RTC_DCHECK(message != NULL);
RTC_DCHECK(media_desc != NULL);
StringBuilder os;
- if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type == webrtc::MediaType::VIDEO) {
for (const cricket::Codec& codec : media_desc->codecs()) {
// RFC 4566
// a=rtpmap:<payload type> <encoding name>/<clock rate>
@@ -1895,7 +1895,7 @@
AddRtcpFbLines(codec, message);
AddFmtpLine(codec, message);
}
- } else if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ } else if (media_type == webrtc::MediaType::AUDIO) {
std::vector<int> ptimes;
std::vector<int> maxptimes;
int max_minptime = 0;
@@ -2585,7 +2585,7 @@
static std::unique_ptr<MediaContentDescription> ParseContentDescription(
absl::string_view message,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
int mline_index,
absl::string_view protocol,
const std::vector<int>& payload_types,
@@ -2597,9 +2597,9 @@
std::vector<std::unique_ptr<JsepIceCandidate>>* candidates,
SdpParseError* error) {
std::unique_ptr<MediaContentDescription> media_desc;
- if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
media_desc = std::make_unique<AudioContentDescription>();
- } else if (media_type == cricket::MediaType::MEDIA_TYPE_VIDEO) {
+ } else if (media_type == webrtc::MediaType::VIDEO) {
media_desc = std::make_unique<VideoContentDescription>();
} else {
RTC_DCHECK_NOTREACHED();
@@ -2720,12 +2720,12 @@
}
if (media_type == kSdpMediaTypeVideo) {
content = ParseContentDescription(
- message, cricket::MEDIA_TYPE_VIDEO, mline_index, protocol,
+ message, webrtc::MediaType::VIDEO, mline_index, protocol,
payload_types, pos, &content_name, &bundle_only,
§ion_msid_signaling, &transport, candidates, error);
} else if (media_type == kSdpMediaTypeAudio) {
content = ParseContentDescription(
- message, cricket::MEDIA_TYPE_AUDIO, mline_index, protocol,
+ message, webrtc::MediaType::AUDIO, mline_index, protocol,
payload_types, pos, &content_name, &bundle_only,
§ion_msid_signaling, &transport, candidates, error);
} else if (media_type == kSdpMediaTypeData &&
@@ -2746,10 +2746,10 @@
} else if (fields[3] == kDefaultSctpmapProtocol) {
data_desc->set_use_sctpmap(false);
}
- if (!ParseContent(message, cricket::MEDIA_TYPE_DATA, mline_index,
- protocol, payload_types, pos, &content_name,
- &bundle_only, §ion_msid_signaling, data_desc.get(),
- &transport, candidates, error)) {
+ if (!ParseContent(message, webrtc::MediaType::DATA, mline_index, protocol,
+ payload_types, pos, &content_name, &bundle_only,
+ §ion_msid_signaling, data_desc.get(), &transport,
+ candidates, error)) {
return false;
}
data_desc->set_protocol(protocol);
@@ -2758,7 +2758,7 @@
RTC_LOG(LS_WARNING) << "Unsupported media type: " << *mline;
auto unsupported_desc =
std::make_unique<UnsupportedContentDescription>(media_type);
- if (!ParseContent(message, cricket::MEDIA_TYPE_UNSUPPORTED, mline_index,
+ if (!ParseContent(message, webrtc::MediaType::UNSUPPORTED, mline_index,
protocol, payload_types, pos, &content_name,
&bundle_only, §ion_msid_signaling,
unsupported_desc.get(), &transport, candidates,
@@ -2873,14 +2873,14 @@
// is no Codec associated with that payload type it returns an empty codec
// with that payload type.
cricket::Codec GetCodecWithPayloadType(
- cricket::MediaType type,
+ webrtc::MediaType type,
const std::vector<cricket::Codec>& codecs,
int payload_type) {
const cricket::Codec* codec = FindCodecById(codecs, payload_type);
if (codec)
return *codec;
// Return empty codec with `payload_type`.
- if (type == cricket::MEDIA_TYPE_AUDIO) {
+ if (type == webrtc::MediaType::AUDIO) {
return cricket::CreateAudioCodec(payload_type, "", 0, 0);
} else {
return cricket::CreateVideoCodec(payload_type, "");
@@ -2993,7 +2993,7 @@
}
bool ParseContent(absl::string_view message,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
int mline_index,
absl::string_view protocol,
const std::vector<int>& payload_types,
@@ -3009,7 +3009,7 @@
RTC_DCHECK(content_name != NULL);
RTC_DCHECK(transport != NULL);
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
MaybeCreateStaticPayloadAudioCodecs(payload_types, media_desc);
}
@@ -3154,7 +3154,7 @@
return false;
}
} else if (cricket::IsDtlsSctp(protocol) &&
- media_type == cricket::MEDIA_TYPE_DATA) {
+ media_type == webrtc::MediaType::DATA) {
//
// SCTP specific attributes
//
@@ -3356,7 +3356,7 @@
if (!ssrc_infos.empty()) {
CreateTracksFromSsrcInfos(ssrc_infos, stream_ids, track_id, &tracks,
*msid_signaling);
- } else if (media_type != cricket::MEDIA_TYPE_DATA &&
+ } else if (media_type != webrtc::MediaType::DATA &&
(*msid_signaling & kMsidSignalingMediaSection)) {
// If the stream_ids/track_id was signaled but SSRCs were unsignaled we
// still create a track. This isn't done for data media types because
@@ -3396,7 +3396,7 @@
})) {
return ParseFailed("Failed to parse codecs correctly.", error);
}
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ if (media_type == webrtc::MediaType::AUDIO) {
AddAudioAttribute(kCodecParamMaxPTime, maxptime_as_string, media_desc);
AddAudioAttribute(kCodecParamPTime, ptime_as_string, media_desc);
}
@@ -3556,7 +3556,7 @@
}
bool ParseRtpmapAttribute(absl::string_view line,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
const std::vector<int>& payload_types,
MediaContentDescription* media_desc,
SdpParseError* error) {
@@ -3600,7 +3600,7 @@
return false;
}
- if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type == webrtc::MediaType::VIDEO) {
for (const cricket::Codec& existing_codec : media_desc->codecs()) {
if (!existing_codec.name.empty() && payload_type == existing_codec.id &&
(!absl::EqualsIgnoreCase(encoding_name, existing_codec.name) ||
@@ -3616,7 +3616,7 @@
}
}
UpdateCodec(payload_type, encoding_name, media_desc);
- } else if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ } else if (media_type == webrtc::MediaType::AUDIO) {
// RFC 4566
// For audio streams, <encoding parameters> indicates the number
// of audio channels. This parameter is OPTIONAL and may be
@@ -3689,11 +3689,11 @@
}
bool ParseFmtpAttributes(absl::string_view line,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
MediaContentDescription* media_desc,
SdpParseError* error) {
- if (media_type != cricket::MEDIA_TYPE_AUDIO &&
- media_type != cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type != webrtc::MediaType::AUDIO &&
+ media_type != webrtc::MediaType::VIDEO) {
return true;
}
@@ -3729,18 +3729,18 @@
return false;
}
- if (media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO) {
UpdateCodec(media_desc, payload_type, codec_params);
}
return true;
}
bool ParsePacketizationAttribute(absl::string_view line,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
MediaContentDescription* media_desc,
SdpParseError* error) {
- if (media_type != cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type != webrtc::MediaType::VIDEO) {
return true;
}
std::vector<absl::string_view> packetization_fields =
@@ -3764,11 +3764,11 @@
}
bool ParseRtcpFbAttribute(absl::string_view line,
- const cricket::MediaType media_type,
+ const webrtc::MediaType media_type,
MediaContentDescription* media_desc,
SdpParseError* error) {
- if (media_type != cricket::MEDIA_TYPE_AUDIO &&
- media_type != cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type != webrtc::MediaType::AUDIO &&
+ media_type != webrtc::MediaType::VIDEO) {
return true;
}
std::vector<absl::string_view> rtcp_fb_fields =
@@ -3796,8 +3796,8 @@
}
const cricket::FeedbackParam feedback_param(id, param);
- if (media_type == cricket::MEDIA_TYPE_AUDIO ||
- media_type == cricket::MEDIA_TYPE_VIDEO) {
+ if (media_type == webrtc::MediaType::AUDIO ||
+ media_type == webrtc::MediaType::VIDEO) {
UpdateCodec(media_desc, payload_type, feedback_param);
}
return true;
diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index e022c30..f644292 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -3611,9 +3611,9 @@
JsepSessionDescription jdesc(kDummyType);
const std::string media_content_sdps[3] = {kSdpAudioString, kSdpVideoString,
kSdpSctpDataChannelString};
- const cricket::MediaType media_types[3] = {cricket::MEDIA_TYPE_AUDIO,
- cricket::MEDIA_TYPE_VIDEO,
- cricket::MEDIA_TYPE_DATA};
+ const webrtc::MediaType media_types[3] = {webrtc::MediaType::AUDIO,
+ webrtc::MediaType::VIDEO,
+ webrtc::MediaType::DATA};
// Verifies all 6 permutations.
for (size_t i = 0; i < 6; ++i) {
diff --git a/sdk/android/api/org/webrtc/MediaStreamTrack.java b/sdk/android/api/org/webrtc/MediaStreamTrack.java
index 2e4c3e1..8e39d3c 100644
--- a/sdk/android/api/org/webrtc/MediaStreamTrack.java
+++ b/sdk/android/api/org/webrtc/MediaStreamTrack.java
@@ -28,7 +28,7 @@
}
}
- // Must be kept in sync with cricket::MediaType.
+ // Must be kept in sync with webrtc::MediaType.
public enum MediaType {
MEDIA_TYPE_AUDIO(0),
MEDIA_TYPE_VIDEO(1);
diff --git a/sdk/android/src/jni/pc/media_stream_track.cc b/sdk/android/src/jni/pc/media_stream_track.cc
index 297bbf6..20c0253 100644
--- a/sdk/android/src/jni/pc/media_stream_track.cc
+++ b/sdk/android/src/jni/pc/media_stream_track.cc
@@ -20,13 +20,13 @@
ScopedJavaLocalRef<jobject> NativeToJavaMediaType(
JNIEnv* jni,
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
return Java_MediaType_fromNativeIndex(jni, static_cast<int>(media_type));
}
-cricket::MediaType JavaToNativeMediaType(JNIEnv* jni,
- const JavaRef<jobject>& j_media_type) {
- return static_cast<cricket::MediaType>(
+webrtc::MediaType JavaToNativeMediaType(JNIEnv* jni,
+ const JavaRef<jobject>& j_media_type) {
+ return static_cast<webrtc::MediaType>(
Java_MediaType_getNative(jni, j_media_type));
}
diff --git a/sdk/android/src/jni/pc/media_stream_track.h b/sdk/android/src/jni/pc/media_stream_track.h
index 8bfe302..a005b1a 100644
--- a/sdk/android/src/jni/pc/media_stream_track.h
+++ b/sdk/android/src/jni/pc/media_stream_track.h
@@ -19,11 +19,10 @@
namespace webrtc {
namespace jni {
-ScopedJavaLocalRef<jobject> NativeToJavaMediaType(
- JNIEnv* jni,
- cricket::MediaType media_type);
-cricket::MediaType JavaToNativeMediaType(JNIEnv* jni,
- const JavaRef<jobject>& j_media_type);
+ScopedJavaLocalRef<jobject> NativeToJavaMediaType(JNIEnv* jni,
+ webrtc::MediaType media_type);
+webrtc::MediaType JavaToNativeMediaType(JNIEnv* jni,
+ const JavaRef<jobject>& j_media_type);
} // namespace jni
} // namespace webrtc
diff --git a/sdk/android/src/jni/pc/rtp_receiver.cc b/sdk/android/src/jni/pc/rtp_receiver.cc
index 681d757..998fae7 100644
--- a/sdk/android/src/jni/pc/rtp_receiver.cc
+++ b/sdk/android/src/jni/pc/rtp_receiver.cc
@@ -33,7 +33,7 @@
~RtpReceiverObserverJni() override = default;
- void OnFirstPacketReceived(cricket::MediaType media_type) override {
+ void OnFirstPacketReceived(webrtc::MediaType media_type) override {
JNIEnv* const env = AttachCurrentThreadIfNeeded();
Java_Observer_onFirstPacketReceived(env, j_observer_global_,
NativeToJavaMediaType(env, media_type));
diff --git a/sdk/android/src/jni/pc/rtp_sender.cc b/sdk/android/src/jni/pc/rtp_sender.cc
index 2096f3b..31cec9d 100644
--- a/sdk/android/src/jni/pc/rtp_sender.cc
+++ b/sdk/android/src/jni/pc/rtp_sender.cc
@@ -112,9 +112,9 @@
static jni_zero::ScopedJavaLocalRef<jstring> JNI_RtpSender_GetMediaType(
JNIEnv* jni,
jlong j_rtp_sender_pointer) {
- cricket::MediaType media_type =
+ webrtc::MediaType media_type =
reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)->media_type();
- return media_type == cricket::MEDIA_TYPE_AUDIO
+ return media_type == webrtc::MediaType::AUDIO
? NativeToJavaString(jni, "audio")
: NativeToJavaString(jni, "video");
}
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
index 710ff3b..f543ac3 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
@@ -259,7 +259,7 @@
- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpSenderCapabilitiesForKind:
(NSString *)kind {
- cricket::MediaType mediaType = [[self class] mediaTypeForKind:kind];
+ webrtc::MediaType mediaType = [[self class] mediaTypeForKind:kind];
webrtc::RtpCapabilities rtpCapabilities =
_nativeFactory->GetRtpSenderCapabilities(mediaType);
@@ -269,7 +269,7 @@
- (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpReceiverCapabilitiesForKind:
(NSString *)kind {
- cricket::MediaType mediaType = [[self class] mediaTypeForKind:kind];
+ webrtc::MediaType mediaType = [[self class] mediaTypeForKind:kind];
webrtc::RtpCapabilities rtpCapabilities =
_nativeFactory->GetRtpReceiverCapabilities(mediaType);
@@ -431,14 +431,14 @@
#pragma mark - Private
-+ (cricket::MediaType)mediaTypeForKind:(NSString *)kind {
++ (webrtc::MediaType)mediaTypeForKind:(NSString *)kind {
if (kind == kRTCMediaStreamTrackKindAudio) {
- return cricket::MEDIA_TYPE_AUDIO;
+ return webrtc::MediaType::AUDIO;
} else if (kind == kRTCMediaStreamTrackKindVideo) {
- return cricket::MEDIA_TYPE_VIDEO;
+ return webrtc::MediaType::VIDEO;
} else {
RTC_DCHECK_NOTREACHED();
- return cricket::MEDIA_TYPE_UNSUPPORTED;
+ return webrtc::MediaType::UNSUPPORTED;
}
}
diff --git a/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm b/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm
index e972694..0cfe53a 100644
--- a/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpCodecCapability.mm
@@ -41,10 +41,10 @@
}
_name = [NSString stringForStdString:nativeRtpCodecCapability.name];
switch (nativeRtpCodecCapability.kind) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
_kind = kRTCMediaStreamTrackKindAudio;
break;
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
_kind = kRTCMediaStreamTrackKindVideo;
break;
default:
@@ -96,9 +96,9 @@
// NSString pointer comparison is safe here since "kind" is readonly and only
// populated above.
if (_kind == kRTCMediaStreamTrackKindAudio) {
- rtpCodecCapability.kind = cricket::MEDIA_TYPE_AUDIO;
+ rtpCodecCapability.kind = webrtc::MediaType::AUDIO;
} else if (_kind == kRTCMediaStreamTrackKindVideo) {
- rtpCodecCapability.kind = cricket::MEDIA_TYPE_VIDEO;
+ rtpCodecCapability.kind = webrtc::MediaType::VIDEO;
} else {
RTC_DCHECK_NOTREACHED();
}
diff --git a/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm b/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
index 579607b..904e563 100644
--- a/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
@@ -53,10 +53,10 @@
_payloadType = nativeParameters.payload_type;
_name = [NSString stringForStdString:nativeParameters.name];
switch (nativeParameters.kind) {
- case cricket::MEDIA_TYPE_AUDIO:
+ case webrtc::MediaType::AUDIO:
_kind = kRTCMediaStreamTrackKindAudio;
break;
- case cricket::MEDIA_TYPE_VIDEO:
+ case webrtc::MediaType::VIDEO:
_kind = kRTCMediaStreamTrackKindVideo;
break;
default:
@@ -86,9 +86,9 @@
// NSString pointer comparison is safe here since "kind" is readonly and only
// populated above.
if (_kind == kRTCMediaStreamTrackKindAudio) {
- parameters.kind = cricket::MEDIA_TYPE_AUDIO;
+ parameters.kind = webrtc::MediaType::AUDIO;
} else if (_kind == kRTCMediaStreamTrackKindVideo) {
- parameters.kind = cricket::MEDIA_TYPE_VIDEO;
+ parameters.kind = webrtc::MediaType::VIDEO;
} else {
RTC_DCHECK_NOTREACHED();
}
diff --git a/sdk/objc/api/peerconnection/RTCRtpReceiver+Private.h b/sdk/objc/api/peerconnection/RTCRtpReceiver+Private.h
index 38bdea2..cf7ebc5 100644
--- a/sdk/objc/api/peerconnection/RTCRtpReceiver+Private.h
+++ b/sdk/objc/api/peerconnection/RTCRtpReceiver+Private.h
@@ -22,7 +22,7 @@
public:
RtpReceiverDelegateAdapter(RTC_OBJC_TYPE(RTCRtpReceiver) * receiver);
- void OnFirstPacketReceived(cricket::MediaType media_type) override;
+ void OnFirstPacketReceived(webrtc::MediaType media_type) override;
private:
__weak RTC_OBJC_TYPE(RTCRtpReceiver) * receiver_;
@@ -44,9 +44,9 @@
NS_DESIGNATED_INITIALIZER;
+ (RTCRtpMediaType)mediaTypeForNativeMediaType:
- (cricket::MediaType)nativeMediaType;
+ (webrtc::MediaType)nativeMediaType;
-+ (cricket::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType;
++ (webrtc::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType;
+ (NSString *)stringForMediaType:(RTCRtpMediaType)mediaType;
diff --git a/sdk/objc/api/peerconnection/RTCRtpReceiver.mm b/sdk/objc/api/peerconnection/RTCRtpReceiver.mm
index 3f2dd10..1ceca02 100644
--- a/sdk/objc/api/peerconnection/RTCRtpReceiver.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpReceiver.mm
@@ -28,7 +28,7 @@
}
void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
- cricket::MediaType media_type) {
+ webrtc::MediaType media_type) {
RTCRtpMediaType packet_media_type =
[RTC_OBJC_TYPE(RTCRtpReceiver) mediaTypeForNativeMediaType:media_type];
RTC_OBJC_TYPE(RTCRtpReceiver) *receiver = receiver_;
diff --git a/sdk/objc/api/peerconnection/RTCRtpSender.mm b/sdk/objc/api/peerconnection/RTCRtpSender.mm
index 08ab9ed..fe92abf 100644
--- a/sdk/objc/api/peerconnection/RTCRtpSender.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpSender.mm
@@ -128,7 +128,7 @@
if (self) {
_factory = factory;
_nativeRtpSender = nativeRtpSender;
- if (_nativeRtpSender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+ if (_nativeRtpSender->media_type() == webrtc::MediaType::AUDIO) {
rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender(
_nativeRtpSender->GetDtmfSender());
if (nativeDtmfSender) {
diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index 8e9cf63..20ab330 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -586,14 +586,20 @@
"../../../api:array_view",
"../../../api:libjingle_peerconnection_api",
"../../../api:rtp_parameters",
+ "../../../api:rtp_transceiver_direction",
"../../../api/test/pclf:media_configuration",
"../../../media:media_constants",
"../../../media:rid_description",
+ "../../../media:stream_params",
"../../../p2p:p2p_constants",
+ "../../../p2p:transport_description",
+ "../../../p2p:transport_info",
"../../../pc:sdp_utils",
"../../../pc:session_description",
"../../../pc:simulcast_description",
+ "../../../rtc_base:checks",
"../../../rtc_base:stringutils",
+ "../../../rtc_base:unique_id_generator",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
]
diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc
index 0a58040..606dcbc 100644
--- a/test/pc/e2e/peer_connection_quality_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test.cc
@@ -56,6 +56,7 @@
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread.h"
#include "system_wrappers/include/cpu_info.h"
#include "system_wrappers/include/field_trial.h"
#include "test/field_trial.h"
@@ -521,7 +522,7 @@
// multiple audio streams, then we need transceiver for each Bob's audio
// stream.
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
- alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_AUDIO,
+ alice_->AddTransceiver(webrtc::MediaType::AUDIO,
receive_only_transceiver_init);
RTC_CHECK(result.ok());
alice_transceivers_counter++;
@@ -561,8 +562,7 @@
alice_video_transceivers_non_simulcast_counter++;
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
- alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_VIDEO,
- transceiver_params);
+ alice_->AddTransceiver(webrtc::MediaType::VIDEO, transceiver_params);
RTC_CHECK(result.ok());
alice_transceivers_counter++;
@@ -573,7 +573,7 @@
for (size_t i = alice_video_transceivers_non_simulcast_counter;
i < bob_->configurable_params().video_configs.size(); ++i) {
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
- alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_VIDEO,
+ alice_->AddTransceiver(webrtc::MediaType::VIDEO,
receive_only_transceiver_init);
RTC_CHECK(result.ok());
alice_transceivers_counter++;
@@ -599,19 +599,19 @@
peer->params().video_codecs, true, peer->params().use_ulp_fec,
peer->params().use_flex_fec,
peer->pc_factory()
- ->GetRtpReceiverCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO)
+ ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO)
.codecs);
std::vector<RtpCodecCapability> without_rtx_video_capabilities =
FilterVideoCodecCapabilities(
peer->params().video_codecs, false, peer->params().use_ulp_fec,
peer->params().use_flex_fec,
peer->pc_factory()
- ->GetRtpReceiverCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO)
+ ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO)
.codecs);
// Set codecs for transceivers
for (auto transceiver : peer->pc()->GetTransceivers()) {
- if (transceiver->media_type() == cricket::MediaType::MEDIA_TYPE_VIDEO) {
+ if (transceiver->media_type() == webrtc::MediaType::VIDEO) {
if (transceiver->sender()->init_send_encodings().size() > 1) {
// If transceiver's sender has more then 1 send encodings, it means it
// has multiple simulcast streams, so we need disable RTX on it.
diff --git a/test/pc/e2e/sdp/sdp_changer.cc b/test/pc/e2e/sdp/sdp_changer.cc
index d497cb8..065cc34 100644
--- a/test/pc/e2e/sdp/sdp_changer.cc
+++ b/test/pc/e2e/sdp/sdp_changer.cc
@@ -10,15 +10,34 @@
#include "test/pc/e2e/sdp/sdp_changer.h"
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <map>
+#include <memory>
+#include <string>
#include <utility>
+#include <vector>
-#include "absl/memory/memory.h"
+#include "api/array_view.h"
+#include "api/jsep.h"
#include "api/jsep_session_description.h"
+#include "api/media_types.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_transceiver_direction.h"
#include "api/test/pclf/media_configuration.h"
#include "media/base/media_constants.h"
+#include "media/base/rid_description.h"
+#include "media/base/stream_params.h"
#include "p2p/base/p2p_constants.h"
+#include "p2p/base/transport_description.h"
+#include "p2p/base/transport_info.h"
#include "pc/sdp_utils.h"
+#include "pc/session_description.h"
+#include "pc/simulcast_description.h"
+#include "rtc_base/checks.h"
#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/unique_id_generator.h"
namespace webrtc {
namespace webrtc_pc_e2e {
@@ -114,7 +133,7 @@
SessionDescriptionInterface* offer) {
for (auto& content : offer->description()->contents()) {
MediaContentDescription* media_desc = content.media_description();
- if (media_desc->type() != cricket::MediaType::MEDIA_TYPE_VIDEO) {
+ if (media_desc->type() != webrtc::MediaType::VIDEO) {
continue;
}
if (media_desc->HasSimulcast()) {
@@ -170,7 +189,7 @@
for (auto& content : offer->description()->contents()) {
context_.mids_order.push_back(content.mid());
MediaContentDescription* media_desc = content.media_description();
- if (media_desc->type() != cricket::MediaType::MEDIA_TYPE_VIDEO) {
+ if (media_desc->type() != webrtc::MediaType::VIDEO) {
continue;
}
if (content.media_description()->streams().empty()) {
@@ -315,8 +334,7 @@
}
for (auto& content : offer->description()->contents()) {
- if (content.media_description()->type() !=
- cricket::MediaType::MEDIA_TYPE_VIDEO) {
+ if (content.media_description()->type() != webrtc::MediaType::VIDEO) {
// We are interested in only video tracks
continue;
}
@@ -365,7 +383,7 @@
const VideoCodecConfig& first_codec) {
for (auto& content : answer->description()->contents()) {
MediaContentDescription* media_desc = content.media_description();
- if (media_desc->type() != cricket::MediaType::MEDIA_TYPE_VIDEO) {
+ if (media_desc->type() != webrtc::MediaType::VIDEO) {
continue;
}
if (content.media_description()->direction() !=
diff --git a/test/pc/e2e/test_peer.h b/test/pc/e2e/test_peer.h
index 64b2947..bb6474e 100644
--- a/test/pc/e2e/test_peer.h
+++ b/test/pc/e2e/test_peer.h
@@ -106,7 +106,7 @@
std::string* error_out = nullptr);
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
- cricket::MediaType media_type,
+ webrtc::MediaType media_type,
const RtpTransceiverInit& init) {
RTC_CHECK(wrapper_) << "TestPeer is already closed";
return wrapper_->AddTransceiver(media_type, init);
diff --git a/test/peer_scenario/tests/BUILD.gn b/test/peer_scenario/tests/BUILD.gn
index 1ad0e00..2ad8c85 100644
--- a/test/peer_scenario/tests/BUILD.gn
+++ b/test/peer_scenario/tests/BUILD.gn
@@ -23,7 +23,14 @@
"../../:create_frame_generator_capturer",
"../../:field_trial",
"../../:test_support",
+ "../../../api:libjingle_peerconnection_api",
+ "../../../api:make_ref_counted",
"../../../api:rtc_stats_api",
+ "../../../api:rtp_parameters",
+ "../../../api:rtp_sender_interface",
+ "../../../api:rtp_transceiver_direction",
+ "../../../api:scoped_refptr",
+ "../../../api/test/network_emulation",
"../../../api/transport:ecn_marking",
"../../../api/units:data_rate",
"../../../api/units:time_delta",
@@ -32,6 +39,7 @@
"../../../pc:media_session",
"../../../pc:pc_test_utils",
"../../../pc:session_description",
+ "../../../rtc_base:checks",
"../../../rtc_base:logging",
]
if (rtc_enable_protobuf) {
diff --git a/test/peer_scenario/tests/bwe_ramp_up_test.cc b/test/peer_scenario/tests/bwe_ramp_up_test.cc
index 3c09523..184dd76 100644
--- a/test/peer_scenario/tests/bwe_ramp_up_test.cc
+++ b/test/peer_scenario/tests/bwe_ramp_up_test.cc
@@ -9,9 +9,18 @@
*/
#include <atomic>
+#include <string>
#include <utility>
+#include "api/jsep.h"
+#include "api/make_ref_counted.h"
+#include "api/media_types.h"
+#include "api/rtp_sender_interface.h"
+#include "api/rtp_transceiver_direction.h"
+#include "api/scoped_refptr.h"
+#include "api/stats/rtc_stats_report.h"
#include "api/stats/rtcstats_objects.h"
+#include "api/test/network_emulation/network_emulation_interfaces.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
@@ -20,6 +29,7 @@
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "pc/media_session.h"
#include "pc/test/mock_peer_connection_observers.h"
+#include "rtc_base/checks.h"
#include "test/create_frame_generator_capturer.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -209,7 +219,7 @@
class MockRtpSenderObserver : public RtpSenderObserverInterface {
public:
- MOCK_METHOD(void, OnFirstPacketSent, (cricket::MediaType));
+ MOCK_METHOD(void, OnFirstPacketSent, (webrtc::MediaType));
};
// Test that caller and callee BWE rampup even if no media packets are sent.
@@ -222,7 +232,7 @@
PeerScenarioClient* caller = s.CreateClient({});
PeerScenarioClient* callee = s.CreateClient({});
- auto transceiver = caller->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+ auto transceiver = caller->pc()->AddTransceiver(webrtc::MediaType::VIDEO);
ASSERT_TRUE(transceiver.error().ok());
MockRtpSenderObserver observer;
diff --git a/video/buffered_frame_decryptor.cc b/video/buffered_frame_decryptor.cc
index 61e8812..a23b064 100644
--- a/video/buffered_frame_decryptor.cc
+++ b/video/buffered_frame_decryptor.cc
@@ -10,13 +10,21 @@
#include "video/buffered_frame_decryptor.h"
+#include <cstddef>
+#include <cstdint>
+#include <memory>
#include <utility>
#include <vector>
+#include "api/array_view.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/field_trials_view.h"
+#include "api/media_types.h"
+#include "api/scoped_refptr.h"
#include "modules/rtp_rtcp/source/frame_object.h"
#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
+#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
-#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -65,7 +73,7 @@
}
// Retrieve the maximum possible size of the decrypted payload.
const size_t max_plaintext_byte_size =
- frame_decryptor_->GetMaxPlaintextByteSize(cricket::MEDIA_TYPE_VIDEO,
+ frame_decryptor_->GetMaxPlaintextByteSize(webrtc::MediaType::VIDEO,
frame->size());
RTC_CHECK_LE(max_plaintext_byte_size, frame->size());
// Place the decrypted frame inline into the existing frame.
@@ -80,7 +88,7 @@
// Attempt to decrypt the video frame.
const FrameDecryptorInterface::Result decrypt_result =
- frame_decryptor_->Decrypt(cricket::MEDIA_TYPE_VIDEO, /*csrcs=*/{},
+ frame_decryptor_->Decrypt(webrtc::MediaType::VIDEO, /*csrcs=*/{},
additional_data, *frame,
inline_decrypted_bitstream);
// Optionally call the callback if there was a change in status
