Move RtpExtension to api/ directory and config.h/.cc to call/.
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Original-Commit-Position: refs/heads/master@{#19639}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 1acbd68718476b2754a7872fb72e3a8a74166eb9
diff --git a/BUILD.gn b/BUILD.gn
index b5c832d..4216bd3 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -347,7 +347,6 @@
sources = [
"common_types.cc",
"common_types.h",
- "config.cc",
"config.h",
"typedefs.h",
]
@@ -384,9 +383,6 @@
rtc_test("rtc_unittests") {
testonly = true
- sources = [
- "config_unittest.cc",
- ]
deps = [
":webrtc_common",
diff --git a/DEPS b/DEPS
index 276991d..0459f4c 100644
--- a/DEPS
+++ b/DEPS
@@ -9,8 +9,8 @@
"+libyuv",
"-webrtc", # Has to be disabled; otherwise all dirs below will be allowed.
# Individual headers that will be moved out of here, see webrtc:4243.
+ "+webrtc/call/rtp_config.h",
"+webrtc/common_types.h",
- "+webrtc/config.h",
"+webrtc/transport.h",
"+webrtc/typedefs.h",
"+webrtc/voice_engine_configurations.h",
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 3d19f03..e58f393 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -62,6 +62,7 @@
"proxy.h",
"rtcerror.cc",
"rtcerror.h",
+ "rtpparameters.cc",
"rtpparameters.h",
"rtpreceiverinterface.h",
"rtpsender.h",
@@ -258,6 +259,7 @@
"ortc/mediadescription_unittest.cc",
"ortc/sessiondescription_unittest.cc",
"rtcerror_unittest.cc",
+ "rtpparameters_unittest.cc",
]
if (!build_with_chromium && is_clang) {
diff --git a/api/rtpparameters.cc b/api/rtpparameters.cc
new file mode 100644
index 0000000..29b14fb
--- /dev/null
+++ b/api/rtpparameters.cc
@@ -0,0 +1,190 @@
+/*
+ * Copyright (c) 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 "webrtc/api/rtpparameters.h"
+
+#include <algorithm>
+#include <sstream>
+#include <string>
+
+#include "webrtc/rtc_base/checks.h"
+
+namespace webrtc {
+
+RtcpFeedback::RtcpFeedback() {}
+RtcpFeedback::RtcpFeedback(RtcpFeedbackType type) : type(type) {}
+RtcpFeedback::RtcpFeedback(RtcpFeedbackType type,
+ RtcpFeedbackMessageType message_type)
+ : type(type), message_type(message_type) {}
+RtcpFeedback::~RtcpFeedback() {}
+
+RtpCodecCapability::RtpCodecCapability() {}
+RtpCodecCapability::~RtpCodecCapability() {}
+
+RtpHeaderExtensionCapability::RtpHeaderExtensionCapability() {}
+RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
+ const std::string& uri)
+ : uri(uri) {}
+RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
+ const std::string& uri,
+ int preferred_id)
+ : uri(uri), preferred_id(preferred_id) {}
+RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() {}
+
+RtpExtension::RtpExtension() {}
+RtpExtension::RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
+RtpExtension::RtpExtension(const std::string& uri, int id, bool encrypt)
+ : uri(uri), id(id), encrypt(encrypt) {}
+RtpExtension::~RtpExtension() {}
+
+RtpFecParameters::RtpFecParameters() {}
+RtpFecParameters::RtpFecParameters(FecMechanism mechanism)
+ : mechanism(mechanism) {}
+RtpFecParameters::RtpFecParameters(FecMechanism mechanism, uint32_t ssrc)
+ : ssrc(ssrc), mechanism(mechanism) {}
+RtpFecParameters::~RtpFecParameters() {}
+
+RtpRtxParameters::RtpRtxParameters() {}
+RtpRtxParameters::RtpRtxParameters(uint32_t ssrc) : ssrc(ssrc) {}
+RtpRtxParameters::~RtpRtxParameters() {}
+
+RtpEncodingParameters::RtpEncodingParameters() {}
+RtpEncodingParameters::~RtpEncodingParameters() {}
+
+RtpCodecParameters::RtpCodecParameters() {}
+RtpCodecParameters::~RtpCodecParameters() {}
+
+RtpCapabilities::RtpCapabilities() {}
+RtpCapabilities::~RtpCapabilities() {}
+
+RtpParameters::RtpParameters() {}
+RtpParameters::~RtpParameters() {}
+
+std::string RtpExtension::ToString() const {
+ std::stringstream ss;
+ ss << "{uri: " << uri;
+ ss << ", id: " << id;
+ if (encrypt) {
+ ss << ", encrypt";
+ }
+ ss << '}';
+ return ss.str();
+}
+
+const char RtpExtension::kAudioLevelUri[] =
+ "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+const int RtpExtension::kAudioLevelDefaultId = 1;
+
+const char RtpExtension::kTimestampOffsetUri[] =
+ "urn:ietf:params:rtp-hdrext:toffset";
+const int RtpExtension::kTimestampOffsetDefaultId = 2;
+
+const char RtpExtension::kAbsSendTimeUri[] =
+ "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
+const int RtpExtension::kAbsSendTimeDefaultId = 3;
+
+const char RtpExtension::kVideoRotationUri[] = "urn:3gpp:video-orientation";
+const int RtpExtension::kVideoRotationDefaultId = 4;
+
+const char RtpExtension::kTransportSequenceNumberUri[] =
+ "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
+const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
+
+// This extension allows applications to adaptively limit the playout delay
+// on frames as per the current needs. For example, a gaming application
+// has very different needs on end-to-end delay compared to a video-conference
+// application.
+const char RtpExtension::kPlayoutDelayUri[] =
+ "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
+const int RtpExtension::kPlayoutDelayDefaultId = 6;
+
+const char RtpExtension::kVideoContentTypeUri[] =
+ "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
+const int RtpExtension::kVideoContentTypeDefaultId = 7;
+
+const char RtpExtension::kVideoTimingUri[] =
+ "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
+const int RtpExtension::kVideoTimingDefaultId = 8;
+
+const char RtpExtension::kEncryptHeaderExtensionsUri[] =
+ "urn:ietf:params:rtp-hdrext:encrypt";
+
+const int RtpExtension::kMinId = 1;
+const int RtpExtension::kMaxId = 14;
+
+bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
+ return uri == webrtc::RtpExtension::kAudioLevelUri ||
+ uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
+}
+
+bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
+ return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
+ uri == webrtc::RtpExtension::kAbsSendTimeUri ||
+ uri == webrtc::RtpExtension::kVideoRotationUri ||
+ uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
+ uri == webrtc::RtpExtension::kPlayoutDelayUri ||
+ uri == webrtc::RtpExtension::kVideoContentTypeUri ||
+ uri == webrtc::RtpExtension::kVideoTimingUri;
+}
+
+bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
+ return uri == webrtc::RtpExtension::kAudioLevelUri ||
+ uri == webrtc::RtpExtension::kTimestampOffsetUri ||
+#if !defined(ENABLE_EXTERNAL_AUTH)
+ // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri"
+ // here and filter out later if external auth is really used in
+ // srtpfilter. External auth is used by Chromium and replaces the
+ // extension header value of "kAbsSendTimeUri", so it must not be
+ // encrypted (which can't be done by Chromium).
+ uri == webrtc::RtpExtension::kAbsSendTimeUri ||
+#endif
+ uri == webrtc::RtpExtension::kVideoRotationUri ||
+ uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
+ uri == webrtc::RtpExtension::kPlayoutDelayUri ||
+ uri == webrtc::RtpExtension::kVideoContentTypeUri;
+}
+
+const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
+ const std::vector<RtpExtension>& extensions,
+ const std::string& uri) {
+ for (const auto& extension : extensions) {
+ if (extension.uri == uri) {
+ return &extension;
+ }
+ }
+ return nullptr;
+}
+
+std::vector<RtpExtension> RtpExtension::FilterDuplicateNonEncrypted(
+ const std::vector<RtpExtension>& extensions) {
+ std::vector<RtpExtension> filtered;
+ for (auto extension = extensions.begin(); extension != extensions.end();
+ ++extension) {
+ if (extension->encrypt) {
+ filtered.push_back(*extension);
+ continue;
+ }
+
+ // Only add non-encrypted extension if no encrypted with the same URI
+ // is also present...
+ if (std::find_if(extension + 1, extensions.end(),
+ [extension](const RtpExtension& check) {
+ return extension->uri == check.uri;
+ }) != extensions.end()) {
+ continue;
+ }
+
+ // ...and has not been added before.
+ if (!FindHeaderExtensionByUri(filtered, extension->uri)) {
+ filtered.push_back(*extension);
+ }
+ }
+ return filtered;
+}
+} // namespace webrtc
diff --git a/api/rtpparameters.h b/api/rtpparameters.h
index 94d9d73..46f7139 100644
--- a/api/rtpparameters.h
+++ b/api/rtpparameters.h
@@ -16,7 +16,6 @@
#include <vector>
#include "webrtc/api/mediatypes.h"
-#include "webrtc/config.h"
#include "webrtc/rtc_base/optional.h"
namespace webrtc {
@@ -85,10 +84,10 @@
rtc::Optional<RtcpFeedbackMessageType> message_type;
// Constructors for convenience.
- RtcpFeedback() {}
- explicit RtcpFeedback(RtcpFeedbackType type) : type(type) {}
- RtcpFeedback(RtcpFeedbackType type, RtcpFeedbackMessageType message_type)
- : type(type), message_type(message_type) {}
+ RtcpFeedback();
+ explicit RtcpFeedback(RtcpFeedbackType type);
+ RtcpFeedback(RtcpFeedbackType type, RtcpFeedbackMessageType message_type);
+ ~RtcpFeedback();
bool operator==(const RtcpFeedback& o) const {
return type == o.type && message_type == o.message_type;
@@ -100,6 +99,9 @@
// RtpParameters. This represents the static capabilities of an endpoint's
// implementation of a codec.
struct RtpCodecCapability {
+ RtpCodecCapability();
+ ~RtpCodecCapability();
+
// Build MIME "type/subtype" string from |name| and |kind|.
std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
@@ -196,10 +198,10 @@
bool preferred_encrypt = false;
// Constructors for convenience.
- RtpHeaderExtensionCapability() = default;
- explicit RtpHeaderExtensionCapability(const std::string& uri) : uri(uri) {}
- RtpHeaderExtensionCapability(const std::string& uri, int preferred_id)
- : uri(uri), preferred_id(preferred_id) {}
+ RtpHeaderExtensionCapability();
+ explicit RtpHeaderExtensionCapability(const std::string& uri);
+ RtpHeaderExtensionCapability(const std::string& uri, int preferred_id);
+ ~RtpHeaderExtensionCapability();
bool operator==(const RtpHeaderExtensionCapability& o) const {
return uri == o.uri && preferred_id == o.preferred_id &&
@@ -210,6 +212,83 @@
}
};
+// RTP header extension, see RFC 5285.
+struct RtpExtension {
+ RtpExtension();
+ RtpExtension(const std::string& uri, int id);
+ RtpExtension(const std::string& uri, int id, bool encrypt);
+ ~RtpExtension();
+ std::string ToString() const;
+ bool operator==(const RtpExtension& rhs) const {
+ return uri == rhs.uri && id == rhs.id && encrypt == rhs.encrypt;
+ }
+ static bool IsSupportedForAudio(const std::string& uri);
+ static bool IsSupportedForVideo(const std::string& uri);
+ // Return "true" if the given RTP header extension URI may be encrypted.
+ static bool IsEncryptionSupported(const std::string& uri);
+
+ // Returns the named header extension if found among all extensions,
+ // nullptr otherwise.
+ static const RtpExtension* FindHeaderExtensionByUri(
+ const std::vector<RtpExtension>& extensions,
+ const std::string& uri);
+
+ // Return a list of RTP header extensions with the non-encrypted extensions
+ // removed if both the encrypted and non-encrypted extension is present for
+ // the same URI.
+ static std::vector<RtpExtension> FilterDuplicateNonEncrypted(
+ const std::vector<RtpExtension>& extensions);
+
+ // Header extension for audio levels, as defined in:
+ // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
+ static const char kAudioLevelUri[];
+ static const int kAudioLevelDefaultId;
+
+ // Header extension for RTP timestamp offset, see RFC 5450 for details:
+ // http://tools.ietf.org/html/rfc5450
+ static const char kTimestampOffsetUri[];
+ static const int kTimestampOffsetDefaultId;
+
+ // Header extension for absolute send time, see url for details:
+ // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+ static const char kAbsSendTimeUri[];
+ static const int kAbsSendTimeDefaultId;
+
+ // Header extension for coordination of video orientation, see url for
+ // details:
+ // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf
+ static const char kVideoRotationUri[];
+ static const int kVideoRotationDefaultId;
+
+ // Header extension for video content type. E.g. default or screenshare.
+ static const char kVideoContentTypeUri[];
+ static const int kVideoContentTypeDefaultId;
+
+ // Header extension for video timing.
+ static const char kVideoTimingUri[];
+ static const int kVideoTimingDefaultId;
+
+ // Header extension for transport sequence number, see url for details:
+ // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
+ static const char kTransportSequenceNumberUri[];
+ static const int kTransportSequenceNumberDefaultId;
+
+ static const char kPlayoutDelayUri[];
+ static const int kPlayoutDelayDefaultId;
+
+ // Encryption of Header Extensions, see RFC 6904 for details:
+ // https://tools.ietf.org/html/rfc6904
+ static const char kEncryptHeaderExtensionsUri[];
+
+ // Inclusive min and max IDs for one-byte header extensions, per RFC5285.
+ static const int kMinId;
+ static const int kMaxId;
+
+ std::string uri;
+ int id = 0;
+ bool encrypt = false;
+};
+
// See webrtc/config.h. Has "uri" and "id" fields.
// TODO(deadbeef): This is missing the "encrypt" flag, which is unimplemented.
typedef RtpExtension RtpHeaderExtensionParameters;
@@ -222,10 +301,10 @@
FecMechanism mechanism = FecMechanism::RED;
// Constructors for convenience.
- RtpFecParameters() = default;
- explicit RtpFecParameters(FecMechanism mechanism) : mechanism(mechanism) {}
- RtpFecParameters(FecMechanism mechanism, uint32_t ssrc)
- : ssrc(ssrc), mechanism(mechanism) {}
+ RtpFecParameters();
+ explicit RtpFecParameters(FecMechanism mechanism);
+ RtpFecParameters(FecMechanism mechanism, uint32_t ssrc);
+ ~RtpFecParameters();
bool operator==(const RtpFecParameters& o) const {
return ssrc == o.ssrc && mechanism == o.mechanism;
@@ -239,14 +318,18 @@
rtc::Optional<uint32_t> ssrc;
// Constructors for convenience.
- RtpRtxParameters() = default;
- explicit RtpRtxParameters(uint32_t ssrc) : ssrc(ssrc) {}
+ RtpRtxParameters();
+ explicit RtpRtxParameters(uint32_t ssrc);
+ ~RtpRtxParameters();
bool operator==(const RtpRtxParameters& o) const { return ssrc == o.ssrc; }
bool operator!=(const RtpRtxParameters& o) const { return !(*this == o); }
};
struct RtpEncodingParameters {
+ RtpEncodingParameters();
+ ~RtpEncodingParameters();
+
// If unset, a value is chosen by the implementation.
//
// Note that the chosen value is NOT returned by GetParameters, because it
@@ -338,6 +421,9 @@
};
struct RtpCodecParameters {
+ RtpCodecParameters();
+ ~RtpCodecParameters();
+
// Build MIME "type/subtype" string from |name| and |kind|.
std::string mime_type() const { return MediaTypeToString(kind) + "/" + name; }
@@ -400,6 +486,9 @@
// endpoint. An application can use these capabilities to construct an
// RtpParameters.
struct RtpCapabilities {
+ RtpCapabilities();
+ ~RtpCapabilities();
+
// Supported codecs.
std::vector<RtpCodecCapability> codecs;
@@ -422,6 +511,9 @@
// RtpParameters, because our API includes an additional "RtpTransport"
// abstraction on which RTCP parameters are set.
struct RtpParameters {
+ RtpParameters();
+ ~RtpParameters();
+
// Used when calling getParameters/setParameters with a PeerConnection
// RtpSender, to ensure that outdated parameters are not unintentionally
// applied successfully.
diff --git a/config_unittest.cc b/api/rtpparameters_unittest.cc
similarity index 92%
rename from config_unittest.cc
rename to api/rtpparameters_unittest.cc
index 03784d4..6cb90c2 100644
--- a/config_unittest.cc
+++ b/api/rtpparameters_unittest.cc
@@ -8,11 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/config.h"
+#include <utility>
-#include <vector>
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/test/gtest.h"
-#include "webrtc/rtc_base/gunit.h"
+namespace webrtc {
using webrtc::RtpExtension;
@@ -47,3 +48,4 @@
EXPECT_EQ(2u, filtered.size());
EXPECT_EQ(extensions, filtered);
}
+} // namespace webrtc
diff --git a/call/BUILD.gn b/call/BUILD.gn
index d86df1e..263f16d 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -38,6 +38,8 @@
rtc_source_set("rtp_interfaces") {
sources = [
"rtcp_packet_sink_interface.h",
+ "rtp_config.cc",
+ "rtp_config.h",
"rtp_packet_sink_interface.h",
"rtp_stream_receiver_controller_interface.h",
"rtp_transport_controller_send_interface.h",
@@ -100,6 +102,7 @@
public_deps = [
":call_interfaces",
"../api:call_api",
+ "../api:libjingle_peerconnection_api",
]
deps = [
@@ -107,6 +110,7 @@
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
+ ":video_stream_api",
"..:webrtc_common",
"../api:transport_api",
"../audio",
@@ -127,13 +131,17 @@
rtc_source_set("video_stream_api") {
sources = [
+ "video_config.cc",
+ "video_config.h",
"video_receive_stream.cc",
"video_receive_stream.h",
"video_send_stream.cc",
"video_send_stream.h",
]
deps = [
+ ":rtp_interfaces",
"../:webrtc_common",
+ "../api:libjingle_peerconnection_api",
"../api:transport_api",
"../common_video:common_video",
"../rtc_base:rtc_base_approved",
@@ -209,6 +217,7 @@
]
deps = [
":call_interfaces",
+ ":video_stream_api",
"..:webrtc_common",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../logging:rtc_event_log_api",
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index fdf1698..018ad35 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -18,9 +18,10 @@
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/api/call/transport.h"
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/api/rtpreceiverinterface.h"
+#include "webrtc/call/rtp_config.h"
#include "webrtc/common_types.h"
-#include "webrtc/config.h"
#include "webrtc/rtc_base/optional.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "webrtc/typedefs.h"
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index fa5b5ee..8720249 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -18,8 +18,10 @@
#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/call/transport.h"
-#include "webrtc/config.h"
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/call/rtp_config.h"
#include "webrtc/rtc_base/optional.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/call/call.cc b/call/call.cc
index b1f56d3..edb9bf3 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -26,7 +26,6 @@
#include "webrtc/call/flexfec_receive_stream_impl.h"
#include "webrtc/call/rtp_stream_receiver_controller.h"
#include "webrtc/call/rtp_transport_controller_send.h"
-#include "webrtc/config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 530ad29..29dd0e4 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -15,7 +15,7 @@
#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
#include "webrtc/call/call.h"
-#include "webrtc/config.h"
+#include "webrtc/call/video_config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
diff --git a/call/flexfec_receive_stream.h b/call/flexfec_receive_stream.h
index 84447ff..400bca4 100644
--- a/call/flexfec_receive_stream.h
+++ b/call/flexfec_receive_stream.h
@@ -17,8 +17,9 @@
#include <vector>
#include "webrtc/api/call/transport.h"
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/call/rtp_packet_sink_interface.h"
-#include "webrtc/config.h"
+#include "webrtc/common_types.h"
namespace webrtc {
diff --git a/call/rtp_config.cc b/call/rtp_config.cc
new file mode 100644
index 0000000..dc3ea21
--- /dev/null
+++ b/call/rtp_config.cc
@@ -0,0 +1,38 @@
+/*
+ * Copyright (c) 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 "webrtc/call/rtp_config.h"
+
+#include <sstream>
+
+namespace webrtc {
+
+std::string NackConfig::ToString() const {
+ std::stringstream ss;
+ ss << "{rtp_history_ms: " << rtp_history_ms;
+ ss << '}';
+ return ss.str();
+}
+
+std::string UlpfecConfig::ToString() const {
+ std::stringstream ss;
+ ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
+ ss << ", red_payload_type: " << red_payload_type;
+ ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
+ ss << '}';
+ return ss.str();
+}
+
+bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
+ return ulpfec_payload_type == other.ulpfec_payload_type &&
+ red_payload_type == other.red_payload_type &&
+ red_rtx_payload_type == other.red_rtx_payload_type;
+}
+} // namespace webrtc
diff --git a/call/rtp_config.h b/call/rtp_config.h
new file mode 100644
index 0000000..f33103d
--- /dev/null
+++ b/call/rtp_config.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 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 WEBRTC_CALL_RTP_CONFIG_H_
+#define WEBRTC_CALL_RTP_CONFIG_H_
+
+#include <string>
+
+namespace webrtc {
+// Settings for NACK, see RFC 4585 for details.
+struct NackConfig {
+ NackConfig() : rtp_history_ms(0) {}
+ std::string ToString() const;
+ // Send side: the time RTP packets are stored for retransmissions.
+ // Receive side: the time the receiver is prepared to wait for
+ // retransmissions.
+ // Set to '0' to disable.
+ int rtp_history_ms;
+};
+
+// Settings for ULPFEC forward error correction.
+// Set the payload types to '-1' to disable.
+struct UlpfecConfig {
+ UlpfecConfig()
+ : ulpfec_payload_type(-1),
+ red_payload_type(-1),
+ red_rtx_payload_type(-1) {}
+ std::string ToString() const;
+ bool operator==(const UlpfecConfig& other) const;
+
+ // Payload type used for ULPFEC packets.
+ int ulpfec_payload_type;
+
+ // Payload type used for RED packets.
+ int red_payload_type;
+
+ // RTX payload type for RED payload.
+ int red_rtx_payload_type;
+};
+} // namespace webrtc
+#endif // WEBRTC_CALL_RTP_CONFIG_H_
diff --git a/call/video_config.cc b/call/video_config.cc
new file mode 100644
index 0000000..e06d5e8
--- /dev/null
+++ b/call/video_config.cc
@@ -0,0 +1,139 @@
+/*
+ * Copyright (c) 2014 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 "webrtc/call/video_config.h"
+
+#include <algorithm>
+#include <sstream>
+#include <string>
+
+#include "webrtc/rtc_base/checks.h"
+
+namespace webrtc {
+VideoStream::VideoStream()
+ : width(0),
+ height(0),
+ max_framerate(-1),
+ min_bitrate_bps(-1),
+ target_bitrate_bps(-1),
+ max_bitrate_bps(-1),
+ max_qp(-1) {}
+
+VideoStream::~VideoStream() = default;
+
+std::string VideoStream::ToString() const {
+ std::stringstream ss;
+ ss << "{width: " << width;
+ ss << ", height: " << height;
+ ss << ", max_framerate: " << max_framerate;
+ ss << ", min_bitrate_bps:" << min_bitrate_bps;
+ ss << ", target_bitrate_bps:" << target_bitrate_bps;
+ ss << ", max_bitrate_bps:" << max_bitrate_bps;
+ ss << ", max_qp: " << max_qp;
+
+ ss << ", temporal_layer_thresholds_bps: [";
+ for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
+ ss << temporal_layer_thresholds_bps[i];
+ if (i != temporal_layer_thresholds_bps.size() - 1)
+ ss << ", ";
+ }
+ ss << ']';
+
+ ss << '}';
+ return ss.str();
+}
+
+VideoEncoderConfig::VideoEncoderConfig()
+ : content_type(ContentType::kRealtimeVideo),
+ encoder_specific_settings(nullptr),
+ min_transmit_bitrate_bps(0),
+ max_bitrate_bps(0),
+ number_of_streams(0) {}
+
+VideoEncoderConfig::VideoEncoderConfig(VideoEncoderConfig&&) = default;
+
+VideoEncoderConfig::~VideoEncoderConfig() = default;
+
+std::string VideoEncoderConfig::ToString() const {
+ std::stringstream ss;
+ ss << "{content_type: ";
+ switch (content_type) {
+ case ContentType::kRealtimeVideo:
+ ss << "kRealtimeVideo";
+ break;
+ case ContentType::kScreen:
+ ss << "kScreenshare";
+ break;
+ }
+ ss << ", encoder_specific_settings: ";
+ ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
+
+ ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
+ ss << '}';
+ return ss.str();
+}
+
+VideoEncoderConfig::VideoEncoderConfig(const VideoEncoderConfig&) = default;
+
+void VideoEncoderConfig::EncoderSpecificSettings::FillEncoderSpecificSettings(
+ VideoCodec* codec) const {
+ if (codec->codecType == kVideoCodecH264) {
+ FillVideoCodecH264(codec->H264());
+ } else if (codec->codecType == kVideoCodecVP8) {
+ FillVideoCodecVp8(codec->VP8());
+ } else if (codec->codecType == kVideoCodecVP9) {
+ FillVideoCodecVp9(codec->VP9());
+ } else {
+ RTC_NOTREACHED() << "Encoder specifics set/used for unknown codec type.";
+ }
+}
+
+void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecH264(
+ VideoCodecH264* h264_settings) const {
+ RTC_NOTREACHED();
+}
+
+void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecVp8(
+ VideoCodecVP8* vp8_settings) const {
+ RTC_NOTREACHED();
+}
+
+void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecVp9(
+ VideoCodecVP9* vp9_settings) const {
+ RTC_NOTREACHED();
+}
+
+VideoEncoderConfig::H264EncoderSpecificSettings::H264EncoderSpecificSettings(
+ const VideoCodecH264& specifics)
+ : specifics_(specifics) {}
+
+void VideoEncoderConfig::H264EncoderSpecificSettings::FillVideoCodecH264(
+ VideoCodecH264* h264_settings) const {
+ *h264_settings = specifics_;
+}
+
+VideoEncoderConfig::Vp8EncoderSpecificSettings::Vp8EncoderSpecificSettings(
+ const VideoCodecVP8& specifics)
+ : specifics_(specifics) {}
+
+void VideoEncoderConfig::Vp8EncoderSpecificSettings::FillVideoCodecVp8(
+ VideoCodecVP8* vp8_settings) const {
+ *vp8_settings = specifics_;
+}
+
+VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings(
+ const VideoCodecVP9& specifics)
+ : specifics_(specifics) {}
+
+void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
+ VideoCodecVP9* vp9_settings) const {
+ *vp9_settings = specifics_;
+}
+
+} // namespace webrtc
diff --git a/call/video_config.h b/call/video_config.h
new file mode 100644
index 0000000..cdff513
--- /dev/null
+++ b/call/video_config.h
@@ -0,0 +1,157 @@
+/*
+ * Copyright (c) 2013 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 WEBRTC_CALL_VIDEO_CONFIG_H_
+#define WEBRTC_CALL_VIDEO_CONFIG_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/rtc_base/basictypes.h"
+#include "webrtc/rtc_base/optional.h"
+#include "webrtc/rtc_base/refcount.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct VideoStream {
+ VideoStream();
+ ~VideoStream();
+ std::string ToString() const;
+
+ size_t width;
+ size_t height;
+ int max_framerate;
+
+ int min_bitrate_bps;
+ int target_bitrate_bps;
+ int max_bitrate_bps;
+
+ int max_qp;
+
+ // Bitrate thresholds for enabling additional temporal layers. Since these are
+ // thresholds in between layers, we have one additional layer. One threshold
+ // gives two temporal layers, one below the threshold and one above, two give
+ // three, and so on.
+ // The VideoEncoder may redistribute bitrates over the temporal layers so a
+ // bitrate threshold of 100k and an estimate of 105k does not imply that we
+ // get 100k in one temporal layer and 5k in the other, just that the bitrate
+ // in the first temporal layer should not exceed 100k.
+ // TODO(kthelgason): Apart from a special case for two-layer screencast these
+ // thresholds are not propagated to the VideoEncoder. To be implemented.
+ std::vector<int> temporal_layer_thresholds_bps;
+};
+
+class VideoEncoderConfig {
+ public:
+ // These are reference counted to permit copying VideoEncoderConfig and be
+ // kept alive until all encoder_specific_settings go out of scope.
+ // TODO(kthelgason): Consider removing the need for copying VideoEncoderConfig
+ // and use rtc::Optional for encoder_specific_settings instead.
+ class EncoderSpecificSettings : public rtc::RefCountInterface {
+ public:
+ // TODO(pbos): Remove FillEncoderSpecificSettings as soon as VideoCodec is
+ // not in use and encoder implementations ask for codec-specific structs
+ // directly.
+ void FillEncoderSpecificSettings(VideoCodec* codec_struct) const;
+
+ virtual void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const;
+ virtual void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const;
+ virtual void FillVideoCodecH264(VideoCodecH264* h264_settings) const;
+
+ private:
+ ~EncoderSpecificSettings() override {}
+ friend class VideoEncoderConfig;
+ };
+
+ class H264EncoderSpecificSettings : public EncoderSpecificSettings {
+ public:
+ explicit H264EncoderSpecificSettings(const VideoCodecH264& specifics);
+ void FillVideoCodecH264(VideoCodecH264* h264_settings) const override;
+
+ private:
+ VideoCodecH264 specifics_;
+ };
+
+ class Vp8EncoderSpecificSettings : public EncoderSpecificSettings {
+ public:
+ explicit Vp8EncoderSpecificSettings(const VideoCodecVP8& specifics);
+ void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const override;
+
+ private:
+ VideoCodecVP8 specifics_;
+ };
+
+ class Vp9EncoderSpecificSettings : public EncoderSpecificSettings {
+ public:
+ explicit Vp9EncoderSpecificSettings(const VideoCodecVP9& specifics);
+ void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const override;
+
+ private:
+ VideoCodecVP9 specifics_;
+ };
+
+ enum class ContentType {
+ kRealtimeVideo,
+ kScreen,
+ };
+
+ class VideoStreamFactoryInterface : public rtc::RefCountInterface {
+ public:
+ // An implementation should return a std::vector<VideoStream> with the
+ // wanted VideoStream settings for the given video resolution.
+ // The size of the vector may not be larger than
+ // |encoder_config.number_of_streams|.
+ virtual std::vector<VideoStream> CreateEncoderStreams(
+ int width,
+ int height,
+ const VideoEncoderConfig& encoder_config) = 0;
+
+ protected:
+ ~VideoStreamFactoryInterface() override {}
+ };
+
+ VideoEncoderConfig& operator=(VideoEncoderConfig&&) = default;
+ VideoEncoderConfig& operator=(const VideoEncoderConfig&) = delete;
+
+ // Mostly used by tests. Avoid creating copies if you can.
+ VideoEncoderConfig Copy() const { return VideoEncoderConfig(*this); }
+
+ VideoEncoderConfig();
+ VideoEncoderConfig(VideoEncoderConfig&&);
+ ~VideoEncoderConfig();
+ std::string ToString() const;
+
+ rtc::scoped_refptr<VideoStreamFactoryInterface> video_stream_factory;
+ std::vector<SpatialLayer> spatial_layers;
+ ContentType content_type;
+ rtc::scoped_refptr<const EncoderSpecificSettings> encoder_specific_settings;
+
+ // Padding will be used up to this bitrate regardless of the bitrate produced
+ // by the encoder. Padding above what's actually produced by the encoder helps
+ // maintaining a higher bitrate estimate. Padding will however not be sent
+ // unless the estimated bandwidth indicates that the link can handle it.
+ int min_transmit_bitrate_bps;
+ int max_bitrate_bps;
+
+ // Max number of encoded VideoStreams to produce.
+ size_t number_of_streams;
+
+ private:
+ // Access to the copy constructor is private to force use of the Copy()
+ // method for those exceptional cases where we do use it.
+ VideoEncoderConfig(const VideoEncoderConfig&);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_CALL_VIDEO_CONFIG_H_
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 18e904d..8ed93a1 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -17,9 +17,10 @@
#include <vector>
#include "webrtc/api/call/transport.h"
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/call/rtp_config.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/include/frame_callback.h"
-#include "webrtc/config.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/rtc_base/platform_file.h"
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index a176709..a458203 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -17,9 +17,11 @@
#include <vector>
#include "webrtc/api/call/transport.h"
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/call/rtp_config.h"
+#include "webrtc/call/video_config.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/include/frame_callback.h"
-#include "webrtc/config.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
#include "webrtc/rtc_base/platform_file.h"
diff --git a/config.cc b/config.cc
deleted file mode 100644
index 19a9a96..0000000
--- a/config.cc
+++ /dev/null
@@ -1,283 +0,0 @@
-/*
- * Copyright (c) 2014 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 "webrtc/config.h"
-
-#include <algorithm>
-#include <sstream>
-#include <string>
-
-#include "webrtc/rtc_base/checks.h"
-
-namespace webrtc {
-std::string NackConfig::ToString() const {
- std::stringstream ss;
- ss << "{rtp_history_ms: " << rtp_history_ms;
- ss << '}';
- return ss.str();
-}
-
-std::string UlpfecConfig::ToString() const {
- std::stringstream ss;
- ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
- ss << ", red_payload_type: " << red_payload_type;
- ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
- ss << '}';
- return ss.str();
-}
-
-bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
- return ulpfec_payload_type == other.ulpfec_payload_type &&
- red_payload_type == other.red_payload_type &&
- red_rtx_payload_type == other.red_rtx_payload_type;
-}
-
-std::string RtpExtension::ToString() const {
- std::stringstream ss;
- ss << "{uri: " << uri;
- ss << ", id: " << id;
- if (encrypt) {
- ss << ", encrypt";
- }
- ss << '}';
- return ss.str();
-}
-
-const char RtpExtension::kAudioLevelUri[] =
- "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
-const int RtpExtension::kAudioLevelDefaultId = 1;
-
-const char RtpExtension::kTimestampOffsetUri[] =
- "urn:ietf:params:rtp-hdrext:toffset";
-const int RtpExtension::kTimestampOffsetDefaultId = 2;
-
-const char RtpExtension::kAbsSendTimeUri[] =
- "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
-const int RtpExtension::kAbsSendTimeDefaultId = 3;
-
-const char RtpExtension::kVideoRotationUri[] = "urn:3gpp:video-orientation";
-const int RtpExtension::kVideoRotationDefaultId = 4;
-
-const char RtpExtension::kTransportSequenceNumberUri[] =
- "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
-const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
-
-// This extension allows applications to adaptively limit the playout delay
-// on frames as per the current needs. For example, a gaming application
-// has very different needs on end-to-end delay compared to a video-conference
-// application.
-const char RtpExtension::kPlayoutDelayUri[] =
- "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
-const int RtpExtension::kPlayoutDelayDefaultId = 6;
-
-const char RtpExtension::kVideoContentTypeUri[] =
- "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
-const int RtpExtension::kVideoContentTypeDefaultId = 7;
-
-const char RtpExtension::kVideoTimingUri[] =
- "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
-const int RtpExtension::kVideoTimingDefaultId = 8;
-
-const char RtpExtension::kEncryptHeaderExtensionsUri[] =
- "urn:ietf:params:rtp-hdrext:encrypt";
-
-const int RtpExtension::kMinId = 1;
-const int RtpExtension::kMaxId = 14;
-
-bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
- return uri == webrtc::RtpExtension::kAudioLevelUri ||
- uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
-}
-
-bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
- return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
- uri == webrtc::RtpExtension::kAbsSendTimeUri ||
- uri == webrtc::RtpExtension::kVideoRotationUri ||
- uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
- uri == webrtc::RtpExtension::kPlayoutDelayUri ||
- uri == webrtc::RtpExtension::kVideoContentTypeUri ||
- uri == webrtc::RtpExtension::kVideoTimingUri;
-}
-
-bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
- return uri == webrtc::RtpExtension::kAudioLevelUri ||
- uri == webrtc::RtpExtension::kTimestampOffsetUri ||
-#if !defined(ENABLE_EXTERNAL_AUTH)
- // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri"
- // here and filter out later if external auth is really used in
- // srtpfilter. External auth is used by Chromium and replaces the
- // extension header value of "kAbsSendTimeUri", so it must not be
- // encrypted (which can't be done by Chromium).
- uri == webrtc::RtpExtension::kAbsSendTimeUri ||
-#endif
- uri == webrtc::RtpExtension::kVideoRotationUri ||
- uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
- uri == webrtc::RtpExtension::kPlayoutDelayUri ||
- uri == webrtc::RtpExtension::kVideoContentTypeUri;
-}
-
-const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
- const std::vector<RtpExtension>& extensions,
- const std::string& uri) {
- for (const auto& extension : extensions) {
- if (extension.uri == uri) {
- return &extension;
- }
- }
- return nullptr;
-}
-
-std::vector<RtpExtension> RtpExtension::FilterDuplicateNonEncrypted(
- const std::vector<RtpExtension>& extensions) {
- std::vector<RtpExtension> filtered;
- for (auto extension = extensions.begin(); extension != extensions.end();
- ++extension) {
- if (extension->encrypt) {
- filtered.push_back(*extension);
- continue;
- }
-
- // Only add non-encrypted extension if no encrypted with the same URI
- // is also present...
- if (std::find_if(extension + 1, extensions.end(),
- [extension](const RtpExtension& check) {
- return extension->uri == check.uri;
- }) != extensions.end()) {
- continue;
- }
-
- // ...and has not been added before.
- if (!FindHeaderExtensionByUri(filtered, extension->uri)) {
- filtered.push_back(*extension);
- }
- }
- return filtered;
-}
-
-VideoStream::VideoStream()
- : width(0),
- height(0),
- max_framerate(-1),
- min_bitrate_bps(-1),
- target_bitrate_bps(-1),
- max_bitrate_bps(-1),
- max_qp(-1) {}
-
-VideoStream::~VideoStream() = default;
-
-std::string VideoStream::ToString() const {
- std::stringstream ss;
- ss << "{width: " << width;
- ss << ", height: " << height;
- ss << ", max_framerate: " << max_framerate;
- ss << ", min_bitrate_bps:" << min_bitrate_bps;
- ss << ", target_bitrate_bps:" << target_bitrate_bps;
- ss << ", max_bitrate_bps:" << max_bitrate_bps;
- ss << ", max_qp: " << max_qp;
-
- ss << ", temporal_layer_thresholds_bps: [";
- for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
- ss << temporal_layer_thresholds_bps[i];
- if (i != temporal_layer_thresholds_bps.size() - 1)
- ss << ", ";
- }
- ss << ']';
-
- ss << '}';
- return ss.str();
-}
-
-VideoEncoderConfig::VideoEncoderConfig()
- : content_type(ContentType::kRealtimeVideo),
- encoder_specific_settings(nullptr),
- min_transmit_bitrate_bps(0),
- max_bitrate_bps(0),
- number_of_streams(0) {}
-
-VideoEncoderConfig::VideoEncoderConfig(VideoEncoderConfig&&) = default;
-
-VideoEncoderConfig::~VideoEncoderConfig() = default;
-
-std::string VideoEncoderConfig::ToString() const {
- std::stringstream ss;
- ss << "{content_type: ";
- switch (content_type) {
- case ContentType::kRealtimeVideo:
- ss << "kRealtimeVideo";
- break;
- case ContentType::kScreen:
- ss << "kScreenshare";
- break;
- }
- ss << ", encoder_specific_settings: ";
- ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
-
- ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
- ss << '}';
- return ss.str();
-}
-
-VideoEncoderConfig::VideoEncoderConfig(const VideoEncoderConfig&) = default;
-
-void VideoEncoderConfig::EncoderSpecificSettings::FillEncoderSpecificSettings(
- VideoCodec* codec) const {
- if (codec->codecType == kVideoCodecH264) {
- FillVideoCodecH264(codec->H264());
- } else if (codec->codecType == kVideoCodecVP8) {
- FillVideoCodecVp8(codec->VP8());
- } else if (codec->codecType == kVideoCodecVP9) {
- FillVideoCodecVp9(codec->VP9());
- } else {
- RTC_NOTREACHED() << "Encoder specifics set/used for unknown codec type.";
- }
-}
-
-void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecH264(
- VideoCodecH264* h264_settings) const {
- RTC_NOTREACHED();
-}
-
-void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecVp8(
- VideoCodecVP8* vp8_settings) const {
- RTC_NOTREACHED();
-}
-
-void VideoEncoderConfig::EncoderSpecificSettings::FillVideoCodecVp9(
- VideoCodecVP9* vp9_settings) const {
- RTC_NOTREACHED();
-}
-
-VideoEncoderConfig::H264EncoderSpecificSettings::H264EncoderSpecificSettings(
- const VideoCodecH264& specifics)
- : specifics_(specifics) {}
-
-void VideoEncoderConfig::H264EncoderSpecificSettings::FillVideoCodecH264(
- VideoCodecH264* h264_settings) const {
- *h264_settings = specifics_;
-}
-
-VideoEncoderConfig::Vp8EncoderSpecificSettings::Vp8EncoderSpecificSettings(
- const VideoCodecVP8& specifics)
- : specifics_(specifics) {}
-
-void VideoEncoderConfig::Vp8EncoderSpecificSettings::FillVideoCodecVp8(
- VideoCodecVP8* vp8_settings) const {
- *vp8_settings = specifics_;
-}
-
-VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings(
- const VideoCodecVP9& specifics)
- : specifics_(specifics) {}
-
-void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
- VideoCodecVP9* vp9_settings) const {
- *vp9_settings = specifics_;
-}
-
-} // namespace webrtc
diff --git a/config.h b/config.h
index 962e0f2..46c1f42 100644
--- a/config.h
+++ b/config.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ * Copyright (c) 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
@@ -8,259 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-// TODO(pbos): Move Config from common.h to here.
-
#ifndef WEBRTC_CONFIG_H_
#define WEBRTC_CONFIG_H_
-#include <string>
-#include <vector>
+// TODO(holmer): Delete this file once downstream projects have been updated.
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/optional.h"
-#include "webrtc/rtc_base/refcount.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// Settings for NACK, see RFC 4585 for details.
-struct NackConfig {
- NackConfig() : rtp_history_ms(0) {}
- std::string ToString() const;
- // Send side: the time RTP packets are stored for retransmissions.
- // Receive side: the time the receiver is prepared to wait for
- // retransmissions.
- // Set to '0' to disable.
- int rtp_history_ms;
-};
-
-// Settings for ULPFEC forward error correction.
-// Set the payload types to '-1' to disable.
-struct UlpfecConfig {
- UlpfecConfig()
- : ulpfec_payload_type(-1),
- red_payload_type(-1),
- red_rtx_payload_type(-1) {}
- std::string ToString() const;
- bool operator==(const UlpfecConfig& other) const;
-
- // Payload type used for ULPFEC packets.
- int ulpfec_payload_type;
-
- // Payload type used for RED packets.
- int red_payload_type;
-
- // RTX payload type for RED payload.
- int red_rtx_payload_type;
-};
-
-// RTP header extension, see RFC 5285.
-struct RtpExtension {
- RtpExtension() {}
- RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
- RtpExtension(const std::string& uri, int id, bool encrypt) : uri(uri),
- id(id), encrypt(encrypt) {}
- std::string ToString() const;
- bool operator==(const RtpExtension& rhs) const {
- return uri == rhs.uri && id == rhs.id && encrypt == rhs.encrypt;
- }
- static bool IsSupportedForAudio(const std::string& uri);
- static bool IsSupportedForVideo(const std::string& uri);
- // Return "true" if the given RTP header extension URI may be encrypted.
- static bool IsEncryptionSupported(const std::string& uri);
-
- // Returns the named header extension if found among all extensions,
- // nullptr otherwise.
- static const RtpExtension* FindHeaderExtensionByUri(
- const std::vector<RtpExtension>& extensions,
- const std::string& uri);
-
- // Return a list of RTP header extensions with the non-encrypted extensions
- // removed if both the encrypted and non-encrypted extension is present for
- // the same URI.
- static std::vector<RtpExtension> FilterDuplicateNonEncrypted(
- const std::vector<RtpExtension>& extensions);
-
- // Header extension for audio levels, as defined in:
- // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
- static const char kAudioLevelUri[];
- static const int kAudioLevelDefaultId;
-
- // Header extension for RTP timestamp offset, see RFC 5450 for details:
- // http://tools.ietf.org/html/rfc5450
- static const char kTimestampOffsetUri[];
- static const int kTimestampOffsetDefaultId;
-
- // Header extension for absolute send time, see url for details:
- // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
- static const char kAbsSendTimeUri[];
- static const int kAbsSendTimeDefaultId;
-
- // Header extension for coordination of video orientation, see url for
- // details:
- // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf
- static const char kVideoRotationUri[];
- static const int kVideoRotationDefaultId;
-
- // Header extension for video content type. E.g. default or screenshare.
- static const char kVideoContentTypeUri[];
- static const int kVideoContentTypeDefaultId;
-
- // Header extension for video timing.
- static const char kVideoTimingUri[];
- static const int kVideoTimingDefaultId;
-
- // Header extension for transport sequence number, see url for details:
- // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
- static const char kTransportSequenceNumberUri[];
- static const int kTransportSequenceNumberDefaultId;
-
- static const char kPlayoutDelayUri[];
- static const int kPlayoutDelayDefaultId;
-
- // Encryption of Header Extensions, see RFC 6904 for details:
- // https://tools.ietf.org/html/rfc6904
- static const char kEncryptHeaderExtensionsUri[];
-
- // Inclusive min and max IDs for one-byte header extensions, per RFC5285.
- static const int kMinId;
- static const int kMaxId;
-
- std::string uri;
- int id = 0;
- bool encrypt = false;
-};
-
-struct VideoStream {
- VideoStream();
- ~VideoStream();
- std::string ToString() const;
-
- size_t width;
- size_t height;
- int max_framerate;
-
- int min_bitrate_bps;
- int target_bitrate_bps;
- int max_bitrate_bps;
-
- int max_qp;
-
- // Bitrate thresholds for enabling additional temporal layers. Since these are
- // thresholds in between layers, we have one additional layer. One threshold
- // gives two temporal layers, one below the threshold and one above, two give
- // three, and so on.
- // The VideoEncoder may redistribute bitrates over the temporal layers so a
- // bitrate threshold of 100k and an estimate of 105k does not imply that we
- // get 100k in one temporal layer and 5k in the other, just that the bitrate
- // in the first temporal layer should not exceed 100k.
- // TODO(kthelgason): Apart from a special case for two-layer screencast these
- // thresholds are not propagated to the VideoEncoder. To be implemented.
- std::vector<int> temporal_layer_thresholds_bps;
-};
-
-class VideoEncoderConfig {
- public:
- // These are reference counted to permit copying VideoEncoderConfig and be
- // kept alive until all encoder_specific_settings go out of scope.
- // TODO(kthelgason): Consider removing the need for copying VideoEncoderConfig
- // and use rtc::Optional for encoder_specific_settings instead.
- class EncoderSpecificSettings : public rtc::RefCountInterface {
- public:
- // TODO(pbos): Remove FillEncoderSpecificSettings as soon as VideoCodec is
- // not in use and encoder implementations ask for codec-specific structs
- // directly.
- void FillEncoderSpecificSettings(VideoCodec* codec_struct) const;
-
- virtual void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const;
- virtual void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const;
- virtual void FillVideoCodecH264(VideoCodecH264* h264_settings) const;
- private:
- ~EncoderSpecificSettings() override {}
- friend class VideoEncoderConfig;
- };
-
- class H264EncoderSpecificSettings : public EncoderSpecificSettings {
- public:
- explicit H264EncoderSpecificSettings(const VideoCodecH264& specifics);
- void FillVideoCodecH264(VideoCodecH264* h264_settings) const override;
-
- private:
- VideoCodecH264 specifics_;
- };
-
- class Vp8EncoderSpecificSettings : public EncoderSpecificSettings {
- public:
- explicit Vp8EncoderSpecificSettings(const VideoCodecVP8& specifics);
- void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const override;
-
- private:
- VideoCodecVP8 specifics_;
- };
-
- class Vp9EncoderSpecificSettings : public EncoderSpecificSettings {
- public:
- explicit Vp9EncoderSpecificSettings(const VideoCodecVP9& specifics);
- void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const override;
-
- private:
- VideoCodecVP9 specifics_;
- };
-
- enum class ContentType {
- kRealtimeVideo,
- kScreen,
- };
-
- class VideoStreamFactoryInterface : public rtc::RefCountInterface {
- public:
- // An implementation should return a std::vector<VideoStream> with the
- // wanted VideoStream settings for the given video resolution.
- // The size of the vector may not be larger than
- // |encoder_config.number_of_streams|.
- virtual std::vector<VideoStream> CreateEncoderStreams(
- int width,
- int height,
- const VideoEncoderConfig& encoder_config) = 0;
-
- protected:
- ~VideoStreamFactoryInterface() override {}
- };
-
- VideoEncoderConfig& operator=(VideoEncoderConfig&&) = default;
- VideoEncoderConfig& operator=(const VideoEncoderConfig&) = delete;
-
- // Mostly used by tests. Avoid creating copies if you can.
- VideoEncoderConfig Copy() const { return VideoEncoderConfig(*this); }
-
- VideoEncoderConfig();
- VideoEncoderConfig(VideoEncoderConfig&&);
- ~VideoEncoderConfig();
- std::string ToString() const;
-
- rtc::scoped_refptr<VideoStreamFactoryInterface> video_stream_factory;
- std::vector<SpatialLayer> spatial_layers;
- ContentType content_type;
- rtc::scoped_refptr<const EncoderSpecificSettings> encoder_specific_settings;
-
- // Padding will be used up to this bitrate regardless of the bitrate produced
- // by the encoder. Padding above what's actually produced by the encoder helps
- // maintaining a higher bitrate estimate. Padding will however not be sent
- // unless the estimated bandwidth indicates that the link can handle it.
- int min_transmit_bitrate_bps;
- int max_bitrate_bps;
-
- // Max number of encoded VideoStreams to produce.
- size_t number_of_streams;
-
- private:
- // Access to the copy constructor is private to force use of the Copy()
- // method for those exceptional cases where we do use it.
- VideoEncoderConfig(const VideoEncoderConfig&);
-};
-
-} // namespace webrtc
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/call/rtp_config.h"
#endif // WEBRTC_CONFIG_H_
diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index 8f1e493..6b1e549 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -29,6 +29,7 @@
]
deps = [
"..:webrtc_common",
+ "../api:libjingle_peerconnection_api",
"../call:video_stream_api",
"../rtc_base:rtc_base_approved",
]
@@ -159,6 +160,7 @@
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_parser",
+ "../call:video_stream_api",
"../rtc_base:rtc_base_approved",
# TODO(kwiberg): Remove this dependency.
diff --git a/logging/rtc_event_log/rtc_event_log.h b/logging/rtc_event_log/rtc_event_log.h
index dc33511..d26dccb 100644
--- a/logging/rtc_event_log/rtc_event_log.h
+++ b/logging/rtc_event_log/rtc_event_log.h
@@ -15,7 +15,8 @@
#include <string>
#include <vector>
-#include "webrtc/config.h"
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/common_types.h"
#include "webrtc/rtc_base/platform_file.h"
namespace webrtc {
diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index 3f03108..bba6ace 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -16,8 +16,8 @@
#include <string>
#include <utility> // pair
+#include "webrtc/call/video_config.h"
#include "webrtc/common_types.h"
-#include "webrtc/config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 99ed291..1ab4123 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -18,7 +18,7 @@
#include "webrtc/api/rtpparameters.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/api/video/video_timing.h"
-#include "webrtc/config.h"
+#include "webrtc/call/video_config.h"
#include "webrtc/media/base/codec.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/streamparams.h"
diff --git a/media/engine/simulcast.h b/media/engine/simulcast.h
index 4531bcc..81668e4 100644
--- a/media/engine/simulcast.h
+++ b/media/engine/simulcast.h
@@ -13,7 +13,7 @@
#include <vector>
-#include "webrtc/config.h"
+#include "webrtc/call/video_config.h"
#include "webrtc/rtc_base/basictypes.h"
namespace cricket {
diff --git a/media/engine/webrtcmediaengine.h b/media/engine/webrtcmediaengine.h
index edbda66..8740917 100644
--- a/media/engine/webrtcmediaengine.h
+++ b/media/engine/webrtcmediaengine.h
@@ -15,7 +15,6 @@
#include <vector>
#include "webrtc/call/call.h"
-#include "webrtc/config.h"
#include "webrtc/media/base/mediaengine.h"
namespace webrtc {
diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h
index f938446..a42459a 100644
--- a/media/engine/webrtcvoiceengine.h
+++ b/media/engine/webrtcvoiceengine.h
@@ -20,7 +20,6 @@
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/call/audio_state.h"
#include "webrtc/call/call.h"
-#include "webrtc/config.h"
#include "webrtc/media/base/rtputils.h"
#include "webrtc/media/engine/apm_helpers.h"
#include "webrtc/media/engine/webrtccommon.h"
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 50819f1..569d214 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1569,8 +1569,11 @@
":isac_fix",
":webrtc_opus",
"../..:webrtc_common",
+ "../../api:libjingle_peerconnection_api",
"../../rtc_base:rtc_base_approved",
+ "../../system_wrappers:metrics_default",
"../../system_wrappers:system_wrappers_default",
+ "../../test:field_trial",
"../../test:test_main",
"../audio_processing",
"//testing/gtest",
@@ -1702,6 +1705,8 @@
"../..:webrtc_common",
"../../common_audio",
"../../rtc_base:rtc_base_approved",
+ "../../system_wrappers:metrics_default",
+ "../../test:field_trial",
]
configs += [ ":RTPencode_config" ]
diff --git a/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index db2aacb..f8707c3 100644
--- a/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -14,7 +14,6 @@
#include <memory>
#include <vector>
-#include "webrtc/config.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/array_view.h"
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 75e5aab..e49b632 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -10,7 +10,6 @@
#include "webrtc/modules/audio_processing/audio_processing_impl.h"
-#include "webrtc/config.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/gmock.h"
diff --git a/modules/audio_processing/audio_processing_performance_unittest.cc b/modules/audio_processing/audio_processing_performance_unittest.cc
index 6d0a61b..e478c72 100644
--- a/modules/audio_processing/audio_processing_performance_unittest.cc
+++ b/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -15,7 +15,6 @@
#include <memory>
#include <vector>
-#include "webrtc/config.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/array_view.h"
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 775ebcb..07ac7ae 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -341,6 +341,7 @@
":rtp_rtcp",
"..:module_api",
"../..:webrtc_common",
+ "../../api:libjingle_peerconnection_api",
"../../api:transport_api",
"../../common_video:common_video",
"../../rtc_base:rtc_base_approved",
diff --git a/modules/rtp_rtcp/include/flexfec_sender.h b/modules/rtp_rtcp/include/flexfec_sender.h
index 3bcd2eb..82a01cc 100644
--- a/modules/rtp_rtcp/include/flexfec_sender.h
+++ b/modules/rtp_rtcp/include/flexfec_sender.h
@@ -14,7 +14,7 @@
#include <memory>
#include <vector>
-#include "webrtc/config.h"
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
diff --git a/modules/rtp_rtcp/include/rtp_header_extension_map.h b/modules/rtp_rtcp/include/rtp_header_extension_map.h
index 98bc9ec..93cbd10 100644
--- a/modules/rtp_rtcp/include/rtp_header_extension_map.h
+++ b/modules/rtp_rtcp/include/rtp_header_extension_map.h
@@ -13,7 +13,7 @@
#include <string>
-#include "webrtc/config.h"
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/rtc_base/array_view.h"
#include "webrtc/rtc_base/basictypes.h"
diff --git a/modules/rtp_rtcp/source/flexfec_sender_unittest.cc b/modules/rtp_rtcp/source/flexfec_sender_unittest.cc
index 7c26b80..da33837 100644
--- a/modules/rtp_rtcp/source/flexfec_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/flexfec_sender_unittest.cc
@@ -10,7 +10,7 @@
#include <vector>
-#include "webrtc/config.h"
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 790319a..31d1b0e 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -16,8 +16,8 @@
#include <set>
#include <string>
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/common_types.h"
-#include "webrtc/config.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
diff --git a/pc/webrtcsdp.cc b/pc/webrtcsdp.cc
index e514fe7..7dc2605 100644
--- a/pc/webrtcsdp.cc
+++ b/pc/webrtcsdp.cc
@@ -22,13 +22,8 @@
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/messagedigest.h"
-#include "webrtc/rtc_base/stringutils.h"
// for RtpExtension
-#include "webrtc/config.h"
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/media/base/codec.h"
#include "webrtc/media/base/cryptoparams.h"
#include "webrtc/media/base/mediaconstants.h"
@@ -38,6 +33,11 @@
#include "webrtc/p2p/base/p2pconstants.h"
#include "webrtc/p2p/base/port.h"
#include "webrtc/pc/mediasession.h"
+#include "webrtc/rtc_base/arraysize.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/messagedigest.h"
+#include "webrtc/rtc_base/stringutils.h"
using cricket::AudioContentDescription;
using cricket::Candidate;
diff --git a/test/call_test.cc b/test/call_test.cc
index 3efc022..b5d7236 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -15,7 +15,7 @@
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
#include "webrtc/call/rtp_transport_controller_send.h"
-#include "webrtc/config.h"
+#include "webrtc/call/video_config.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/event.h"
diff --git a/video/payload_router.h b/video/payload_router.h
index 8b51861..bfd4b8e 100644
--- a/video/payload_router.h
+++ b/video/payload_router.h
@@ -15,7 +15,6 @@
#include "webrtc/api/video_codecs/video_encoder.h"
#include "webrtc/common_types.h"
-#include "webrtc/config.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/thread_annotations.h"
diff --git a/video/payload_router_unittest.cc b/video/payload_router_unittest.cc
index 9f58ceb..2293a24 100644
--- a/video/payload_router_unittest.cc
+++ b/video/payload_router_unittest.cc
@@ -10,6 +10,7 @@
#include <memory>
+#include "webrtc/call/video_config.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 4678b8f..0dbb2f7 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -14,8 +14,8 @@
#include <utility>
#include <vector>
+#include "webrtc/call/video_config.h"
#include "webrtc/common_types.h"
-#include "webrtc/config.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 901d80c..039d935 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -15,7 +15,6 @@
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/call/rtp_transport_controller_send_interface.h"
-#include "webrtc/config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
