// Store current audio level in the RTP/RTCP module.
// The level will be used in combination with voice-activity state
// (frameType) to add an RTP header extension
- _rtpRtcpModule->SetAudioLevel(rtp_audioproc_->level_estimator()->RMS());
+ _rtpRtcpModule->SetAudioLevel(rms_level_.RMS());
}
// Push data from ACM to RTP/RTCP-module to deliver audio frame for
isAlive = (_outputSpeechType != AudioFrame::kPLCCNG);
}
- UpdateDeadOrAliveCounters(isAlive);
-
// Send callback to the registered observer
if (_connectionObserver)
{
MixAudioWithFile(audioFrame, audioFrame.sample_rate_hz_);
}
- // Place channel in on-hold state (~muted) if on-hold is activated
- if (state.output_is_on_hold)
- {
- AudioFrameOperations::Mute(audioFrame);
- }
-
// External media
if (_outputExternalMedia)
{
_channelId(channelId),
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(
- new RTPPayloadRegistry(channelId,
- RTPPayloadStrategy::CreateStrategy(true))),
+ new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
rtp_receive_statistics_(ReceiveStatistics::Create(
Clock::GetRealTimeClock())),
rtp_receiver_(RtpReceiver::CreateAudioReceiver(
VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this,
this, this, rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
- audio_coding_(config.Get<AudioCodingModuleFactory>().Create(
+ audio_coding_(AudioCodingModule::Create(
VoEModuleId(instanceId, channelId))),
_rtpDumpIn(*RtpDump::CreateRtpDump()),
_rtpDumpOut(*RtpDump::CreateRtpDump()),
_voiceEngineObserverPtr(NULL),
_callbackCritSectPtr(NULL),
_transportPtr(NULL),
- rx_audioproc_(AudioProcessing::Create(VoEModuleId(instanceId, channelId))),
_rxVadObserverPtr(NULL),
_oldVadDecision(-1),
_sendFrameType(0),
_rtcpObserverPtr(NULL),
_externalPlayout(false),
_externalMixing(false),
- _inputIsOnHold(false),
_mixFileWithMicrophone(false),
_rtpObserver(false),
_rtcpObserver(false),
_rtpTimeOutSeconds(0),
_connectionObserver(false),
_connectionObserverPtr(NULL),
- _countAliveDetections(0),
- _countDeadDetections(0),
_outputSpeechType(AudioFrame::kNormalSpeech),
vie_network_(NULL),
video_channel_(-1),
statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC()));
rtp_receive_statistics_->RegisterRtcpStatisticsCallback(
statistics_proxy_.get());
+
+ Config audioproc_config;
+ audioproc_config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
+ rx_audioproc_.reset(AudioProcessing::Create(audioproc_config));
}
Channel::~Channel()
return 0;
}
- // Recover DTMF detection status.
- telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
- RegisterReceiveCodecsToRTPModule();
channel_state_.SetReceiving(false);
return 0;
}
}
int32_t
-Channel::SetOnHoldStatus(bool enable, OnHoldModes mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetOnHoldStatus()");
- if (mode == kHoldSendAndPlay)
- {
- channel_state_.SetOutputIsOnHold(enable);
- _inputIsOnHold = enable;
- }
- else if (mode == kHoldPlayOnly)
- {
- channel_state_.SetOutputIsOnHold(enable);
- }
- if (mode == kHoldSendOnly)
- {
- _inputIsOnHold = enable;
- }
- return 0;
-}
-
-int32_t
-Channel::GetOnHoldStatus(bool& enabled, OnHoldModes& mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetOnHoldStatus()");
- bool output_is_on_hold = channel_state_.Get().output_is_on_hold;
- enabled = (output_is_on_hold || _inputIsOnHold);
- if (output_is_on_hold && _inputIsOnHold)
- {
- mode = kHoldSendAndPlay;
- }
- else if (output_is_on_hold && !_inputIsOnHold)
- {
- mode = kHoldPlayOnly;
- }
- else if (!output_is_on_hold && _inputIsOnHold)
- {
- mode = kHoldSendOnly;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetOnHoldStatus() => enabled=%d, mode=%d",
- enabled, mode);
- return 0;
-}
-
-int32_t
Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
}
int32_t
-Channel::SetAMREncFormat(AmrMode mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetAMREncFormat()");
-
- // ACM doesn't support AMR
- return -1;
-}
-
-int32_t
-Channel::SetAMRDecFormat(AmrMode mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetAMRDecFormat()");
-
- // ACM doesn't support AMR
- return -1;
-}
-
-int32_t
-Channel::SetAMRWbEncFormat(AmrMode mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetAMRWbEncFormat()");
-
- // ACM doesn't support AMR
- return -1;
-
-}
-
-int32_t
-Channel::SetAMRWbDecFormat(AmrMode mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetAMRWbDecFormat()");
-
- // ACM doesn't support AMR
- return -1;
-}
-
-int32_t
Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
return 0;
}
-int32_t
-Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetISACInitTargetRate()");
-
- CodecInst sendCodec;
- if (audio_coding_->SendCodec(&sendCodec) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACInitTargetRate() failed to retrieve send codec");
- return -1;
- }
- if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
- {
- // This API is only valid if iSAC is setup to run in channel-adaptive
- // mode.
- // We do not validate the adaptive mode here. It is done later in the
- // ConfigISACBandwidthEstimator() API.
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACInitTargetRate() send codec is not iSAC");
- return -1;
- }
-
- uint8_t initFrameSizeMsec(0);
- if (16000 == sendCodec.plfreq)
- {
- // Note that 0 is a valid and corresponds to "use default
- if ((rateBps != 0 &&
- rateBps < kVoiceEngineMinIsacInitTargetRateBpsWb) ||
- (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsWb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACInitTargetRate() invalid target rate - 1");
- return -1;
- }
- // 30 or 60ms
- initFrameSizeMsec = (uint8_t)(sendCodec.pacsize / 16);
- }
- else if (32000 == sendCodec.plfreq)
- {
- if ((rateBps != 0 &&
- rateBps < kVoiceEngineMinIsacInitTargetRateBpsSwb) ||
- (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsSwb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACInitTargetRate() invalid target rate - 2");
- return -1;
- }
- initFrameSizeMsec = (uint8_t)(sendCodec.pacsize / 32); // 30ms
- }
-
- if (audio_coding_->ConfigISACBandwidthEstimator(
- initFrameSizeMsec, rateBps, useFixedFrameSize) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetISACInitTargetRate() iSAC BWE config failed");
- return -1;
- }
-
- return 0;
-}
-
-int32_t
-Channel::SetISACMaxRate(int rateBps)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetISACMaxRate()");
-
- CodecInst sendCodec;
- if (audio_coding_->SendCodec(&sendCodec) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACMaxRate() failed to retrieve send codec");
- return -1;
- }
- if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
- {
- // This API is only valid if iSAC is selected as sending codec.
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACMaxRate() send codec is not iSAC");
- return -1;
- }
- if (16000 == sendCodec.plfreq)
- {
- if ((rateBps < kVoiceEngineMinIsacMaxRateBpsWb) ||
- (rateBps > kVoiceEngineMaxIsacMaxRateBpsWb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACMaxRate() invalid max rate - 1");
- return -1;
- }
- }
- else if (32000 == sendCodec.plfreq)
- {
- if ((rateBps < kVoiceEngineMinIsacMaxRateBpsSwb) ||
- (rateBps > kVoiceEngineMaxIsacMaxRateBpsSwb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACMaxRate() invalid max rate - 2");
- return -1;
- }
- }
- if (channel_state_.Get().sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SENDING, kTraceError,
- "SetISACMaxRate() unable to set max rate while sending");
- return -1;
- }
-
- // Set the maximum instantaneous rate of iSAC (works for both adaptive
- // and non-adaptive mode)
- if (audio_coding_->SetISACMaxRate(rateBps) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetISACMaxRate() failed to set max rate");
- return -1;
- }
-
- return 0;
-}
-
-int32_t
-Channel::SetISACMaxPayloadSize(int sizeBytes)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetISACMaxPayloadSize()");
- CodecInst sendCodec;
- if (audio_coding_->SendCodec(&sendCodec) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACMaxPayloadSize() failed to retrieve send codec");
- return -1;
- }
- if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACMaxPayloadSize() send codec is not iSAC");
- return -1;
- }
- if (16000 == sendCodec.plfreq)
- {
- if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesWb) ||
- (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesWb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACMaxPayloadSize() invalid max payload - 1");
- return -1;
- }
- }
- else if (32000 == sendCodec.plfreq)
- {
- if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesSwb) ||
- (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACMaxPayloadSize() invalid max payload - 2");
- return -1;
- }
- }
- if (channel_state_.Get().sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SENDING, kTraceError,
- "SetISACMaxPayloadSize() unable to set max rate while sending");
- return -1;
- }
-
- if (audio_coding_->SetISACMaxPayloadSize(sizeBytes) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetISACMaxPayloadSize() failed to set max payload size");
- return -1;
- }
- return 0;
-}
-
int32_t Channel::RegisterExternalTransport(Transport& transport)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
}
int Channel::SetSendAudioLevelIndicationStatus(bool enable, unsigned char id) {
- if (rtp_audioproc_.get() == NULL) {
- rtp_audioproc_.reset(AudioProcessing::Create(VoEModuleId(_instanceId,
- _channelId)));
- }
+ _includeAudioLevelIndication = enable;
+ return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
+}
- if (rtp_audioproc_->level_estimator()->Enable(enable) !=
- AudioProcessing::kNoError) {
- _engineStatisticsPtr->SetLastError(VE_APM_ERROR, kTraceError,
- "Failed to enable AudioProcessing::level_estimator()");
+int Channel::SetReceiveAudioLevelIndicationStatus(bool enable,
+ unsigned char id) {
+ rtp_header_parser_->DeregisterRtpHeaderExtension(
+ kRtpExtensionAudioLevel);
+ if (enable && !rtp_header_parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionAudioLevel, id)) {
return -1;
}
-
- _includeAudioLevelIndication = enable;
-
- return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
+ return 0;
}
int Channel::SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
return 0;
}
-// TODO(xians): This method borrows quite some code from
-// TransmitMixer::GenerateAudioFrame(), refactor these two methods and reduce
-// code duplication.
void Channel::Demultiplex(const int16_t* audio_data,
int sample_rate,
int number_of_frames,
int number_of_channels) {
- // The highest sample rate that WebRTC supports for mono audio is 96kHz.
- static const int kMaxNumberOfFrames = 960;
- assert(number_of_frames <= kMaxNumberOfFrames);
-
- // Get the send codec information for doing resampling or downmixing later on.
CodecInst codec;
GetSendCodec(codec);
- assert(codec.channels == 1 || codec.channels == 2);
- int support_sample_rate = std::min(32000,
- std::min(sample_rate, codec.plfreq));
-
- // Downmix the data to mono if needed.
- const int16_t* audio_ptr = audio_data;
- if (number_of_channels == 2 && codec.channels == 1) {
- if (!mono_recording_audio_.get())
- mono_recording_audio_.reset(new int16_t[kMaxNumberOfFrames]);
-
- AudioFrameOperations::StereoToMono(audio_data, number_of_frames,
- mono_recording_audio_.get());
- audio_ptr = mono_recording_audio_.get();
- }
- // Resample the data to the sample rate that the codec is using.
- if (input_resampler_.InitializeIfNeeded(sample_rate,
- support_sample_rate,
- codec.channels)) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "Channel::Demultiplex() unable to resample");
- return;
- }
-
- int out_length = input_resampler_.Resample(audio_ptr,
- number_of_frames * codec.channels,
- _audioFrame.data_,
- AudioFrame::kMaxDataSizeSamples);
- if (out_length == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "Channel::Demultiplex() resampling failed");
- return;
+ if (!mono_recording_audio_.get()) {
+ // Temporary space for DownConvertToCodecFormat.
+ mono_recording_audio_.reset(new int16_t[kMaxMonoDataSizeSamples]);
}
-
- _audioFrame.samples_per_channel_ = out_length / codec.channels;
- _audioFrame.timestamp_ = -1;
- _audioFrame.sample_rate_hz_ = support_sample_rate;
- _audioFrame.speech_type_ = AudioFrame::kNormalSpeech;
- _audioFrame.vad_activity_ = AudioFrame::kVadUnknown;
- _audioFrame.num_channels_ = codec.channels;
- _audioFrame.id_ = _channelId;
+ DownConvertToCodecFormat(audio_data,
+ number_of_frames,
+ number_of_channels,
+ sample_rate,
+ codec.channels,
+ codec.plfreq,
+ mono_recording_audio_.get(),
+ &input_resampler_,
+ &_audioFrame);
}
uint32_t
InsertInbandDtmfTone();
if (_includeAudioLevelIndication) {
- // Performs level analysis only; does not affect the signal.
- int err = rtp_audioproc_->ProcessStream(&_audioFrame);
- if (err) {
- LOG(LS_ERROR) << "ProcessStream() error: " << err;
- assert(false);
- }
+ int length = _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
+ rms_level_.Process(_audioFrame.data_, length);
}
return 0;
}
int
-Channel::ResetRTCPStatistics()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ResetRTCPStatistics()");
- uint32_t remoteSSRC(0);
- remoteSSRC = rtp_receiver_->SSRC();
- return _rtpRtcpModule->ResetRTT(remoteSSRC);
-}
-
-int
-Channel::GetRoundTripTimeSummary(StatVal& delaysMs) const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRoundTripTimeSummary()");
- // Override default module outputs for the case when RTCP is disabled.
- // This is done to ensure that we are backward compatible with the
- // VoiceEngine where we did not use RTP/RTCP module.
- if (!_rtpRtcpModule->RTCP())
- {
- delaysMs.min = -1;
- delaysMs.max = -1;
- delaysMs.average = -1;
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRoundTripTimeSummary() RTCP is disabled =>"
- " valid RTT measurements cannot be retrieved");
- return 0;
- }
-
- uint32_t remoteSSRC;
- uint16_t RTT;
- uint16_t avgRTT;
- uint16_t maxRTT;
- uint16_t minRTT;
- // The remote SSRC will be zero if no RTP packet has been received.
- remoteSSRC = rtp_receiver_->SSRC();
- if (remoteSSRC == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRoundTripTimeSummary() unable to measure RTT"
- " since no RTP packet has been received yet");
- }
-
- // Retrieve RTT statistics from the RTP/RTCP module for the specified
- // channel and SSRC. The SSRC is required to parse out the correct source
- // in conference scenarios.
- if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT,&maxRTT) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "GetRoundTripTimeSummary unable to retrieve RTT values"
- " from the RTCP layer");
- delaysMs.min = -1; delaysMs.max = -1; delaysMs.average = -1;
- }
- else
- {
- delaysMs.min = minRTT;
- delaysMs.max = maxRTT;
- delaysMs.average = avgRTT;
- }
- return 0;
-}
-
-int
Channel::GetNetworkStatistics(NetworkStatistics& stats)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
int32_t
Channel::MixOrReplaceAudioWithFile(int mixingFrequency)
{
- scoped_array<int16_t> fileBuffer(new int16_t[640]);
+ scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
int fileSamples(0);
{
{
// Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo.
- Utility::MixWithSat(_audioFrame.data_,
- _audioFrame.num_channels_,
- fileBuffer.get(),
- 1,
- fileSamples);
+ MixWithSat(_audioFrame.data_,
+ _audioFrame.num_channels_,
+ fileBuffer.get(),
+ 1,
+ fileSamples);
}
else
{
{
assert(mixingFrequency <= 32000);
- scoped_array<int16_t> fileBuffer(new int16_t[640]);
+ scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
int fileSamples(0);
{
{
// Currently file stream is always mono.
// TODO(xians): Change the code when FilePlayer supports real stereo.
- Utility::MixWithSat(audioFrame.data_,
- audioFrame.num_channels_,
- fileBuffer.get(),
- 1,
- fileSamples);
+ MixWithSat(audioFrame.data_,
+ audioFrame.num_channels_,
+ fileBuffer.get(),
+ 1,
+ fileSamples);
}
else
{
return 0;
}
-void
-Channel::ResetDeadOrAliveCounters()
-{
- _countDeadDetections = 0;
- _countAliveDetections = 0;
-}
-
-void
-Channel::UpdateDeadOrAliveCounters(bool alive)
-{
- if (alive)
- _countAliveDetections++;
- else
- _countDeadDetections++;
-}
-
-int
-Channel::GetDeadOrAliveCounters(int& countDead, int& countAlive) const
-{
- return 0;
-}
-
int32_t
Channel::SendPacketRaw(const void *data, int len, bool RTCP)
{
projects / platform / framework / web / crosswalk.git / blobdiff