blob: e59cae11adae0f2420753783c717094cfeafda38 [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 <string>
#include <vector>
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/optional.h"
#include "audio/audio_level.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_processing/rms_level.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_receiver.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/event.h"
#include "rtc_base/task_queue.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 PacketRouter;
class ProcessThread;
class RateLimiter;
class ReceiveStatistics;
class RemoteNtpTimeEstimator;
class RtcEventLog;
class RTPPayloadRegistry;
class RTPReceiverAudio;
class RtpPacketReceived;
class RtpRtcp;
class RtpTransportControllerSendInterface;
class TelephoneEventHandler;
struct SenderInfo;
struct CallStatistics {
unsigned short fractionLost; // NOLINT
unsigned int cumulativeLost;
unsigned int extendedMax;
unsigned int jitterSamples;
int64_t rttMs;
size_t bytesSent;
int packetsSent;
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_;
// See section 6.4.2 in for details.
struct ReportBlock {
uint32_t sender_SSRC; // SSRC of sender
uint32_t source_SSRC;
uint8_t fraction_lost;
uint32_t cumulative_num_packets_lost;
uint32_t extended_highest_sequence_number;
uint32_t interarrival_jitter;
uint32_t last_SR_timestamp;
uint32_t delay_since_last_SR;
namespace voe {
class RtcEventLogProxy;
class RtcpRttStatsProxy;
class RtpPacketSenderProxy;
class TransportFeedbackProxy;
class TransportSequenceNumberProxy;
class VoERtcpObserver;
// 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 ChannelState {
struct State {
bool playing = false;
bool sending = false;
ChannelState() {}
virtual ~ChannelState() {}
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;
void SetSending(bool enable) {
rtc::CritScope lock(&lock_);
state_.sending = enable;
rtc::CriticalSection lock_;
State state_;
class Channel
: public RtpData,
public RtpFeedback,
public Transport,
public AudioPacketizationCallback, // receive encoded packets from the
// ACM
public OverheadObserver {
friend class VoERtcpObserver;
enum { KNumSocketThreads = 1 };
enum { KNumberOfSocketBuffers = 8 };
// Used for send streams.
Channel(rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module);
// Used for receive streams.
Channel(ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
virtual ~Channel();
void SetSink(AudioSinkInterface* sink);
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
// Send using this encoder, with this payload type.
bool SetEncoder(int payload_type, std::unique_ptr<AudioEncoder> encoder);
void ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
// API methods
// VoEBase
int32_t StartPlayout();
int32_t StopPlayout();
int32_t StartSend();
void StopSend();
// Codecs
int32_t GetRecCodec(CodecInst& codec); // NOLINT
void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
bool EnableAudioNetworkAdaptor(const std::string& config_string);
void DisableAudioNetworkAdaptor();
void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms);
// Network
void RegisterTransport(Transport* transport);
// 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 SetInputMute(bool enable);
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;
ANAStats GetANAStatistics() const;
// Audio+Video Sync.
uint32_t GetDelayEstimate() const;
int SetMinimumPlayoutDelay(int delayMs);
int GetPlayoutTimestamp(unsigned int& timestamp); // NOLINT
int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
// DTMF.
int SendTelephoneEventOutband(int event, int duration_ms);
int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
int SetLocalSSRC(unsigned int ssrc);
int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
void EnableSendTransportSequenceNumber(int id);
void RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer);
void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
void ResetSenderCongestionControlObjects();
void ResetReceiverCongestionControlObjects();
void SetRTCPStatus(bool enable);
int SetRTCP_CNAME(const char cName[256]);
int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
int GetRTPStatistics(CallStatistics& stats); // NOLINT
void SetNACKStatus(bool enable, int maxNumberOfPackets);
// From AudioPacketizationCallback in the ACM
int32_t SendData(FrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
// From RtpData in the RTP/RTCP module
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
const WebRtcRTPHeader* rtpHeader) override;
// From RtpFeedback in the RTP/RTCP module
int32_t OnInitializeDecoder(int payload_type,
const SdpAudioFormat& audio_format,
uint32_t rate) override;
void OnIncomingSSRCChanged(uint32_t ssrc) override;
void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
// From Transport (called by the RTP/RTCP module)
bool SendRtp(const uint8_t* data,
size_t len,
const PacketOptions& packet_options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
// From AudioMixer::Source.
AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame);
int PreferredSampleRate() const;
bool Playing() const { return channel_state_.Get().playing; }
bool Sending() const { return channel_state_.Get().sending; }
RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
// ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
// which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
// the actual processing of the audio takes place. The processing mainly
// consists of encoding and preparing the result for sending by adding it to a
// send queue.
// The main reason for using a task queue here is to release the native,
// OS-specific, audio capture thread as soon as possible to ensure that it
// can go back to sleep and be prepared to deliver an new captured audio
// packet.
void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame);
// Associate to a send channel.
// Used for obtaining RTT for a receive-only channel.
void SetAssociatedSendChannel(Channel* channel);
// Set a RtcEventLog logging object.
void SetRtcEventLog(RtcEventLog* event_log);
void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
void SetTransportOverhead(size_t transport_overhead_per_packet);
// From OverheadObserver in the RTP/RTCP module
void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
// The existence of this function alongside OnUplinkPacketLossRate is
// a compromise. We want the encoder to be agnostic of the PLR source, but
// we also don't want it to receive conflicting information from TWCC and
// from RTCP-XR.
void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate);
std::vector<RtpSource> GetSources() const {
return rtp_receiver_->GetSources();
class ProcessAndEncodeAudioTask;
void Init();
void Terminate();
int GetRemoteSSRC(unsigned int& ssrc); // NOLINT
void OnUplinkPacketLossRate(float packet_loss_rate);
bool InputMute() const;
bool ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header);
bool IsPacketInOrder(const RTPHeader& header) const;
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
int ResendPackets(const uint16_t* sequence_numbers, int length);
void UpdatePlayoutTimestamp(bool rtcp);
int SetSendRtpHeaderExtension(bool enable,
RTPExtensionType type,
unsigned char id);
void UpdateOverheadForEncoder()
int GetRtpTimestampRateHz() const;
int64_t GetRTT(bool allow_associate_channel) const;
// Called on the encoder task queue when a new input audio frame is ready
// for encoding.
void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input);
rtc::CriticalSection _callbackCritSect;
rtc::CriticalSection volume_settings_critsect_;
ChannelState channel_state_;
std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_;
std::unique_ptr<voe::RtcpRttStatsProxy> rtcp_rtt_stats_proxy_;
std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<RtpReceiver> rtp_receiver_;
TelephoneEventHandler* telephone_event_handler_;
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
std::unique_ptr<AudioCodingModule> audio_coding_;
AudioSinkInterface* audio_sink_ = nullptr;
AudioLevel _outputAudioLevel;
uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);
RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
// Timestamp of the audio pulled from NetEq.
rtc::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_);
uint16_t send_sequence_number_;
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;
Transport* _transportPtr; // WebRtc socket or external transport
RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
// TODO(henrika): can today be accessed on the main thread and on the
// task queue; hence potential race.
bool _includeAudioLevelIndication;
size_t transport_overhead_per_packet_
size_t rtp_overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_);
rtc::CriticalSection overhead_per_packet_lock_;
// RtcpBandwidthObserver
std::unique_ptr<VoERtcpObserver> rtcp_observer_;
// An associated send channel.
rtc::CriticalSection assoc_send_channel_lock_;
Channel* associated_send_channel_ RTC_GUARDED_BY(assoc_send_channel_lock_);
bool pacing_enabled_ = true;
PacketRouter* packet_router_ = nullptr;
std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
rtc::ThreadChecker construction_thread_;
const bool use_twcc_plr_for_ana_;
rtc::CriticalSection encoder_queue_lock_;
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
rtc::TaskQueue* encoder_queue_ = nullptr;
} // namespace voe
} // namespace webrtc
#endif // AUDIO_CHANNEL_H_