blob: 0c509623a766e96e8951b07b18b1ea3433568c95 [file] [log] [blame]
* Copyright (c) 2012 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 <vector>
#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/crypto/cryptooptions.h"
#include "api/media_transport_interface.h"
#include "api/rtpreceiverinterface.h"
#include "audio/audio_level.h"
#include "call/syncable.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/contributing_sources.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_checker.h"
// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
// warnings about use of unsigned short, and non-const reference arguments.
// These need cleanup, in a separate cl.
namespace rtc {
class TimestampWrapAroundHandler;
namespace webrtc {
class AudioDeviceModule;
class FrameDecryptorInterface;
class PacketRouter;
class ProcessThread;
class RateLimiter;
class ReceiveStatistics;
class RtcEventLog;
class RtpPacketReceived;
class RtpRtcp;
struct CallReceiveStatistics {
unsigned short fractionLost; // NOLINT
unsigned int cumulativeLost;
unsigned int extendedMax;
unsigned int jitterSamples;
int64_t rttMs;
size_t bytesReceived;
int packetsReceived;
// The capture ntp time (in local timebase) of the first played out audio
// frame.
int64_t capture_start_ntp_time_ms_;
namespace voe {
class ChannelSend;
// Helper class to simplify locking scheme for members that are accessed from
// multiple threads.
// Example: a member can be set on thread T1 and read by an internal audio
// thread T2. Accessing the member via this class ensures that we are
// safe and also avoid TSan v2 warnings.
class ChannelReceiveState {
struct State {
bool playing = false;
ChannelReceiveState() {}
virtual ~ChannelReceiveState() {}
void Reset() {
rtc::CritScope lock(&lock_);
state_ = State();
State Get() const {
rtc::CritScope lock(&lock_);
return state_;
void SetPlaying(bool enable) {
rtc::CritScope lock(&lock_);
state_.playing = enable;
rtc::CriticalSection lock_;
State state_;
class ChannelReceive : public RtpData, public MediaTransportAudioSinkInterface {
// Used for receive streams.
ChannelReceive(ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
MediaTransportInterface* media_transport,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options);
virtual ~ChannelReceive();
void SetSink(AudioSinkInterface* sink);
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
// API methods
// VoEBase
int32_t StartPlayout();
int32_t StopPlayout();
// Codecs
int32_t GetRecCodec(CodecInst& codec); // NOLINT
// TODO(nisse, solenberg): Delete when VoENetwork is deleted.
int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
void OnRtpPacket(const RtpPacketReceived& packet);
// Muting, Volume and Level.
void SetChannelOutputVolumeScaling(float scaling);
int GetSpeechOutputLevelFullRange() const;
// See description of "totalAudioEnergy" in the WebRTC stats spec:
double GetTotalOutputEnergy() const;
double GetTotalOutputDuration() const;
// Stats.
int GetNetworkStatistics(NetworkStatistics& stats); // NOLINT
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
// Audio+Video Sync.
uint32_t GetDelayEstimate() const;
int SetMinimumPlayoutDelay(int delayMs);
int GetPlayoutTimestamp(unsigned int& timestamp); // NOLINT
// Produces the transport-related timestamps; current_delay_ms is left unset.
absl::optional<Syncable::Info> GetSyncInfo() const;
int SetLocalSSRC(unsigned int ssrc);
void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
void ResetReceiverCongestionControlObjects();
int GetRTPStatistics(CallReceiveStatistics& stats); // NOLINT
void SetNACKStatus(bool enable, int maxNumberOfPackets);
// MediaTransportAudioSinkInterface override;
void OnData(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) override;
// From RtpData in the RTP/RTCP module
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
const WebRtcRTPHeader* rtpHeader) override;
// From AudioMixer::Source.
AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame);
int PreferredSampleRate() const;
// Associate to a send channel.
// Used for obtaining RTT for a receive-only channel.
void SetAssociatedSendChannel(ChannelSend* channel);
std::vector<RtpSource> GetSources() const;
void Init();
void Terminate();
int GetRemoteSSRC(unsigned int& ssrc); // NOLINT
bool ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header);
int ResendPackets(const uint16_t* sequence_numbers, int length);
void UpdatePlayoutTimestamp(bool rtcp);
int GetRtpTimestampRateHz() const;
int64_t GetRTT() const;
rtc::CriticalSection _callbackCritSect;
rtc::CriticalSection volume_settings_critsect_;
ChannelReceiveState channel_state_;
RtcEventLog* const event_log_;
// Indexed by payload type.
std::map<uint8_t, int> payload_type_frequencies_;
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
const uint32_t remote_ssrc_;
// Info for GetSources and GetSyncInfo is updated on network or worker thread,
// queried on the worker thread.
rtc::CriticalSection rtp_sources_lock_;
ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_);
absl::optional<uint32_t> last_received_rtp_timestamp_
absl::optional<int64_t> last_received_rtp_system_time_ms_
absl::optional<uint8_t> last_received_rtp_audio_level_
std::unique_ptr<AudioCodingModule> audio_coding_;
AudioSinkInterface* audio_sink_ = nullptr;
AudioLevel _outputAudioLevel;
RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
// Timestamp of the audio pulled from NetEq.
absl::optional<uint32_t> jitter_buffer_playout_timestamp_;
rtc::CriticalSection video_sync_lock_;
uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_);
uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_);
rtc::CriticalSection ts_stats_lock_;
std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
// The rtp timestamp of the first played out audio frame.
int64_t capture_start_rtp_time_stamp_;
// The capture ntp time (in local timebase) of the first played out audio
// frame.
int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
// uses
ProcessThread* _moduleProcessThreadPtr;
AudioDeviceModule* _audioDeviceModulePtr;
float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
// An associated send channel.
rtc::CriticalSection assoc_send_channel_lock_;
ChannelSend* associated_send_channel_
PacketRouter* packet_router_ = nullptr;
rtc::ThreadChecker construction_thread_;
MediaTransportInterface* const media_transport_;
// E2EE Audio Frame Decryption
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
webrtc::CryptoOptions crypto_options_;
} // namespace voe
} // namespace webrtc