| /* |
| * 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 <map> |
| #include <memory> |
| #include <vector> |
| |
| #include "api/audio/audio_mixer.h" |
| #include "api/neteq/neteq_factory.h" |
| #include "api/rtp_headers.h" |
| #include "api/sequence_checker.h" |
| #include "audio/audio_state.h" |
| #include "call/audio_receive_stream.h" |
| #include "call/syncable.h" |
| #include "modules/rtp_rtcp/source/source_tracker.h" |
| #include "rtc_base/system/no_unique_address.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, |
| PacketRouter* packet_router, |
| ProcessThread* module_process_thread, |
| NetEqFactory* neteq_factory, |
| 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, |
| 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() = delete; |
| AudioReceiveStream(const AudioReceiveStream&) = delete; |
| AudioReceiveStream& operator=(const AudioReceiveStream&) = delete; |
| |
| // Destruction happens on the worker thread. Prior to destruction the caller |
| // must ensure that a registration with the transport has been cleared. See |
| // `RegisterWithTransport` for details. |
| // TODO(tommi): As a further improvement to this, performing the full |
| // destruction on the network thread could be made the default. |
| ~AudioReceiveStream() override; |
| |
| // Called on the network thread to register/unregister with the network |
| // transport. |
| void RegisterWithTransport( |
| RtpStreamReceiverControllerInterface* receiver_controller); |
| // If registration has previously been done (via `RegisterWithTransport`) then |
| // `UnregisterFromTransport` must be called prior to destruction, on the |
| // network thread. |
| void UnregisterFromTransport(); |
| |
| // webrtc::AudioReceiveStream implementation. |
| void Start() override; |
| void Stop() override; |
| bool IsRunning() const override; |
| void SetDepacketizerToDecoderFrameTransformer( |
| rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) |
| override; |
| void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override; |
| void SetUseTransportCcAndNackHistory(bool use_transport_cc, |
| int history_ms) override; |
| |
| webrtc::AudioReceiveStream::Stats GetStats( |
| bool get_and_clear_legacy_stats) 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; |
| |
| // AudioMixer::Source |
| AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, |
| AudioFrame* audio_frame) override; |
| int Ssrc() const override; |
| int PreferredSampleRate() const override; |
| |
| // Syncable |
| uint32_t id() const override; |
| absl::optional<Syncable::Info> GetInfo() const override; |
| bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp, |
| int64_t* time_ms) const override; |
| void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms, |
| int64_t time_ms) override; |
| bool SetMinimumPlayoutDelay(int delay_ms) override; |
| |
| void AssociateSendStream(AudioSendStream* send_stream); |
| void DeliverRtcp(const uint8_t* packet, size_t length); |
| |
| uint32_t local_ssrc() const { |
| // The local_ssrc member variable of config_ will never change and can be |
| // considered const. |
| return config_.rtp.local_ssrc; |
| } |
| |
| uint32_t remote_ssrc() const { |
| // The remote_ssrc member variable of config_ will never change and can be |
| // considered const. |
| return config_.rtp.remote_ssrc; |
| } |
| |
| const webrtc::AudioReceiveStream::Config& config() const; |
| const AudioSendStream* GetAssociatedSendStreamForTesting() const; |
| |
| // TODO(tommi): Remove this method. |
| void ReconfigureForTesting(const webrtc::AudioReceiveStream::Config& config); |
| |
| private: |
| AudioState* audio_state() const; |
| |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_; |
| // TODO(bugs.webrtc.org/11993): This checker conceptually represents |
| // operations that belong to the network thread. The Call class is currently |
| // moving towards handling network packets on the network thread and while |
| // that work is ongoing, this checker may in practice represent the worker |
| // thread, but still serves as a mechanism of grouping together concepts |
| // that belong to the network thread. Once the packets are fully delivered |
| // on the network thread, this comment will be deleted. |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_; |
| webrtc::AudioReceiveStream::Config config_; |
| rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
| SourceTracker source_tracker_; |
| const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_; |
| AudioSendStream* associated_send_stream_ |
| RTC_GUARDED_BY(packet_sequence_checker_) = nullptr; |
| |
| bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false; |
| |
| std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| }; |
| } // namespace internal |
| } // namespace webrtc |
| |
| #endif // AUDIO_AUDIO_RECEIVE_STREAM_H_ |