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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
102 lines
3.6 KiB
C++
102 lines
3.6 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 AUDIO_AUDIO_RECEIVE_STREAM_H_
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#define AUDIO_AUDIO_RECEIVE_STREAM_H_
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#include <memory>
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "audio/audio_state.h"
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#include "call/audio_receive_stream.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "call/syncable.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/thread_checker.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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class RtpStreamReceiverControllerInterface;
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class RtpStreamReceiverInterface;
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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(RtpStreamReceiverControllerInterface* receiver_controller,
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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): We don't formally implement RtpPacketSinkInterface, and this
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// method shouldn't be needed. But it's currently used by the
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// AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test
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// shuld be refactored or deleted, and then delete this method.
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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_ RTC_ACCESS_ON(worker_thread_checker_) = false;
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std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
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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 // AUDIO_AUDIO_RECEIVE_STREAM_H_
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