blob: 61ebc2719f589200de43213527a437c70b8b12db [file] [log] [blame]
* 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.
#include <map>
#include <memory>
#include <string>
#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 {
AudioReceiveStream(Clock* clock,
PacketRouter* packet_router,
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.
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;
const RtpConfig& rtp_config() const override { return config_.rtp; }
bool IsRunning() const override;
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override;
void SetUseTransportCcAndNackHistory(bool use_transport_cc,
int history_ms) override;
void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override;
void SetRtpExtensions(std::vector<RtpExtension> extensions) 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);
void SetSyncGroup(const std::string& sync_group);
void SetLocalSsrc(uint32_t local_ssrc);
uint32_t local_ssrc() const;
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);
AudioState* audio_state() const;
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
// TODO( 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_
} // namespace internal
} // namespace webrtc