Remove most of api/ortc/.
It's not currently used or maintained, so it shouldn't be a part of out API.
Bug: webrtc:9824
Change-Id: Ic44c5ea3a9eab8fb75e87a5005cbf6cdd4b1d4ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107645
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25593}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 0d16698..9bee2cd 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -209,34 +209,16 @@
rtc_source_set("ortc_api") {
visibility = [ "*" ]
sources = [
- "ortc/mediadescription.cc",
- "ortc/mediadescription.h",
- "ortc/ortcfactoryinterface.h",
- "ortc/ortcrtpreceiverinterface.h",
- "ortc/ortcrtpsenderinterface.h",
"ortc/packettransportinterface.h",
- "ortc/rtptransportcontrollerinterface.h",
"ortc/rtptransportinterface.h",
- "ortc/sessiondescription.cc",
- "ortc/sessiondescription.h",
"ortc/srtptransportinterface.h",
- "ortc/udptransportinterface.h",
]
- # For mediastreaminterface.h, etc.
- # TODO(deadbeef): Create a separate target for the common things ORTC and
- # PeerConnection code shares, so that ortc_api can depend on that instead of
- # libjingle_peerconnection_api.
deps = [
":libjingle_peerconnection_api",
"..:webrtc_common",
- "../rtc_base:rtc_base",
"//third_party/abseil-cpp/absl/types:optional",
]
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
}
rtc_source_set("rtc_stats_api") {
@@ -633,8 +615,6 @@
sources = [
"array_view_unittest.cc",
- "ortc/mediadescription_unittest.cc",
- "ortc/sessiondescription_unittest.cc",
"rtcerror_unittest.cc",
"rtpparameters_unittest.cc",
"test/loopback_media_transport_unittest.cc",
@@ -649,7 +629,6 @@
":array_view",
":libjingle_peerconnection_api",
":loopback_media_transport",
- ":ortc_api",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
diff --git a/api/ortc/mediadescription.cc b/api/ortc/mediadescription.cc
deleted file mode 100644
index d5155f2..0000000
--- a/api/ortc/mediadescription.cc
+++ /dev/null
@@ -1,13 +0,0 @@
-/*
- * Copyright 2017 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/ortc/mediadescription.h"
-
-namespace webrtc {}
diff --git a/api/ortc/mediadescription.h b/api/ortc/mediadescription.h
deleted file mode 100644
index 5cf1d1a..0000000
--- a/api/ortc/mediadescription.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#ifndef API_ORTC_MEDIADESCRIPTION_H_
-#define API_ORTC_MEDIADESCRIPTION_H_
-
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "absl/types/optional.h"
-#include "api/cryptoparams.h"
-
-namespace webrtc {
-
-// A structured representation of a media description within an SDP session
-// description.
-class MediaDescription {
- public:
- explicit MediaDescription(std::string mid) : mid_(std::move(mid)) {}
-
- ~MediaDescription() {}
-
- // The mid(media stream identification) is used for identifying media streams
- // within a session description.
- // https://tools.ietf.org/html/rfc5888#section-6
- absl::optional<std::string> mid() const { return mid_; }
- void set_mid(std::string mid) { mid_.emplace(std::move(mid)); }
-
- // Security keys and parameters for this media stream. Can be used to
- // negotiate parameters for SRTP.
- // https://tools.ietf.org/html/rfc4568#page-5
- std::vector<cricket::CryptoParams>& sdes_params() { return sdes_params_; }
- const std::vector<cricket::CryptoParams>& sdes_params() const {
- return sdes_params_;
- }
-
- private:
- absl::optional<std::string> mid_;
-
- std::vector<cricket::CryptoParams> sdes_params_;
-};
-
-} // namespace webrtc
-
-#endif // API_ORTC_MEDIADESCRIPTION_H_
diff --git a/api/ortc/mediadescription_unittest.cc b/api/ortc/mediadescription_unittest.cc
deleted file mode 100644
index 9ff943a..0000000
--- a/api/ortc/mediadescription_unittest.cc
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * Copyright 2017 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/ortc/mediadescription.h"
-#include "test/gtest.h"
-
-namespace webrtc {
-
-class MediaDescriptionTest : public testing::Test {};
-
-TEST_F(MediaDescriptionTest, CreateMediaDescription) {
- MediaDescription m("a");
- EXPECT_EQ("a", m.mid());
-}
-
-TEST_F(MediaDescriptionTest, AddSdesParam) {
- MediaDescription m("a");
- m.sdes_params().push_back(cricket::CryptoParams());
- const std::vector<cricket::CryptoParams>& params = m.sdes_params();
- EXPECT_EQ(1u, params.size());
-}
-
-} // namespace webrtc
diff --git a/api/ortc/ortcfactoryinterface.h b/api/ortc/ortcfactoryinterface.h
deleted file mode 100644
index 9937352..0000000
--- a/api/ortc/ortcfactoryinterface.h
+++ /dev/null
@@ -1,232 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#ifndef API_ORTC_ORTCFACTORYINTERFACE_H_
-#define API_ORTC_ORTCFACTORYINTERFACE_H_
-
-#include <memory>
-#include <string>
-#include <utility> // For std::move.
-
-#include "api/mediastreaminterface.h"
-#include "api/mediatypes.h"
-#include "api/ortc/ortcrtpreceiverinterface.h"
-#include "api/ortc/ortcrtpsenderinterface.h"
-#include "api/ortc/packettransportinterface.h"
-#include "api/ortc/rtptransportcontrollerinterface.h"
-#include "api/ortc/rtptransportinterface.h"
-#include "api/ortc/srtptransportinterface.h"
-#include "api/ortc/udptransportinterface.h"
-#include "api/peerconnectioninterface.h"
-#include "api/rtcerror.h"
-#include "api/rtpparameters.h"
-#include "rtc_base/network.h"
-#include "rtc_base/scoped_ref_ptr.h"
-#include "rtc_base/thread.h"
-
-namespace webrtc {
-
-// TODO(deadbeef): This should be part of /api/, but currently it's not and
-// including its header violates checkdeps rules.
-class AudioDeviceModule;
-
-// WARNING: This is experimental/under development, so use at your own risk; no
-// guarantee about API stability is guaranteed here yet.
-//
-// This class is the ORTC analog of PeerConnectionFactory. It acts as a factory
-// for ORTC objects that can be connected to each other.
-//
-// Some of these objects may not be represented by the ORTC specification, but
-// follow the same general principles.
-//
-// If one of the factory methods takes another object as an argument, it MUST
-// have been created by the same OrtcFactory.
-//
-// On object lifetimes: objects should be destroyed in this order:
-// 1. Objects created by the factory.
-// 2. The factory itself.
-// 3. Objects passed into OrtcFactoryInterface::Create.
-class OrtcFactoryInterface {
- public:
- // |network_thread| is the thread on which packets are sent and received.
- // If null, a new rtc::Thread with a default socket server is created.
- //
- // |signaling_thread| is used for callbacks to the consumer of the API. If
- // null, the current thread will be used, which assumes that the API consumer
- // is running a message loop on this thread (either using an existing
- // rtc::Thread, or by calling rtc::Thread::Current()->ProcessMessages).
- //
- // |network_manager| is used to determine which network interfaces are
- // available. This is used for ICE, for example. If null, a default
- // implementation will be used. Only accessed on |network_thread|.
- //
- // |socket_factory| is used (on the network thread) for creating sockets. If
- // it's null, a default implementation will be used, which assumes
- // |network_thread| is a normal rtc::Thread.
- //
- // |adm| is optional, and allows a different audio device implementation to
- // be injected; otherwise a platform-specific module will be used that will
- // use the default audio input.
- //
- // |audio_encoder_factory| and |audio_decoder_factory| are used to
- // instantiate audio codecs; they determine what codecs are supported.
- //
- // Note that the OrtcFactoryInterface does not take ownership of any of the
- // objects passed in by raw pointer, and as previously stated, these objects
- // can't be destroyed before the factory is.
- static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create(
- rtc::Thread* network_thread,
- rtc::Thread* signaling_thread,
- rtc::NetworkManager* network_manager,
- rtc::PacketSocketFactory* socket_factory,
- AudioDeviceModule* adm,
- rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
- rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
-
- // Constructor for convenience which uses default implementations where
- // possible (though does still require that the current thread runs a message
- // loop; see above).
- static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create(
- rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
- rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
- return Create(nullptr, nullptr, nullptr, nullptr, nullptr,
- audio_encoder_factory, audio_decoder_factory);
- }
-
- virtual ~OrtcFactoryInterface() {}
-
- // Creates an RTP transport controller, which is used in calls to
- // CreateRtpTransport methods. If your application has some notion of a
- // "call", you should create one transport controller per call.
- //
- // However, if you only are using one RtpTransport object, this doesn't need
- // to be called explicitly; CreateRtpTransport will create one automatically
- // if |rtp_transport_controller| is null. See below.
- //
- // TODO(deadbeef): Add MediaConfig and RtcEventLog arguments?
- virtual RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
- CreateRtpTransportController() = 0;
-
- // Creates an RTP transport using the provided packet transports and
- // transport controller.
- //
- // |rtp| will be used for sending RTP packets, and |rtcp| for RTCP packets.
- //
- // |rtp| can't be null. |rtcp| must be non-null if and only if
- // |rtp_parameters.rtcp.mux| is false, indicating that RTCP muxing isn't used.
- // Note that if RTCP muxing isn't enabled initially, it can still enabled
- // later through SetParameters.
- //
- // If |transport_controller| is null, one will automatically be created, and
- // its lifetime managed by the returned RtpTransport. This should only be
- // done if a single RtpTransport is being used to communicate with the remote
- // endpoint.
- virtual RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateRtpTransport(
- const RtpTransportParameters& rtp_parameters,
- PacketTransportInterface* rtp,
- PacketTransportInterface* rtcp,
- RtpTransportControllerInterface* transport_controller) = 0;
-
- // Creates an SrtpTransport which is an RTP transport that uses SRTP.
- virtual RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
- CreateSrtpTransport(
- const RtpTransportParameters& rtp_parameters,
- PacketTransportInterface* rtp,
- PacketTransportInterface* rtcp,
- RtpTransportControllerInterface* transport_controller) = 0;
-
- // Returns the capabilities of an RTP sender of type |kind|. These
- // capabilities can be used to determine what RtpParameters to use to create
- // an RtpSender.
- //
- // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
- virtual RtpCapabilities GetRtpSenderCapabilities(
- cricket::MediaType kind) const = 0;
-
- // Creates an RTP sender with |track|. Will not start sending until Send is
- // called. This is provided as a convenience; it's equivalent to calling
- // CreateRtpSender with a kind (see below), followed by SetTrack.
- //
- // |track| and |transport| must not be null.
- virtual RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
- rtc::scoped_refptr<MediaStreamTrackInterface> track,
- RtpTransportInterface* transport) = 0;
-
- // Overload of CreateRtpSender allows creating the sender without a track.
- //
- // |kind| must be MEDIA_TYPE_AUDIO or MEDIA_TYPE_VIDEO.
- virtual RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
- cricket::MediaType kind,
- RtpTransportInterface* transport) = 0;
-
- // Returns the capabilities of an RTP receiver of type |kind|. These
- // capabilities can be used to determine what RtpParameters to use to create
- // an RtpReceiver.
- //
- // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
- virtual RtpCapabilities GetRtpReceiverCapabilities(
- cricket::MediaType kind) const = 0;
-
- // Creates an RTP receiver of type |kind|. Will not start receiving media
- // until Receive is called.
- //
- // |kind| must be MEDIA_TYPE_AUDIO or MEDIA_TYPE_VIDEO.
- //
- // |transport| must not be null.
- virtual RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
- CreateRtpReceiver(cricket::MediaType kind,
- RtpTransportInterface* transport) = 0;
-
- // Create a UDP transport with IP address family |family|, using a port
- // within the specified range.
- //
- // |family| must be AF_INET or AF_INET6.
- //
- // |min_port|/|max_port| values of 0 indicate no range restriction.
- //
- // Returns an error if the transport wasn't successfully created.
- virtual RTCErrorOr<std::unique_ptr<UdpTransportInterface>>
- CreateUdpTransport(int family, uint16_t min_port, uint16_t max_port) = 0;
-
- // Method for convenience that has no port range restrictions.
- RTCErrorOr<std::unique_ptr<UdpTransportInterface>> CreateUdpTransport(
- int family) {
- return CreateUdpTransport(family, 0, 0);
- }
-
- // NOTE: The methods below to create tracks/sources return scoped_refptrs
- // rather than unique_ptrs, because these interfaces are also used with
- // PeerConnection, where everything is ref-counted.
-
- // Creates a audio source representing the default microphone input.
- // |options| decides audio processing settings.
- virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
- const cricket::AudioOptions& options) = 0;
-
- // Version of the above method that uses default options.
- rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource() {
- return CreateAudioSource(cricket::AudioOptions());
- }
-
- // Creates a new local video track wrapping |source|. The same |source| can
- // be used in several tracks.
- virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
- const std::string& id,
- VideoTrackSourceInterface* source) = 0;
-
- // Creates an new local audio track wrapping |source|.
- virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
- const std::string& id,
- AudioSourceInterface* source) = 0;
-};
-
-} // namespace webrtc
-
-#endif // API_ORTC_ORTCFACTORYINTERFACE_H_
diff --git a/api/ortc/ortcrtpreceiverinterface.h b/api/ortc/ortcrtpreceiverinterface.h
deleted file mode 100644
index 59ff977..0000000
--- a/api/ortc/ortcrtpreceiverinterface.h
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-// This file contains interfaces for RtpReceivers:
-// http://publications.ortc.org/2016/20161202/#rtcrtpreceiver*
-//
-// However, underneath the RtpReceiver is an RtpTransport, rather than a
-// DtlsTransport. This is to allow different types of RTP transports (besides
-// DTLS-SRTP) to be used.
-
-#ifndef API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
-#define API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
-
-#include "api/mediastreaminterface.h"
-#include "api/mediatypes.h"
-#include "api/ortc/rtptransportinterface.h"
-#include "api/rtcerror.h"
-#include "api/rtpparameters.h"
-
-namespace webrtc {
-
-// Note: Since receiver capabilities may depend on how the OrtcFactory was
-// created, instead of a static "GetCapabilities" method on this interface,
-// there is a "GetRtpReceiverCapabilities" method on the OrtcFactory.
-class OrtcRtpReceiverInterface {
- public:
- virtual ~OrtcRtpReceiverInterface() {}
-
- // Returns a track representing the media received by this receiver.
- //
- // Currently, this will return null until Receive has been successfully
- // called. Also, a new track will be created every time the primary SSRC
- // changes.
- //
- // If encodings are removed, GetTrack will return null. Though deactivating
- // an encoding (setting |active| to false) will not do this.
- //
- // In the future, these limitations will be fixed, and GetTrack will return
- // the same track for the lifetime of the RtpReceiver. So it's not
- // recommended to write code that depends on this non-standard behavior.
- virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
-
- // Once supported, will switch to receiving media on a new transport.
- // However, this is not currently supported and will always return an error.
- virtual RTCError SetTransport(RtpTransportInterface* transport) = 0;
- // Returns previously set (or constructed-with) transport.
- virtual RtpTransportInterface* GetTransport() const = 0;
-
- // Start receiving media with |parameters| (if |parameters| contains an
- // active encoding).
- //
- // There are no limitations to how the parameters can be changed after the
- // initial call to Receive, as long as they're valid (for example, they can't
- // use the same payload type for two codecs).
- virtual RTCError Receive(const RtpParameters& parameters) = 0;
- // Returns parameters that were last successfully passed into Receive, or
- // empty parameters if that hasn't yet occurred.
- //
- // Note that for parameters that are described as having an "implementation
- // default" value chosen, GetParameters() will return those chosen defaults,
- // with the exception of SSRCs which have special behavior. See
- // rtpparameters.h for more details.
- virtual RtpParameters GetParameters() const = 0;
-
- // Audio or video receiver?
- //
- // Once GetTrack() starts always returning a track, this method will be
- // redundant, as one can call "GetTrack()->kind()". However, it's still a
- // nice convenience, and is symmetric with OrtcRtpSenderInterface::GetKind.
- virtual cricket::MediaType GetKind() const = 0;
-
- // TODO(deadbeef): GetContributingSources
-};
-
-} // namespace webrtc
-
-#endif // API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
diff --git a/api/ortc/ortcrtpsenderinterface.h b/api/ortc/ortcrtpsenderinterface.h
deleted file mode 100644
index fd4dfaa..0000000
--- a/api/ortc/ortcrtpsenderinterface.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-// This file contains interfaces for RtpSenders:
-// http://publications.ortc.org/2016/20161202/#rtcrtpsender*
-//
-// However, underneath the RtpSender is an RtpTransport, rather than a
-// DtlsTransport. This is to allow different types of RTP transports (besides
-// DTLS-SRTP) to be used.
-
-#ifndef API_ORTC_ORTCRTPSENDERINTERFACE_H_
-#define API_ORTC_ORTCRTPSENDERINTERFACE_H_
-
-#include "api/mediastreaminterface.h"
-#include "api/mediatypes.h"
-#include "api/ortc/rtptransportinterface.h"
-#include "api/rtcerror.h"
-#include "api/rtpparameters.h"
-
-namespace webrtc {
-
-// Note: Since sender capabilities may depend on how the OrtcFactory was
-// created, instead of a static "GetCapabilities" method on this interface,
-// there is a "GetRtpSenderCapabilities" method on the OrtcFactory.
-class OrtcRtpSenderInterface {
- public:
- virtual ~OrtcRtpSenderInterface() {}
-
- // Sets the source of media that will be sent by this sender.
- //
- // If Send has already been called, will immediately switch to sending this
- // track. If |track| is null, will stop sending media.
- //
- // Returns INVALID_PARAMETER error if an audio track is set on a video
- // RtpSender, or vice-versa.
- virtual RTCError SetTrack(MediaStreamTrackInterface* track) = 0;
- // Returns previously set (or constructed-with) track.
- virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
-
- // Once supported, will switch to sending media on a new transport. However,
- // this is not currently supported and will always return an error.
- virtual RTCError SetTransport(RtpTransportInterface* transport) = 0;
- // Returns previously set (or constructed-with) transport.
- virtual RtpTransportInterface* GetTransport() const = 0;
-
- // Start sending media with |parameters| (if |parameters| contains an active
- // encoding).
- //
- // There are no limitations to how the parameters can be changed after the
- // initial call to Send, as long as they're valid (for example, they can't
- // use the same payload type for two codecs).
- virtual RTCError Send(const RtpParameters& parameters) = 0;
- // Returns parameters that were last successfully passed into Send, or empty
- // parameters if that hasn't yet occurred.
- //
- // Note that for parameters that are described as having an "implementation
- // default" value chosen, GetParameters() will return those chosen defaults,
- // with the exception of SSRCs which have special behavior. See
- // rtpparameters.h for more details.
- virtual RtpParameters GetParameters() const = 0;
-
- // Audio or video sender?
- virtual cricket::MediaType GetKind() const = 0;
-
- // TODO(deadbeef): SSRC conflict signal.
-};
-
-} // namespace webrtc
-
-#endif // API_ORTC_ORTCRTPSENDERINTERFACE_H_
diff --git a/api/ortc/rtptransportcontrollerinterface.h b/api/ortc/rtptransportcontrollerinterface.h
deleted file mode 100644
index 85f37fa..0000000
--- a/api/ortc/rtptransportcontrollerinterface.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#ifndef API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
-#define API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
-
-#include <vector>
-
-#include "api/ortc/rtptransportinterface.h"
-
-namespace webrtc {
-
-class RtpTransportControllerAdapter;
-
-// Used to group RTP transports between a local endpoint and the same remote
-// endpoint, for the purpose of sharing bandwidth estimation and other things.
-//
-// Comparing this to the PeerConnection model, non-budled audio/video would use
-// two RtpTransports with a single RtpTransportController, whereas bundled
-// media would use a single RtpTransport, and two PeerConnections would use
-// independent RtpTransportControllers.
-//
-// RtpTransports are associated with this controller when they're created, by
-// passing the controller into OrtcFactory's relevant "CreateRtpTransport"
-// method. When a transport is destroyed, it's automatically disassociated.
-// GetTransports returns all currently associated transports.
-//
-// This is the RTP equivalent of "IceTransportController" in ORTC; RtpTransport
-// is to RtpTransportController as IceTransport is to IceTransportController.
-class RtpTransportControllerInterface {
- public:
- virtual ~RtpTransportControllerInterface() {}
-
- // Returns all transports associated with this controller (see explanation
- // above). No ordering is guaranteed.
- virtual std::vector<RtpTransportInterface*> GetTransports() const = 0;
-
- protected:
- // Only for internal use. Returns a pointer to an internal interface, for use
- // by the implementation.
- virtual RtpTransportControllerAdapter* GetInternal() = 0;
-
- // Classes that can use this internal interface.
- friend class OrtcFactory;
- friend class RtpTransportAdapter;
-};
-
-} // namespace webrtc
-
-#endif // API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
diff --git a/api/ortc/sessiondescription.cc b/api/ortc/sessiondescription.cc
deleted file mode 100644
index 1078884..0000000
--- a/api/ortc/sessiondescription.cc
+++ /dev/null
@@ -1,13 +0,0 @@
-/*
- * Copyright 2017 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/ortc/sessiondescription.h"
-
-namespace webrtc {}
diff --git a/api/ortc/sessiondescription.h b/api/ortc/sessiondescription.h
deleted file mode 100644
index ebbaa27..0000000
--- a/api/ortc/sessiondescription.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#ifndef API_ORTC_SESSIONDESCRIPTION_H_
-#define API_ORTC_SESSIONDESCRIPTION_H_
-
-#include <string>
-#include <utility>
-
-namespace webrtc {
-
-// A structured representation of an SDP session description.
-class SessionDescription {
- public:
- SessionDescription(int64_t session_id, std::string session_version)
- : session_id_(session_id), session_version_(std::move(session_version)) {}
-
- // https://tools.ietf.org/html/rfc4566#section-5.2
- // o=<username> <sess-id> <sess-version> <nettype> <addrtype>
- // <unicast-address>
- // session_id_ is the "sess-id" field.
- // session_version_ is the "sess-version" field.
- int64_t session_id() { return session_id_; }
- void set_session_id(int64_t session_id) { session_id_ = session_id; }
-
- const std::string& session_version() const { return session_version_; }
- void set_session_version(std::string session_version) {
- session_version_ = std::move(session_version);
- }
-
- private:
- int64_t session_id_;
- std::string session_version_;
-};
-
-} // namespace webrtc
-
-#endif // API_ORTC_SESSIONDESCRIPTION_H_
diff --git a/api/ortc/sessiondescription_unittest.cc b/api/ortc/sessiondescription_unittest.cc
deleted file mode 100644
index e4611c6..0000000
--- a/api/ortc/sessiondescription_unittest.cc
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- * Copyright 2017 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/ortc/sessiondescription.h"
-#include "test/gtest.h"
-
-namespace webrtc {
-
-class SessionDescriptionTest : public testing::Test {};
-
-TEST_F(SessionDescriptionTest, CreateSessionDescription) {
- SessionDescription s(-1, "0");
- EXPECT_EQ(-1, s.session_id());
- EXPECT_EQ("0", s.session_version());
-}
-} // namespace webrtc
diff --git a/api/ortc/udptransportinterface.h b/api/ortc/udptransportinterface.h
deleted file mode 100644
index f246a25..0000000
--- a/api/ortc/udptransportinterface.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#ifndef API_ORTC_UDPTRANSPORTINTERFACE_H_
-#define API_ORTC_UDPTRANSPORTINTERFACE_H_
-
-#include "api/ortc/packettransportinterface.h"
-#include "api/proxy.h"
-#include "rtc_base/socketaddress.h"
-
-namespace webrtc {
-
-// Interface for a raw UDP transport (not using ICE), meaning a combination of
-// a local/remote IP address/port.
-//
-// An instance can be instantiated using OrtcFactory.
-//
-// Each instance reserves a UDP port, which will be freed when the
-// UdpTransportInterface destructor is called.
-//
-// Calling SetRemoteAddress sets the destination of outgoing packets; without a
-// destination, packets can't be sent, but they can be received.
-class UdpTransportInterface : public virtual PacketTransportInterface {
- public:
- // Get the address of the socket allocated for this transport.
- virtual rtc::SocketAddress GetLocalAddress() const = 0;
-
- // Sets the address to which packets will be delivered.
- //
- // Calling with a "nil" (default-constructed) address is legal, and unsets
- // any previously set destination.
- //
- // However, calling with an incomplete address (port or IP not set) will
- // fail.
- virtual bool SetRemoteAddress(const rtc::SocketAddress& dest) = 0;
- // Simple getter. If never set, returns nil address.
- virtual rtc::SocketAddress GetRemoteAddress() const = 0;
-};
-
-} // namespace webrtc
-
-#endif // API_ORTC_UDPTRANSPORTINTERFACE_H_
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index 252d182..f0033f1 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -76,8 +76,6 @@
"base/turnport.cc",
"base/turnport.h",
"base/udpport.h",
- "base/udptransport.cc",
- "base/udptransport.h",
"client/basicportallocator.cc",
"client/basicportallocator.h",
"client/relayportfactoryinterface.h",
@@ -176,7 +174,6 @@
"base/transportdescriptionfactory_unittest.cc",
"base/turnport_unittest.cc",
"base/turnserver_unittest.cc",
- "base/udptransport_unittest.cc",
"client/basicportallocator_unittest.cc",
]
deps = [
diff --git a/rtc_tools/network_tester/BUILD.gn b/rtc_tools/network_tester/BUILD.gn
index a5036dc..af84903 100644
--- a/rtc_tools/network_tester/BUILD.gn
+++ b/rtc_tools/network_tester/BUILD.gn
@@ -44,6 +44,7 @@
"../../p2p",
"../../rtc_base:checks",
"../../rtc_base:protobuf_utils",
+ "../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:sequenced_task_checker",
diff --git a/rtc_tools/network_tester/test_controller.cc b/rtc_tools/network_tester/test_controller.cc
index 9d8161b..9bfdfa7 100644
--- a/rtc_tools/network_tester/test_controller.cc
+++ b/rtc_tools/network_tester/test_controller.cc
@@ -10,6 +10,9 @@
#include "rtc_tools/network_tester/test_controller.h"
+#include "absl/types/optional.h"
+#include "rtc_base/thread.h"
+
namespace webrtc {
TestController::TestController(int min_port,
@@ -24,17 +27,15 @@
RTC_DCHECK_RUN_ON(&test_controller_thread_checker_);
packet_sender_checker_.Detach();
send_data_.fill(42);
- auto socket =
+ udp_socket_ =
std::unique_ptr<rtc::AsyncPacketSocket>(socket_factory_.CreateUdpSocket(
rtc::SocketAddress(rtc::GetAnyIP(AF_INET), 0), min_port, max_port));
- socket->SignalReadPacket.connect(this, &TestController::OnReadPacket);
- udp_transport_.reset(
- new cricket::UdpTransport("network tester transport", std::move(socket)));
+ udp_socket_->SignalReadPacket.connect(this, &TestController::OnReadPacket);
}
void TestController::SendConnectTo(const std::string& hostname, int port) {
RTC_DCHECK_RUN_ON(&test_controller_thread_checker_);
- udp_transport_->SetRemoteAddress(rtc::SocketAddress(hostname, port));
+ remote_address_ = rtc::SocketAddress(hostname, port);
NetworkTesterPacket packet;
packet.set_type(NetworkTesterPacket::HAND_SHAKING);
SendData(packet, absl::nullopt);
@@ -57,8 +58,8 @@
packet.SerializeToArray(&send_data_[1], std::numeric_limits<char>::max());
if (data_size && *data_size > packet_size)
packet_size = *data_size;
- udp_transport_->SendPacket(send_data_.data(), packet_size,
- rtc::PacketOptions(), 0);
+ udp_socket_->SendTo((const void*)send_data_.data(), packet_size,
+ remote_address_, rtc::PacketOptions());
}
void TestController::OnTestDone() {
@@ -91,7 +92,7 @@
case NetworkTesterPacket::HAND_SHAKING: {
NetworkTesterPacket packet;
packet.set_type(NetworkTesterPacket::TEST_START);
- udp_transport_->SetRemoteAddress(remote_addr);
+ remote_address_ = remote_addr;
SendData(packet, absl::nullopt);
packet_sender_.reset(new PacketSender(this, config_file_path_));
packet_sender_->StartSending();
diff --git a/rtc_tools/network_tester/test_controller.h b/rtc_tools/network_tester/test_controller.h
index 06a83e9..a65272a 100644
--- a/rtc_tools/network_tester/test_controller.h
+++ b/rtc_tools/network_tester/test_controller.h
@@ -18,7 +18,7 @@
#include <utility>
#include "p2p/base/basicpacketsocketfactory.h"
-#include "p2p/base/udptransport.h"
+#include "rtc_base/asyncpacketsocket.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_tools/network_tester/packet_logger.h"
@@ -70,7 +70,8 @@
bool local_test_done_ RTC_GUARDED_BY(local_test_done_lock_);
bool remote_test_done_;
std::array<char, kEthernetMtu> send_data_;
- std::unique_ptr<cricket::UdpTransport> udp_transport_;
+ std::unique_ptr<rtc::AsyncPacketSocket> udp_socket_;
+ rtc::SocketAddress remote_address_;
std::unique_ptr<PacketSender> packet_sender_;
RTC_DISALLOW_COPY_AND_ASSIGN(TestController);
