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Diffstat (limited to 'third_party/libwebrtc/audio/audio_receive_stream.h')
| -rw-r--r-- | third_party/libwebrtc/audio/audio_receive_stream.h | 174 |
1 files changed, 174 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/audio_receive_stream.h b/third_party/libwebrtc/audio/audio_receive_stream.h new file mode 100644 index 0000000000..51514fb6ce --- /dev/null +++ b/third_party/libwebrtc/audio/audio_receive_stream.h @@ -0,0 +1,174 @@ +/* + * 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 <string> +#include <vector> + +#include "absl/strings/string_view.h" +#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 RtcEventLog; +class RtpStreamReceiverControllerInterface; +class RtpStreamReceiverInterface; + +namespace voe { +class ChannelReceiveInterface; +} // namespace voe + +namespace internal { +class AudioSendStream; +} // namespace internal + +class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface, + public AudioMixer::Source, + public Syncable { + public: + AudioReceiveStreamImpl( + Clock* clock, + PacketRouter* packet_router, + NetEqFactory* neteq_factory, + const webrtc::AudioReceiveStreamInterface::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. + AudioReceiveStreamImpl( + Clock* clock, + PacketRouter* packet_router, + const webrtc::AudioReceiveStreamInterface::Config& config, + const rtc::scoped_refptr<webrtc::AudioState>& audio_state, + webrtc::RtcEventLog* event_log, + std::unique_ptr<voe::ChannelReceiveInterface> channel_receive); + + AudioReceiveStreamImpl() = delete; + AudioReceiveStreamImpl(const AudioReceiveStreamImpl&) = delete; + AudioReceiveStreamImpl& operator=(const AudioReceiveStreamImpl&) = 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. + ~AudioReceiveStreamImpl() 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::AudioReceiveStreamInterface 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 SetNackHistory(int history_ms) override; + void SetNonSenderRttMeasurement(bool enabled) override; + void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface> + frame_decryptor) override; + void SetRtpExtensions(std::vector<RtpExtension> extensions) override; + RtpHeaderExtensionMap GetRtpExtensionMap() const override; + + webrtc::AudioReceiveStreamInterface::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(internal::AudioSendStream* send_stream); + void DeliverRtcp(const uint8_t* packet, size_t length); + + void SetSyncGroup(absl::string_view sync_group); + + void SetLocalSsrc(uint32_t local_ssrc); + + uint32_t local_ssrc() const; + + uint32_t remote_ssrc() const override { + // The remote_ssrc member variable of config_ will never change and can be + // considered const. + return config_.rtp.remote_ssrc; + } + + // Returns a reference to the currently set sync group of the stream. + // Must be called on the packet delivery thread. + const std::string& sync_group() const; + + const AudioSendStream* GetAssociatedSendStreamForTesting() const; + + // TODO(tommi): Remove this method. + void ReconfigureForTesting( + const webrtc::AudioReceiveStreamInterface::Config& config); + + private: + internal::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::AudioReceiveStreamInterface::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 webrtc + +#endif // AUDIO_AUDIO_RECEIVE_STREAM_H_ |