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}
94 lines
3.1 KiB
C++
94 lines
3.1 KiB
C++
/*
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* 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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*/
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#ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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#include <memory>
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#include <vector>
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#include "webrtc/api/audio/audio_mixer.h"
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#include "webrtc/audio/audio_state.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/call/audio_receive_stream.h"
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#include "webrtc/call/syncable.h"
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namespace webrtc {
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class PacketRouter;
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class RtcEventLog;
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class RtpPacketReceived;
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namespace voe {
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class ChannelProxy;
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} // namespace voe
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namespace internal {
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class AudioSendStream;
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class AudioReceiveStream final : public webrtc::AudioReceiveStream,
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public AudioMixer::Source,
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public Syncable {
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public:
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AudioReceiveStream(PacketRouter* packet_router,
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const webrtc::AudioReceiveStream::Config& config,
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const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
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webrtc::RtcEventLog* event_log);
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~AudioReceiveStream() override;
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// webrtc::AudioReceiveStream implementation.
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void Start() override;
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void Stop() override;
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webrtc::AudioReceiveStream::Stats GetStats() const override;
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int GetOutputLevel() const override;
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void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
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void SetGain(float gain) override;
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std::vector<webrtc::RtpSource> GetSources() const override;
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// TODO(nisse): Intended to be part of an RtpPacketReceiver interface.
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void OnRtpPacket(const RtpPacketReceived& packet);
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// AudioMixer::Source
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AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
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AudioFrame* audio_frame) override;
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int Ssrc() const override;
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int PreferredSampleRate() const override;
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// Syncable
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int id() const override;
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rtc::Optional<Syncable::Info> GetInfo() const override;
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uint32_t GetPlayoutTimestamp() const override;
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void SetMinimumPlayoutDelay(int delay_ms) override;
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void AssociateSendStream(AudioSendStream* send_stream);
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void SignalNetworkState(NetworkState state);
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bool DeliverRtcp(const uint8_t* packet, size_t length);
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const webrtc::AudioReceiveStream::Config& config() const;
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private:
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VoiceEngine* voice_engine() const;
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AudioState* audio_state() const;
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int SetVoiceEnginePlayout(bool playout);
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rtc::ThreadChecker worker_thread_checker_;
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rtc::ThreadChecker module_process_thread_checker_;
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const webrtc::AudioReceiveStream::Config config_;
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rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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std::unique_ptr<voe::ChannelProxy> channel_proxy_;
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bool playing_ ACCESS_ON(worker_thread_checker_) = false;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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