| /* |
| * Copyright (c) 2015 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 AUDIO_AUDIO_RECEIVE_STREAM_H_ |
| #define AUDIO_AUDIO_RECEIVE_STREAM_H_ |
| |
| #include <memory> |
| #include <vector> |
| |
| #include "api/audio/audio_mixer.h" |
| #include "api/rtp_headers.h" |
| #include "audio/audio_state.h" |
| #include "call/audio_receive_stream.h" |
| #include "call/syncable.h" |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/thread_checker.h" |
| #include "system_wrappers/include/clock.h" |
| |
| namespace webrtc { |
| class PacketRouter; |
| class ProcessThread; |
| class RtcEventLog; |
| class RtpPacketReceived; |
| class RtpStreamReceiverControllerInterface; |
| class RtpStreamReceiverInterface; |
| |
| namespace voe { |
| class ChannelReceiveInterface; |
| } // namespace voe |
| |
| namespace internal { |
| class AudioSendStream; |
| |
| class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
| public AudioMixer::Source, |
| public Syncable { |
| public: |
| AudioReceiveStream(Clock* clock, |
| RtpStreamReceiverControllerInterface* receiver_controller, |
| PacketRouter* packet_router, |
| ProcessThread* module_process_thread, |
| const webrtc::AudioReceiveStream::Config& config, |
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| webrtc::RtcEventLog* event_log); |
| // For unit tests, which need to supply a mock channel receive. |
| AudioReceiveStream( |
| Clock* clock, |
| RtpStreamReceiverControllerInterface* receiver_controller, |
| PacketRouter* packet_router, |
| const webrtc::AudioReceiveStream::Config& config, |
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| webrtc::RtcEventLog* event_log, |
| std::unique_ptr<voe::ChannelReceiveInterface> channel_receive); |
| ~AudioReceiveStream() override; |
| |
| // webrtc::AudioReceiveStream implementation. |
| void Reconfigure(const webrtc::AudioReceiveStream::Config& config) override; |
| void Start() override; |
| void Stop() override; |
| webrtc::AudioReceiveStream::Stats GetStats() const override; |
| void SetSink(AudioSinkInterface* sink) override; |
| void SetGain(float gain) override; |
| bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override; |
| int GetBaseMinimumPlayoutDelayMs() const override; |
| std::vector<webrtc::RtpSource> GetSources() const override; |
| |
| // TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this |
| // method shouldn't be needed. But it's currently used by the |
| // AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test |
| // shuld be refactored or deleted, and then delete this method. |
| void OnRtpPacket(const RtpPacketReceived& packet); |
| |
| // AudioMixer::Source |
| AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, |
| AudioFrame* audio_frame) override; |
| int Ssrc() const override; |
| int PreferredSampleRate() const override; |
| |
| // Syncable |
| int id() const override; |
| absl::optional<Syncable::Info> GetInfo() const override; |
| uint32_t GetPlayoutTimestamp() const override; |
| void SetMinimumPlayoutDelay(int delay_ms) override; |
| |
| void AssociateSendStream(AudioSendStream* send_stream); |
| void SignalNetworkState(NetworkState state); |
| bool DeliverRtcp(const uint8_t* packet, size_t length); |
| const webrtc::AudioReceiveStream::Config& config() const; |
| const AudioSendStream* GetAssociatedSendStreamForTesting() const; |
| |
| private: |
| static void ConfigureStream(AudioReceiveStream* stream, |
| const Config& new_config, |
| bool first_time); |
| |
| AudioState* audio_state() const; |
| |
| rtc::ThreadChecker worker_thread_checker_; |
| rtc::ThreadChecker module_process_thread_checker_; |
| webrtc::AudioReceiveStream::Config config_; |
| rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
| const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_; |
| AudioSendStream* associated_send_stream_ = nullptr; |
| |
| bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false; |
| |
| std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_; |
| |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); |
| }; |
| } // namespace internal |
| } // namespace webrtc |
| |
| #endif // AUDIO_AUDIO_RECEIVE_STREAM_H_ |