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Unified Diff: webrtc/voice_engine/channel.cc

Issue 3013033002: Remove VoEFile (Closed)
Patch Set: rebase Created 3 years, 3 months ago
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Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d1c3a4bac69ec767a69f86ccb1745932f2b35a85..ac21e8dec322127fddb94b1f118f9a709b7ca8b5 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -651,8 +651,6 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
// Store speech type for dead-or-alive detection
_outputSpeechType = audioFrame->speech_type_;
- ChannelState::State state = channel_state_.Get();
-
{
// Pass the audio buffers to an optional sink callback, before applying
// scaling/panning, as that applies to the mix operation.
@@ -680,21 +678,6 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
AudioFrameOperations::ScaleWithSat(output_gain, audioFrame);
}
- // Mix decoded PCM output with file if file mixing is enabled
- if (state.output_file_playing) {
- MixAudioWithFile(*audioFrame, audioFrame->sample_rate_hz_);
- muted = false; // We may have added non-zero samples.
- }
-
- // Record playout if enabled
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (_outputFileRecording && output_file_recorder_) {
- output_file_recorder_->RecordAudioToFile(*audioFrame);
- }
- }
-
// Measure audio level (0-9)
// TODO(henrik.lundin) Use the |muted| information here too.
// TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see
@@ -774,20 +757,7 @@ int32_t Channel::NeededFrequency(int32_t id) const {
highestNeeded = receiveFrequency;
}
- // Special case, if we're playing a file on the playout side
- // we take that frequency into consideration as well
- // This is not needed on sending side, since the codec will
- // limit the spectrum anyway.
- if (channel_state_.Get().output_file_playing) {
- rtc::CritScope cs(&_fileCritSect);
- if (output_file_player_) {
- if (output_file_player_->Frequency() > highestNeeded) {
- highestNeeded = output_file_player_->Frequency();
- }
- }
- }
-
- return (highestNeeded);
+ return highestNeeded;
}
int32_t Channel::CreateChannel(Channel*& channel,
@@ -808,53 +778,6 @@ int32_t Channel::CreateChannel(Channel*& channel,
return 0;
}
-void Channel::PlayNotification(int32_t id, uint32_t durationMs) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::PlayNotification(id=%d, durationMs=%d)", id,
- durationMs);
-
- // Not implement yet
-}
-
-void Channel::RecordNotification(int32_t id, uint32_t durationMs) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::RecordNotification(id=%d, durationMs=%d)", id,
- durationMs);
-
- // Not implement yet
-}
-
-void Channel::PlayFileEnded(int32_t id) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::PlayFileEnded(id=%d)", id);
-
- if (id == _inputFilePlayerId) {
- channel_state_.SetInputFilePlaying(false);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::PlayFileEnded() => input file player module is"
- " shutdown");
- } else if (id == _outputFilePlayerId) {
- channel_state_.SetOutputFilePlaying(false);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::PlayFileEnded() => output file player module is"
- " shutdown");
- }
-}
-
-void Channel::RecordFileEnded(int32_t id) {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::RecordFileEnded(id=%d)", id);
-
- assert(id == _outputFileRecorderId);
-
- rtc::CritScope cs(&_fileCritSect);
-
- _outputFileRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::RecordFileEnded() => output file recorder module is"
- " shutdown");
-}
-
Channel::Channel(int32_t channelId,
uint32_t instanceId,
const VoEBase::ChannelConfig& config)
@@ -874,12 +797,6 @@ Channel::Channel(int32_t channelId,
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
_outputAudioLevel(),
_externalTransport(false),
- // Avoid conflict with other channels by adding 1024 - 1026,
- // won't use as much as 1024 channels.
- _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
- _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
- _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
- _outputFileRecording(false),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
ntp_estimator_(Clock::GetRealTimeClock()),
@@ -899,7 +816,6 @@ Channel::Channel(int32_t channelId,
input_mute_(false),
previous_frame_muted_(false),
_outputGain(1.0f),
- _mixFileWithMicrophone(false),
_includeAudioLevelIndication(false),
transport_overhead_per_packet_(0),
rtp_overhead_per_packet_(0),
@@ -1088,22 +1004,6 @@ void Channel::Terminate() {
StopSend();
StopPlayout();
- {
- rtc::CritScope cs(&_fileCritSect);
- if (input_file_player_) {
- input_file_player_->RegisterModuleFileCallback(NULL);
- input_file_player_->StopPlayingFile();
- }
- if (output_file_player_) {
- output_file_player_->RegisterModuleFileCallback(NULL);
- output_file_player_->StopPlayingFile();
- }
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_->StopRecording();
- }
- }
-
// The order to safely shutdown modules in a channel is:
// 1. De-register callbacks in modules
// 2. De-register modules in process thread
@@ -1174,8 +1074,6 @@ int32_t Channel::StartPlayout() {
}
channel_state_.SetPlaying(true);
- if (RegisterFilePlayingToMixer() != 0)
- return -1;
return 0;
}
@@ -1823,508 +1721,6 @@ int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
return 0;
}
-int Channel::StartPlayingFileLocally(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartPlayingFileLocally(fileNameUTF8[]=%s, loop=%d,"
- " format=%d, volumeScaling=%5.3f, startPosition=%d, "
- "stopPosition=%d)",
- fileName, loop, format, volumeScaling, startPosition,
- stopPosition);
-
- if (channel_state_.Get().output_file_playing) {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceError,
- "StartPlayingFileLocally() is already playing");
- return -1;
- }
-
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (output_file_player_) {
- output_file_player_->RegisterModuleFileCallback(NULL);
- output_file_player_.reset();
- }
-
- output_file_player_ = FilePlayer::CreateFilePlayer(
- _outputFilePlayerId, (const FileFormats)format);
-
- if (!output_file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileLocally() filePlayer format is not correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (output_file_player_->StartPlayingFile(
- fileName, loop, startPosition, volumeScaling, notificationTime,
- stopPosition, (const CodecInst*)codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- output_file_player_->StopPlayingFile();
- output_file_player_.reset();
- return -1;
- }
- output_file_player_->RegisterModuleFileCallback(this);
- channel_state_.SetOutputFilePlaying(true);
- }
-
- if (RegisterFilePlayingToMixer() != 0)
- return -1;
-
- return 0;
-}
-
-int Channel::StartPlayingFileLocally(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartPlayingFileLocally(format=%d,"
- " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
- format, volumeScaling, startPosition, stopPosition);
-
- if (stream == NULL) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFileLocally() NULL as input stream");
- return -1;
- }
-
- if (channel_state_.Get().output_file_playing) {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceError,
- "StartPlayingFileLocally() is already playing");
- return -1;
- }
-
- {
- rtc::CritScope cs(&_fileCritSect);
-
- // Destroy the old instance
- if (output_file_player_) {
- output_file_player_->RegisterModuleFileCallback(NULL);
- output_file_player_.reset();
- }
-
- // Create the instance
- output_file_player_ = FilePlayer::CreateFilePlayer(
- _outputFilePlayerId, (const FileFormats)format);
-
- if (!output_file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileLocally() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (output_file_player_->StartPlayingFile(stream, startPosition,
- volumeScaling, notificationTime,
- stopPosition, codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to "
- "start file playout");
- output_file_player_->StopPlayingFile();
- output_file_player_.reset();
- return -1;
- }
- output_file_player_->RegisterModuleFileCallback(this);
- channel_state_.SetOutputFilePlaying(true);
- }
-
- if (RegisterFilePlayingToMixer() != 0)
- return -1;
-
- return 0;
-}
-
-int Channel::StopPlayingFileLocally() {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StopPlayingFileLocally()");
-
- if (!channel_state_.Get().output_file_playing) {
- return 0;
- }
-
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (output_file_player_->StopPlayingFile() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopPlayingFile() could not stop playing");
- return -1;
- }
- output_file_player_->RegisterModuleFileCallback(NULL);
- output_file_player_.reset();
- channel_state_.SetOutputFilePlaying(false);
- }
- // _fileCritSect cannot be taken while calling
- // SetAnonymousMixibilityStatus. Refer to comments in
- // StartPlayingFileLocally(const char* ...) for more details.
- if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, false) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StopPlayingFile() failed to stop participant from playing as"
- "file in the mixer");
- return -1;
- }
-
- return 0;
-}
-
-int Channel::IsPlayingFileLocally() const {
- return channel_state_.Get().output_file_playing;
-}
-
-int Channel::RegisterFilePlayingToMixer() {
- // Return success for not registering for file playing to mixer if:
- // 1. playing file before playout is started on that channel.
- // 2. starting playout without file playing on that channel.
- if (!channel_state_.Get().playing ||
- !channel_state_.Get().output_file_playing) {
- return 0;
- }
-
- // |_fileCritSect| cannot be taken while calling
- // SetAnonymousMixabilityStatus() since as soon as the participant is added
- // frames can be pulled by the mixer. Since the frames are generated from
- // the file, _fileCritSect will be taken. This would result in a deadlock.
- if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0) {
- channel_state_.SetOutputFilePlaying(false);
- rtc::CritScope cs(&_fileCritSect);
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StartPlayingFile() failed to add participant as file to mixer");
- output_file_player_->StopPlayingFile();
- output_file_player_.reset();
- return -1;
- }
-
- return 0;
-}
-
-int Channel::StartPlayingFileAsMicrophone(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartPlayingFileAsMicrophone(fileNameUTF8[]=%s, "
- "loop=%d, format=%d, volumeScaling=%5.3f, startPosition=%d, "
- "stopPosition=%d)",
- fileName, loop, format, volumeScaling, startPosition,
- stopPosition);
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (channel_state_.Get().input_file_playing) {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() filePlayer is playing");
- return 0;
- }
-
- // Destroy the old instance
- if (input_file_player_) {
- input_file_player_->RegisterModuleFileCallback(NULL);
- input_file_player_.reset();
- }
-
- // Create the instance
- input_file_player_ = FilePlayer::CreateFilePlayer(_inputFilePlayerId,
- (const FileFormats)format);
-
- if (!input_file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (input_file_player_->StartPlayingFile(
- fileName, loop, startPosition, volumeScaling, notificationTime,
- stopPosition, (const CodecInst*)codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- input_file_player_->StopPlayingFile();
- input_file_player_.reset();
- return -1;
- }
- input_file_player_->RegisterModuleFileCallback(this);
- channel_state_.SetInputFilePlaying(true);
-
- return 0;
-}
-
-int Channel::StartPlayingFileAsMicrophone(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartPlayingFileAsMicrophone(format=%d, "
- "volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
- format, volumeScaling, startPosition, stopPosition);
-
- if (stream == NULL) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFileAsMicrophone NULL as input stream");
- return -1;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (channel_state_.Get().input_file_playing) {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is playing");
- return 0;
- }
-
- // Destroy the old instance
- if (input_file_player_) {
- input_file_player_->RegisterModuleFileCallback(NULL);
- input_file_player_.reset();
- }
-
- // Create the instance
- input_file_player_ = FilePlayer::CreateFilePlayer(_inputFilePlayerId,
- (const FileFormats)format);
-
- if (!input_file_player_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingInputFile() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (input_file_player_->StartPlayingFile(stream, startPosition, volumeScaling,
- notificationTime, stopPosition,
- codecInst) != 0) {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start "
- "file playout");
- input_file_player_->StopPlayingFile();
- input_file_player_.reset();
- return -1;
- }
-
- input_file_player_->RegisterModuleFileCallback(this);
- channel_state_.SetInputFilePlaying(true);
-
- return 0;
-}
-
-int Channel::StopPlayingFileAsMicrophone() {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StopPlayingFileAsMicrophone()");
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (!channel_state_.Get().input_file_playing) {
- return 0;
- }
-
- if (input_file_player_->StopPlayingFile() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopPlayingFile() could not stop playing");
- return -1;
- }
- input_file_player_->RegisterModuleFileCallback(NULL);
- input_file_player_.reset();
- channel_state_.SetInputFilePlaying(false);
-
- return 0;
-}
-
-int Channel::IsPlayingFileAsMicrophone() const {
- return channel_state_.Get().input_file_playing;
-}
-
-int Channel::StartRecordingPlayout(const char* fileName,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartRecordingPlayout(fileName=%s)", fileName);
-
- if (_outputFileRecording) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000};
-
- if ((codecInst != NULL) &&
- ((codecInst->channels < 1) || (codecInst->channels > 2))) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return (-1);
- }
- if (codecInst == NULL) {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname, "PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname, "PCMA") == 0)) {
- format = kFileFormatWavFile;
- } else {
- format = kFileFormatCompressedFile;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- // Destroy the old instance
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- }
-
- output_file_recorder_ = FileRecorder::CreateFileRecorder(
- _outputFileRecorderId, (const FileFormats)format);
- if (!output_file_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (output_file_recorder_->StartRecordingAudioFile(
- fileName, (const CodecInst&)*codecInst, notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- output_file_recorder_->StopRecording();
- output_file_recorder_.reset();
- return -1;
- }
- output_file_recorder_->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int Channel::StartRecordingPlayout(OutStream* stream,
- const CodecInst* codecInst) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartRecordingPlayout()");
-
- if (_outputFileRecording) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const uint32_t notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000};
-
- if (codecInst != NULL && codecInst->channels != 1) {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return (-1);
- }
- if (codecInst == NULL) {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname, "PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname, "PCMA") == 0)) {
- format = kFileFormatWavFile;
- } else {
- format = kFileFormatCompressedFile;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- // Destroy the old instance
- if (output_file_recorder_) {
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- }
-
- output_file_recorder_ = FileRecorder::CreateFileRecorder(
- _outputFileRecorderId, (const FileFormats)format);
- if (!output_file_recorder_) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (output_file_recorder_->StartRecordingAudioFile(stream, *codecInst,
- notificationTime) != 0) {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartRecordingPlayout() failed to "
- "start file recording");
- output_file_recorder_->StopRecording();
- output_file_recorder_.reset();
- return -1;
- }
-
- output_file_recorder_->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int Channel::StopRecordingPlayout() {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "Channel::StopRecordingPlayout()");
-
- if (!_outputFileRecording) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StopRecordingPlayout() isnot recording");
- return -1;
- }
-
- rtc::CritScope cs(&_fileCritSect);
-
- if (output_file_recorder_->StopRecording() != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording() could not stop recording");
- return (-1);
- }
- output_file_recorder_->RegisterModuleFileCallback(NULL);
- output_file_recorder_.reset();
- _outputFileRecording = false;
-
- return 0;
-}
-
-void Channel::SetMixWithMicStatus(bool mix) {
- rtc::CritScope cs(&_fileCritSect);
- _mixFileWithMicrophone = mix;
-}
-
int Channel::GetSpeechOutputLevel() const {
return _outputAudioLevel.Level();
}
@@ -2760,11 +2156,7 @@ void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
RTC_DCHECK_LE(audio_input->num_channels_, 2);
RTC_DCHECK_EQ(audio_input->id_, ChannelId());
- if (channel_state_.Get().input_file_playing) {
- MixOrReplaceAudioWithFile(audio_input);
- }
-
- bool is_muted = InputMute();
+ bool is_muted = InputMute();
ossu 2017/09/14 13:33:39 Indentation looks off here. :(
AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
if (_includeAudioLevelIndication) {
@@ -2903,96 +2295,6 @@ int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule,
return 0;
}
-// TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
-// a shared helper.
-int32_t Channel::MixOrReplaceAudioWithFile(AudioFrame* audio_input) {
- RTC_DCHECK_RUN_ON(encoder_queue_);
- std::unique_ptr<int16_t[]> fileBuffer(new int16_t[640]);
- size_t fileSamples(0);
- const int mixingFrequency = audio_input->sample_rate_hz_;
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (!input_file_player_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() fileplayer"
- " doesnt exist");
- return -1;
- }
-
- if (input_file_player_->Get10msAudioFromFile(fileBuffer.get(), &fileSamples,
- mixingFrequency) == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() file mixing "
- "failed");
- return -1;
- }
- if (fileSamples == 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() file is ended");
- return 0;
- }
- }
-
- RTC_DCHECK_EQ(audio_input->samples_per_channel_, fileSamples);
-
- if (_mixFileWithMicrophone) {
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- MixWithSat(audio_input->mutable_data(), audio_input->num_channels_,
- fileBuffer.get(), 1, fileSamples);
- } else {
- // Replace ACM audio with file.
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- audio_input->UpdateFrame(
- _channelId, 0xFFFFFFFF, fileBuffer.get(), fileSamples, mixingFrequency,
- AudioFrame::kNormalSpeech, AudioFrame::kVadUnknown, 1);
- }
- return 0;
-}
-
-int32_t Channel::MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency) {
- assert(mixingFrequency <= 48000);
-
- std::unique_ptr<int16_t[]> fileBuffer(new int16_t[960]);
- size_t fileSamples(0);
-
- {
- rtc::CritScope cs(&_fileCritSect);
-
- if (!output_file_player_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixAudioWithFile() file mixing failed");
- return -1;
- }
-
- // We should get the frequency we ask for.
- if (output_file_player_->Get10msAudioFromFile(
- fileBuffer.get(), &fileSamples, mixingFrequency) == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixAudioWithFile() file mixing failed");
- return -1;
- }
- }
-
- if (audioFrame.samples_per_channel_ == fileSamples) {
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- MixWithSat(audioFrame.mutable_data(), audioFrame.num_channels_,
- fileBuffer.get(), 1, fileSamples);
- } else {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::MixAudioWithFile() samples_per_channel_(%" PRIuS
- ") != "
- "fileSamples(%" PRIuS ")",
- audioFrame.samples_per_channel_, fileSamples);
- return -1;
- }
-
- return 0;
-}
-
void Channel::UpdatePlayoutTimestamp(bool rtcp) {
jitter_buffer_playout_timestamp_ = audio_coding_->PlayoutTimestamp();
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