8d609f6b6d
Reason for revert: Re-land, reverting did not fix bug. https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) > > Reason for revert: > Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see > > https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 > > Original issue's description: > > Added the GetSources() to the RtpReceiverInterface and implemented > > it for the AudioRtpReceiver. > > > > This method returns a vector of RtpSource(both CSRC source and SSRC > > source) which contains the ID of a source, the timestamp, the source > > type (SSRC or CSRC) and the audio level. > > > > The RtpSource objects are buffered and maintained by the > > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > > the info of the contributing source will be pulled along the object > > chain: > > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > > > BUG=chromium:703122 > > TBR=stefan@webrtc.org, danilchap@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2770233003 > > Cr-Commit-Position: refs/heads/master@{#17591} > > Committed:292084c376> > TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=chromium:703122 > > Review-Url: https://codereview.webrtc.org/2809613002 > Cr-Commit-Position: refs/heads/master@{#17616} > Committed:fbcc5cb386TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2810623003 Cr-Commit-Position: refs/heads/master@{#17621}
137 lines
3.7 KiB
Plaintext
137 lines
3.7 KiB
Plaintext
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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rtc_source_set("call_interfaces") {
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sources = [
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"audio_receive_stream.h",
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"audio_send_stream.cc",
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"audio_send_stream.h",
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"audio_state.h",
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"call.h",
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"flexfec_receive_stream.h",
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"rtp_transport_controller_send.h",
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"syncable.cc",
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"syncable.h",
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]
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deps = [
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"..:webrtc_common",
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"../api:audio_mixer_api",
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"../api:libjingle_peerconnection_api",
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"../api:transport_api",
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"../api/audio_codecs:audio_codecs_api",
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"../base:rtc_base",
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"../base:rtc_base_approved",
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"../modules/audio_coding:audio_encoder_interface",
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]
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}
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rtc_static_library("call") {
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sources = [
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"bitrate_allocator.cc",
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"call.cc",
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"flexfec_receive_stream_impl.cc",
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"flexfec_receive_stream_impl.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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public_deps = [
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":call_interfaces",
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"../api:call_api",
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]
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deps = [
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":call_interfaces",
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"..:webrtc_common",
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"../api:transport_api",
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"../audio",
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"../base:rtc_task_queue",
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_log_impl",
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"../modules/bitrate_controller",
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"../modules/congestion_controller",
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../modules/utility",
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"../system_wrappers",
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"../video",
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]
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}
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if (rtc_include_tests) {
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rtc_source_set("call_tests") {
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testonly = true
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sources = [
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"bitrate_allocator_unittest.cc",
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"bitrate_estimator_tests.cc",
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"call_unittest.cc",
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"flexfec_receive_stream_unittest.cc",
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]
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deps = [
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":call",
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"../base:rtc_base_approved",
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"../logging:rtc_event_log_api",
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer",
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"../modules/bitrate_controller",
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../system_wrappers",
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"../test:audio_codec_mocks",
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"../test:direct_transport",
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"../test:test_common",
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"../test:test_support",
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"../test:video_test_common",
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"//testing/gmock",
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"//testing/gtest",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("call_perf_tests") {
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testonly = true
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sources = [
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"call_perf_tests.cc",
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"rampup_tests.cc",
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"rampup_tests.h",
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]
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deps = [
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":call_interfaces",
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"..:webrtc_common",
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"../base:rtc_base_approved",
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"../logging:rtc_event_log_api",
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"../modules/audio_coding",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/rtp_rtcp",
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"../system_wrappers",
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"../system_wrappers:metrics_default",
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"../test:direct_transport",
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"../test:fake_audio_device",
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"../test:test_support",
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"../test:video_test_common",
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"../video",
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"../voice_engine",
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"//testing/gtest",
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"//webrtc/test:field_trial",
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"//webrtc/test:test_common",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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}
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