rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/voice_engine/channel.h

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22 KiB
C++

/*
* 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.
*/
#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
#define WEBRTC_VOICE_ENGINE_CHANNEL_H_
#include <memory>
#include "webrtc/audio_sink.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/optional.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_processing/rms_level.h"
#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/file_player.h"
#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/network_predictor.h"
#include "webrtc/voice_engine/shared_data.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
namespace rtc {
class TimestampWrapAroundHandler;
}
namespace webrtc {
class AudioDeviceModule;
class Config;
class FileWrapper;
class PacketRouter;
class ProcessThread;
class ReceiveStatistics;
class RemoteNtpTimeEstimator;
class RtcEventLog;
class RTPPayloadRegistry;
class RtpReceiver;
class RTPReceiverAudio;
class RtpRtcp;
class TelephoneEventHandler;
class VoEMediaProcess;
class VoERTPObserver;
class VoiceEngineObserver;
struct CallStatistics;
struct ReportBlock;
struct SenderInfo;
namespace voe {
class OutputMixer;
class RtpPacketSenderProxy;
class Statistics;
class StatisticsProxy;
class TransportFeedbackProxy;
class TransmitMixer;
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 {
public:
struct State {
State()
: rx_apm_is_enabled(false),
input_external_media(false),
output_file_playing(false),
input_file_playing(false),
playing(false),
sending(false),
receiving(false) {}
bool rx_apm_is_enabled;
bool input_external_media;
bool output_file_playing;
bool input_file_playing;
bool playing;
bool sending;
bool receiving;
};
ChannelState() {}
virtual ~ChannelState() {}
void Reset() {
rtc::CritScope lock(&lock_);
state_ = State();
}
State Get() const {
rtc::CritScope lock(&lock_);
return state_;
}
void SetRxApmIsEnabled(bool enable) {
rtc::CritScope lock(&lock_);
state_.rx_apm_is_enabled = enable;
}
void SetInputExternalMedia(bool enable) {
rtc::CritScope lock(&lock_);
state_.input_external_media = enable;
}
void SetOutputFilePlaying(bool enable) {
rtc::CritScope lock(&lock_);
state_.output_file_playing = enable;
}
void SetInputFilePlaying(bool enable) {
rtc::CritScope lock(&lock_);
state_.input_file_playing = enable;
}
void SetPlaying(bool enable) {
rtc::CritScope lock(&lock_);
state_.playing = enable;
}
void SetSending(bool enable) {
rtc::CritScope lock(&lock_);
state_.sending = enable;
}
void SetReceiving(bool enable) {
rtc::CritScope lock(&lock_);
state_.receiving = enable;
}
private:
rtc::CriticalSection lock_;
State state_;
};
class Channel
: public RtpData,
public RtpFeedback,
public FileCallback, // receiving notification from file player &
// recorder
public Transport,
public AudioPacketizationCallback, // receive encoded packets from the
// ACM
public ACMVADCallback, // receive voice activity from the ACM
public MixerParticipant // supplies output mixer with audio frames
{
public:
friend class VoERtcpObserver;
enum { KNumSocketThreads = 1 };
enum { KNumberOfSocketBuffers = 8 };
virtual ~Channel();
static int32_t CreateChannel(Channel*& channel,
int32_t channelId,
uint32_t instanceId,
RtcEventLog* const event_log,
const Config& config);
static int32_t CreateChannel(
Channel*& channel,
int32_t channelId,
uint32_t instanceId,
RtcEventLog* const event_log,
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
Channel(int32_t channelId,
uint32_t instanceId,
RtcEventLog* const event_log,
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
int32_t Init();
int32_t SetEngineInformation(Statistics& engineStatistics,
OutputMixer& outputMixer,
TransmitMixer& transmitMixer,
ProcessThread& moduleProcessThread,
AudioDeviceModule& audioDeviceModule,
VoiceEngineObserver* voiceEngineObserver,
rtc::CriticalSection* callbackCritSect);
int32_t UpdateLocalTimeStamp();
void SetSink(std::unique_ptr<AudioSinkInterface> sink);
// TODO(ossu): Don't use! It's only here to confirm that the decoder factory
// passed into AudioReceiveStream is the same as the one set when creating the
// ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can
// go.
const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
// API methods
// VoEBase
int32_t StartPlayout();
int32_t StopPlayout();
int32_t StartSend();
int32_t StopSend();
int32_t StartReceiving();
int32_t StopReceiving();
int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
int32_t DeRegisterVoiceEngineObserver();
// VoECodec
int32_t GetSendCodec(CodecInst& codec);
int32_t GetRecCodec(CodecInst& codec);
int32_t SetSendCodec(const CodecInst& codec);
void SetBitRate(int bitrate_bps);
int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX);
int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX);
int32_t SetRecPayloadType(const CodecInst& codec);
int32_t GetRecPayloadType(CodecInst& codec);
int32_t SetSendCNPayloadType(int type, PayloadFrequencies frequency);
int SetOpusMaxPlaybackRate(int frequency_hz);
int SetOpusDtx(bool enable_dtx);
// VoENetwork
int32_t RegisterExternalTransport(Transport* transport);
int32_t DeRegisterExternalTransport();
int32_t ReceivedRTPPacket(const uint8_t* received_packet,
size_t length,
const PacketTime& packet_time);
int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
// VoEFile
int StartPlayingFileLocally(const char* fileName,
bool loop,
FileFormats format,
int startPosition,
float volumeScaling,
int stopPosition,
const CodecInst* codecInst);
int StartPlayingFileLocally(InStream* stream,
FileFormats format,
int startPosition,
float volumeScaling,
int stopPosition,
const CodecInst* codecInst);
int StopPlayingFileLocally();
int IsPlayingFileLocally() const;
int RegisterFilePlayingToMixer();
int StartPlayingFileAsMicrophone(const char* fileName,
bool loop,
FileFormats format,
int startPosition,
float volumeScaling,
int stopPosition,
const CodecInst* codecInst);
int StartPlayingFileAsMicrophone(InStream* stream,
FileFormats format,
int startPosition,
float volumeScaling,
int stopPosition,
const CodecInst* codecInst);
int StopPlayingFileAsMicrophone();
int IsPlayingFileAsMicrophone() const;
int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
int StopRecordingPlayout();
void SetMixWithMicStatus(bool mix);
// VoEExternalMediaProcessing
int RegisterExternalMediaProcessing(ProcessingTypes type,
VoEMediaProcess& processObject);
int DeRegisterExternalMediaProcessing(ProcessingTypes type);
int SetExternalMixing(bool enabled);
// VoEVolumeControl
int GetSpeechOutputLevel(uint32_t& level) const;
int GetSpeechOutputLevelFullRange(uint32_t& level) const;
int SetInputMute(bool enable);
bool InputMute() const;
int SetOutputVolumePan(float left, float right);
int GetOutputVolumePan(float& left, float& right) const;
int SetChannelOutputVolumeScaling(float scaling);
int GetChannelOutputVolumeScaling(float& scaling) const;
// VoENetEqStats
int GetNetworkStatistics(NetworkStatistics& stats);
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
// VoEVideoSync
bool GetDelayEstimate(int* jitter_buffer_delay_ms,
int* playout_buffer_delay_ms) const;
uint32_t GetDelayEstimate() const;
int LeastRequiredDelayMs() const;
int SetMinimumPlayoutDelay(int delayMs);
int GetPlayoutTimestamp(unsigned int& timestamp);
int SetInitTimestamp(unsigned int timestamp);
int SetInitSequenceNumber(short sequenceNumber);
// VoEVideoSyncExtended
int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
// DTMF
int SendTelephoneEventOutband(int event, int duration_ms);
int SetSendTelephoneEventPayloadType(int payload_type);
// VoEAudioProcessingImpl
int UpdateRxVadDetection(AudioFrame& audioFrame);
int RegisterRxVadObserver(VoERxVadCallback& observer);
int DeRegisterRxVadObserver();
int VoiceActivityIndicator(int& activity);
#ifdef WEBRTC_VOICE_ENGINE_AGC
int SetRxAgcStatus(bool enable, AgcModes mode);
int GetRxAgcStatus(bool& enabled, AgcModes& mode);
int SetRxAgcConfig(AgcConfig config);
int GetRxAgcConfig(AgcConfig& config);
#endif
#ifdef WEBRTC_VOICE_ENGINE_NR
int SetRxNsStatus(bool enable, NsModes mode);
int GetRxNsStatus(bool& enabled, NsModes& mode);
#endif
// VoERTP_RTCP
int SetLocalSSRC(unsigned int ssrc);
int GetLocalSSRC(unsigned int& ssrc);
int GetRemoteSSRC(unsigned int& ssrc);
int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
void EnableSendTransportSequenceNumber(int id);
void EnableReceiveTransportSequenceNumber(int id);
void RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router);
void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
void ResetCongestionControlObjects();
void SetRTCPStatus(bool enable);
int GetRTCPStatus(bool& enabled);
int SetRTCP_CNAME(const char cName[256]);
int GetRemoteRTCP_CNAME(char cName[256]);
int GetRemoteRTCPData(unsigned int& NTPHigh,
unsigned int& NTPLow,
unsigned int& timestamp,
unsigned int& playoutTimestamp,
unsigned int* jitter,
unsigned short* fractionLost);
int SendApplicationDefinedRTCPPacket(unsigned char subType,
unsigned int name,
const char* data,
unsigned short dataLengthInBytes);
int GetRTPStatistics(unsigned int& averageJitterMs,
unsigned int& maxJitterMs,
unsigned int& discardedPackets);
int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
int GetRTPStatistics(CallStatistics& stats);
int SetCodecFECStatus(bool enable);
bool GetCodecFECStatus();
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 ACMVADCallback in the ACM
int32_t InFrameType(FrameType frame_type) override;
int32_t OnRxVadDetected(int vadDecision);
// From RtpData in the RTP/RTCP module
int32_t OnReceivedPayloadData(const uint8_t* payloadData,
size_t payloadSize,
const WebRtcRTPHeader* rtpHeader) override;
bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
// From RtpFeedback in the RTP/RTCP module
int32_t OnInitializeDecoder(int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
int frequency,
size_t channels,
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 MixerParticipant
MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted(
int32_t id,
AudioFrame* audioFrame) override;
int32_t NeededFrequency(int32_t id) const override;
// From FileCallback
void PlayNotification(int32_t id, uint32_t durationMs) override;
void RecordNotification(int32_t id, uint32_t durationMs) override;
void PlayFileEnded(int32_t id) override;
void RecordFileEnded(int32_t id) override;
uint32_t InstanceId() const { return _instanceId; }
int32_t ChannelId() const { return _channelId; }
bool Playing() const { return channel_state_.Get().playing; }
bool Sending() const { return channel_state_.Get().sending; }
bool Receiving() const { return channel_state_.Get().receiving; }
bool ExternalTransport() const {
rtc::CritScope cs(&_callbackCritSect);
return _externalTransport;
}
bool ExternalMixing() const { return _externalMixing; }
RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
uint32_t Demultiplex(const AudioFrame& audioFrame);
// Demultiplex the data to the channel's |_audioFrame|. The difference
// between this method and the overloaded method above is that |audio_data|
// does not go through transmit_mixer and APM.
void Demultiplex(const int16_t* audio_data,
int sample_rate,
size_t number_of_frames,
size_t number_of_channels);
uint32_t PrepareEncodeAndSend(int mixingFrequency);
uint32_t EncodeAndSend();
// Associate to a send channel.
// Used for obtaining RTT for a receive-only channel.
void set_associate_send_channel(const ChannelOwner& channel) {
assert(_channelId != channel.channel()->ChannelId());
rtc::CritScope lock(&assoc_send_channel_lock_);
associate_send_channel_ = channel;
}
// Disassociate a send channel if it was associated.
void DisassociateSendChannel(int channel_id);
protected:
void OnIncomingFractionLoss(int fraction_lost);
private:
bool ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header,
bool in_order);
bool HandleRtxPacket(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);
int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
void UpdatePlayoutTimestamp(bool rtcp);
void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber);
void RegisterReceiveCodecsToRTPModule();
int SetSendRtpHeaderExtension(bool enable,
RTPExtensionType type,
unsigned char id);
int32_t GetPlayoutFrequency();
int64_t GetRTT(bool allow_associate_channel) const;
rtc::CriticalSection _fileCritSect;
rtc::CriticalSection _callbackCritSect;
rtc::CriticalSection volume_settings_critsect_;
uint32_t _instanceId;
int32_t _channelId;
ChannelState channel_state_;
RtcEventLog* const event_log_;
std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<StatisticsProxy> statistics_proxy_;
std::unique_ptr<RtpReceiver> rtp_receiver_;
TelephoneEventHandler* telephone_event_handler_;
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
std::unique_ptr<AudioCodingModule> audio_coding_;
acm2::CodecManager codec_manager_;
acm2::RentACodec rent_a_codec_;
std::unique_ptr<AudioSinkInterface> audio_sink_;
AudioLevel _outputAudioLevel;
bool _externalTransport;
AudioFrame _audioFrame;
// Downsamples to the codec rate if necessary.
PushResampler<int16_t> input_resampler_;
FilePlayer* _inputFilePlayerPtr;
FilePlayer* _outputFilePlayerPtr;
FileRecorder* _outputFileRecorderPtr;
int _inputFilePlayerId;
int _outputFilePlayerId;
int _outputFileRecorderId;
bool _outputFileRecording;
bool _outputExternalMedia;
VoEMediaProcess* _inputExternalMediaCallbackPtr;
VoEMediaProcess* _outputExternalMediaCallbackPtr;
uint32_t _timeStamp;
RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
// Timestamp of the audio pulled from NetEq.
rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
uint32_t playout_timestamp_rtcp_;
uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
uint32_t _numberOfDiscardedPackets;
uint16_t send_sequence_number_;
uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
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_ GUARDED_BY(ts_stats_lock_);
// uses
Statistics* _engineStatisticsPtr;
OutputMixer* _outputMixerPtr;
TransmitMixer* _transmitMixerPtr;
ProcessThread* _moduleProcessThreadPtr;
AudioDeviceModule* _audioDeviceModulePtr;
VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base
rtc::CriticalSection* _callbackCritSectPtr; // owned by base
Transport* _transportPtr; // WebRtc socket or external transport
RMSLevel rms_level_;
std::unique_ptr<AudioProcessing> rx_audioproc_; // far end AudioProcessing
VoERxVadCallback* _rxVadObserverPtr;
int32_t _oldVadDecision;
int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
// VoEBase
bool _externalMixing;
bool _mixFileWithMicrophone;
// VoEVolumeControl
bool input_mute_ GUARDED_BY(volume_settings_critsect_);
bool previous_frame_muted_; // Only accessed from PrepareEncodeAndSend().
float _panLeft GUARDED_BY(volume_settings_critsect_);
float _panRight GUARDED_BY(volume_settings_critsect_);
float _outputGain GUARDED_BY(volume_settings_critsect_);
// VoeRTP_RTCP
uint32_t _lastLocalTimeStamp;
int8_t _lastPayloadType;
bool _includeAudioLevelIndication;
// VoENetwork
AudioFrame::SpeechType _outputSpeechType;
// VoEVideoSync
rtc::CriticalSection video_sync_lock_;
uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
uint32_t _previousTimestamp;
uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
// VoEAudioProcessing
bool _RxVadDetection;
bool _rxAgcIsEnabled;
bool _rxNsIsEnabled;
bool restored_packet_in_use_;
// RtcpBandwidthObserver
std::unique_ptr<VoERtcpObserver> rtcp_observer_;
std::unique_ptr<NetworkPredictor> network_predictor_;
// An associated send channel.
rtc::CriticalSection assoc_send_channel_lock_;
ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
bool pacing_enabled_;
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_;
// TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
};
} // namespace voe
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_