clang-format -i -style=file webrtc/voice_engine/channel.*
This CL changes literally nothing else. Review URL: https://codereview.webrtc.org/1644633005 Cr-Commit-Position: refs/heads/master@{#11416}
This commit is contained in:
kwiberg
Commit bot
parent
6043f2e5d6
commit
55b97fe388
File diff suppressed because it is too large
Load Diff
@ -84,526 +84,515 @@ class VoERtcpObserver;
|
||||
// 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) {}
|
||||
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;
|
||||
};
|
||||
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() {}
|
||||
ChannelState() {}
|
||||
virtual ~ChannelState() {}
|
||||
|
||||
void Reset() {
|
||||
rtc::CritScope lock(&lock_);
|
||||
state_ = State();
|
||||
}
|
||||
void Reset() {
|
||||
rtc::CritScope lock(&lock_);
|
||||
state_ = State();
|
||||
}
|
||||
|
||||
State Get() const {
|
||||
rtc::CritScope lock(&lock_);
|
||||
return 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 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 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 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 SetInputFilePlaying(bool enable) {
|
||||
rtc::CritScope lock(&lock_);
|
||||
state_.input_file_playing = enable;
|
||||
}
|
||||
|
||||
void SetPlaying(bool enable) {
|
||||
rtc::CritScope lock(&lock_);
|
||||
state_.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 SetSending(bool enable) {
|
||||
rtc::CritScope lock(&lock_);
|
||||
state_.sending = enable;
|
||||
}
|
||||
|
||||
void SetReceiving(bool enable) {
|
||||
rtc::CritScope lock(&lock_);
|
||||
state_.receiving = enable;
|
||||
}
|
||||
void SetReceiving(bool enable) {
|
||||
rtc::CritScope lock(&lock_);
|
||||
state_.receiving = enable;
|
||||
}
|
||||
|
||||
private:
|
||||
mutable rtc::CriticalSection lock_;
|
||||
State state_;
|
||||
private:
|
||||
mutable rtc::CriticalSection lock_;
|
||||
State state_;
|
||||
};
|
||||
|
||||
class Channel:
|
||||
public RtpData,
|
||||
public RtpFeedback,
|
||||
public FileCallback, // receiving notification from file player & recorder
|
||||
public Transport,
|
||||
public RtpAudioFeedback,
|
||||
public AudioPacketizationCallback, // receive encoded packets from the ACM
|
||||
public ACMVADCallback, // receive voice activity from the ACM
|
||||
public MixerParticipant // supplies output mixer with audio frames
|
||||
class Channel
|
||||
: public RtpData,
|
||||
public RtpFeedback,
|
||||
public FileCallback, // receiving notification from file player &
|
||||
// recorder
|
||||
public Transport,
|
||||
public RtpAudioFeedback,
|
||||
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;
|
||||
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);
|
||||
Channel(int32_t channelId,
|
||||
uint32_t instanceId,
|
||||
RtcEventLog* const event_log,
|
||||
const Config& config);
|
||||
int32_t Init();
|
||||
int32_t SetEngineInformation(
|
||||
Statistics& engineStatistics,
|
||||
OutputMixer& outputMixer,
|
||||
TransmitMixer& transmitMixer,
|
||||
ProcessThread& moduleProcessThread,
|
||||
AudioDeviceModule& audioDeviceModule,
|
||||
VoiceEngineObserver* voiceEngineObserver,
|
||||
rtc::CriticalSection* callbackCritSect);
|
||||
int32_t UpdateLocalTimeStamp();
|
||||
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);
|
||||
Channel(int32_t channelId,
|
||||
uint32_t instanceId,
|
||||
RtcEventLog* const event_log,
|
||||
const Config& config);
|
||||
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(rtc::scoped_ptr<AudioSinkInterface> sink);
|
||||
void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
|
||||
|
||||
// API methods
|
||||
// API methods
|
||||
|
||||
// VoEBase
|
||||
int32_t StartPlayout();
|
||||
int32_t StopPlayout();
|
||||
int32_t StartSend();
|
||||
int32_t StopSend();
|
||||
int32_t StartReceiving();
|
||||
int32_t StopReceiving();
|
||||
// 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();
|
||||
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);
|
||||
// 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 int8_t* data, size_t length,
|
||||
const PacketTime& packet_time);
|
||||
int32_t ReceivedRTCPPacket(const int8_t* data, size_t length);
|
||||
// VoENetwork
|
||||
int32_t RegisterExternalTransport(Transport& transport);
|
||||
int32_t DeRegisterExternalTransport();
|
||||
int32_t ReceivedRTPPacket(const int8_t* data,
|
||||
size_t length,
|
||||
const PacketTime& packet_time);
|
||||
int32_t ReceivedRTCPPacket(const int8_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();
|
||||
// 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);
|
||||
void SetMixWithMicStatus(bool mix);
|
||||
|
||||
// VoEExternalMediaProcessing
|
||||
int RegisterExternalMediaProcessing(ProcessingTypes type,
|
||||
VoEMediaProcess& processObject);
|
||||
int DeRegisterExternalMediaProcessing(ProcessingTypes type);
|
||||
int SetExternalMixing(bool enabled);
|
||||
// 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 SetMute(bool enable);
|
||||
bool Mute() const;
|
||||
int SetOutputVolumePan(float left, float right);
|
||||
int GetOutputVolumePan(float& left, float& right) const;
|
||||
int SetChannelOutputVolumeScaling(float scaling);
|
||||
int GetChannelOutputVolumeScaling(float& scaling) const;
|
||||
// VoEVolumeControl
|
||||
int GetSpeechOutputLevel(uint32_t& level) const;
|
||||
int GetSpeechOutputLevelFullRange(uint32_t& level) const;
|
||||
int SetMute(bool enable);
|
||||
bool Mute() 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;
|
||||
// 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);
|
||||
// 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;
|
||||
// VoEVideoSyncExtended
|
||||
int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
|
||||
|
||||
// VoEDtmf
|
||||
int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs,
|
||||
int attenuationDb, bool playDtmfEvent);
|
||||
int SendTelephoneEventInband(unsigned char eventCode, int lengthMs,
|
||||
int attenuationDb, bool playDtmfEvent);
|
||||
int SetSendTelephoneEventPayloadType(unsigned char type);
|
||||
int GetSendTelephoneEventPayloadType(unsigned char& type);
|
||||
// VoEDtmf
|
||||
int SendTelephoneEventOutband(unsigned char eventCode,
|
||||
int lengthMs,
|
||||
int attenuationDb,
|
||||
bool playDtmfEvent);
|
||||
int SendTelephoneEventInband(unsigned char eventCode,
|
||||
int lengthMs,
|
||||
int attenuationDb,
|
||||
bool playDtmfEvent);
|
||||
int SetSendTelephoneEventPayloadType(unsigned char type);
|
||||
int GetSendTelephoneEventPayloadType(unsigned char& type);
|
||||
|
||||
// VoEAudioProcessingImpl
|
||||
int UpdateRxVadDetection(AudioFrame& audioFrame);
|
||||
int RegisterRxVadObserver(VoERxVadCallback &observer);
|
||||
int DeRegisterRxVadObserver();
|
||||
int VoiceActivityIndicator(int &activity);
|
||||
// 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);
|
||||
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);
|
||||
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);
|
||||
// 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 SetCongestionControlObjects(
|
||||
RtpPacketSender* rtp_packet_sender,
|
||||
TransportFeedbackObserver* transport_feedback_observer,
|
||||
PacketRouter* packet_router);
|
||||
void SetCongestionControlObjects(
|
||||
RtpPacketSender* rtp_packet_sender,
|
||||
TransportFeedbackObserver* transport_feedback_observer,
|
||||
PacketRouter* packet_router);
|
||||
|
||||
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 SetREDStatus(bool enable, int redPayloadtype);
|
||||
int GetREDStatus(bool& enabled, int& redPayloadtype);
|
||||
int SetCodecFECStatus(bool enable);
|
||||
bool GetCodecFECStatus();
|
||||
void SetNACKStatus(bool enable, int maxNumberOfPackets);
|
||||
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 SetREDStatus(bool enable, int redPayloadtype);
|
||||
int GetREDStatus(bool& enabled, int& redPayloadtype);
|
||||
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 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;
|
||||
// From ACMVADCallback in the ACM
|
||||
int32_t InFrameType(FrameType frame_type) override;
|
||||
|
||||
int32_t OnRxVadDetected(int vadDecision);
|
||||
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 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 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 RtpAudioFeedback in the RTP/RTCP module
|
||||
void OnPlayTelephoneEvent(uint8_t event,
|
||||
uint16_t lengthMs,
|
||||
uint8_t volume) override;
|
||||
// From RtpAudioFeedback in the RTP/RTCP module
|
||||
void OnPlayTelephoneEvent(uint8_t event,
|
||||
uint16_t lengthMs,
|
||||
uint8_t volume) 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 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
|
||||
int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override;
|
||||
int32_t NeededFrequency(int32_t id) const override;
|
||||
// From MixerParticipant
|
||||
int32_t GetAudioFrame(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;
|
||||
// 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();
|
||||
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;
|
||||
}
|
||||
// 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);
|
||||
// Disassociate a send channel if it was associated.
|
||||
void DisassociateSendChannel(int channel_id);
|
||||
|
||||
protected:
|
||||
void OnIncomingFractionLoss(int fraction_lost);
|
||||
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);
|
||||
int InsertInbandDtmfTone();
|
||||
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();
|
||||
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);
|
||||
int InsertInbandDtmfTone();
|
||||
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 SetRedPayloadType(int red_payload_type);
|
||||
int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type,
|
||||
unsigned char id);
|
||||
int SetRedPayloadType(int red_payload_type);
|
||||
int SetSendRtpHeaderExtension(bool enable,
|
||||
RTPExtensionType type,
|
||||
unsigned char id);
|
||||
|
||||
int32_t GetPlayoutFrequency();
|
||||
int64_t GetRTT(bool allow_associate_channel) const;
|
||||
int32_t GetPlayoutFrequency();
|
||||
int64_t GetRTT(bool allow_associate_channel) const;
|
||||
|
||||
mutable rtc::CriticalSection _fileCritSect;
|
||||
mutable rtc::CriticalSection _callbackCritSect;
|
||||
mutable rtc::CriticalSection volume_settings_critsect_;
|
||||
uint32_t _instanceId;
|
||||
int32_t _channelId;
|
||||
mutable rtc::CriticalSection _fileCritSect;
|
||||
mutable rtc::CriticalSection _callbackCritSect;
|
||||
mutable rtc::CriticalSection volume_settings_critsect_;
|
||||
uint32_t _instanceId;
|
||||
int32_t _channelId;
|
||||
|
||||
ChannelState channel_state_;
|
||||
ChannelState channel_state_;
|
||||
|
||||
RtcEventLog* const event_log_;
|
||||
RtcEventLog* const event_log_;
|
||||
|
||||
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
|
||||
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
|
||||
rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
|
||||
rtc::scoped_ptr<StatisticsProxy> statistics_proxy_;
|
||||
rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
|
||||
TelephoneEventHandler* telephone_event_handler_;
|
||||
rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
|
||||
rtc::scoped_ptr<AudioCodingModule> audio_coding_;
|
||||
rtc::scoped_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;
|
||||
DtmfInbandQueue _inbandDtmfQueue;
|
||||
DtmfInband _inbandDtmfGenerator;
|
||||
bool _outputExternalMedia;
|
||||
VoEMediaProcess* _inputExternalMediaCallbackPtr;
|
||||
VoEMediaProcess* _outputExternalMediaCallbackPtr;
|
||||
uint32_t _timeStamp;
|
||||
uint8_t _sendTelephoneEventPayloadType;
|
||||
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
|
||||
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
|
||||
rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
|
||||
rtc::scoped_ptr<StatisticsProxy> statistics_proxy_;
|
||||
rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
|
||||
TelephoneEventHandler* telephone_event_handler_;
|
||||
rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
|
||||
rtc::scoped_ptr<AudioCodingModule> audio_coding_;
|
||||
rtc::scoped_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;
|
||||
DtmfInbandQueue _inbandDtmfQueue;
|
||||
DtmfInband _inbandDtmfGenerator;
|
||||
bool _outputExternalMedia;
|
||||
VoEMediaProcess* _inputExternalMediaCallbackPtr;
|
||||
VoEMediaProcess* _outputExternalMediaCallbackPtr;
|
||||
uint32_t _timeStamp;
|
||||
uint8_t _sendTelephoneEventPayloadType;
|
||||
|
||||
RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
|
||||
RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
|
||||
|
||||
// Timestamp of the audio pulled from NetEq.
|
||||
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];
|
||||
// Timestamp of the audio pulled from NetEq.
|
||||
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];
|
||||
|
||||
mutable rtc::CriticalSection ts_stats_lock_;
|
||||
mutable rtc::CriticalSection ts_stats_lock_;
|
||||
|
||||
rtc::scoped_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_);
|
||||
rtc::scoped_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_;
|
||||
rtc::scoped_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 _mute;
|
||||
float _panLeft;
|
||||
float _panRight;
|
||||
float _outputGain;
|
||||
// VoEDtmf
|
||||
bool _playOutbandDtmfEvent;
|
||||
bool _playInbandDtmfEvent;
|
||||
// VoeRTP_RTCP
|
||||
uint32_t _lastLocalTimeStamp;
|
||||
int8_t _lastPayloadType;
|
||||
bool _includeAudioLevelIndication;
|
||||
// VoENetwork
|
||||
AudioFrame::SpeechType _outputSpeechType;
|
||||
// VoEVideoSync
|
||||
mutable 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
|
||||
rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
|
||||
rtc::scoped_ptr<NetworkPredictor> network_predictor_;
|
||||
// An associated send channel.
|
||||
mutable rtc::CriticalSection assoc_send_channel_lock_;
|
||||
ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_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_;
|
||||
rtc::scoped_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 _mute;
|
||||
float _panLeft;
|
||||
float _panRight;
|
||||
float _outputGain;
|
||||
// VoEDtmf
|
||||
bool _playOutbandDtmfEvent;
|
||||
bool _playInbandDtmfEvent;
|
||||
// VoeRTP_RTCP
|
||||
uint32_t _lastLocalTimeStamp;
|
||||
int8_t _lastPayloadType;
|
||||
bool _includeAudioLevelIndication;
|
||||
// VoENetwork
|
||||
AudioFrame::SpeechType _outputSpeechType;
|
||||
// VoEVideoSync
|
||||
mutable 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
|
||||
rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
|
||||
rtc::scoped_ptr<NetworkPredictor> network_predictor_;
|
||||
// An associated send channel.
|
||||
mutable rtc::CriticalSection assoc_send_channel_lock_;
|
||||
ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
|
||||
|
||||
bool pacing_enabled_;
|
||||
PacketRouter* packet_router_ = nullptr;
|
||||
rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
|
||||
rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
|
||||
rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
|
||||
bool pacing_enabled_;
|
||||
PacketRouter* packet_router_ = nullptr;
|
||||
rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
|
||||
rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
|
||||
rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
|
||||
};
|
||||
|
||||
} // namespace voe
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user