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a8ef811fe52add24f90973d19a31347226456abc
webrtc/webrtc/voice_engine/output_mixer.cc
andrew@webrtc.org ae1a58bba4 Replace AudioFrame's operator= with CopyFrom(). 
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders.

Review URL: https://webrtc-codereview.appspot.com/1031007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-22 04:44:30 +00:00

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "output_mixer.h"
#include "audio_processing.h"
#include "audio_frame_operations.h"
#include "critical_section_wrapper.h"
#include "file_wrapper.h"
#include "output_mixer_internal.h"
#include "statistics.h"
#include "trace.h"
#include "voe_external_media.h"
namespace webrtc {
namespace voe {
void
OutputMixer::NewMixedAudio(const WebRtc_Word32 id,
const AudioFrame& generalAudioFrame,
const AudioFrame** uniqueAudioFrames,
const WebRtc_UWord32 size)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::NewMixedAudio(id=%d, size=%u)", id, size);
_audioFrame.CopyFrom(generalAudioFrame);
_audioFrame.id_ = id;
}
void OutputMixer::MixedParticipants(
const WebRtc_Word32 id,
const ParticipantStatistics* participantStatistics,
const WebRtc_UWord32 size)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::MixedParticipants(id=%d, size=%u)", id, size);
}
void OutputMixer::VADPositiveParticipants(
const WebRtc_Word32 id,
const ParticipantStatistics* participantStatistics,
const WebRtc_UWord32 size)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::VADPositiveParticipants(id=%d, size=%u)",
id, size);
}
void OutputMixer::MixedAudioLevel(const WebRtc_Word32 id,
const WebRtc_UWord32 level)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::MixedAudioLevel(id=%d, level=%u)", id, level);
}
void OutputMixer::PlayNotification(const WebRtc_Word32 id,
const WebRtc_UWord32 durationMs)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::PlayNotification(id=%d, durationMs=%d)",
id, durationMs);
// Not implement yet
}
void OutputMixer::RecordNotification(const WebRtc_Word32 id,
const WebRtc_UWord32 durationMs)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::RecordNotification(id=%d, durationMs=%d)",
id, durationMs);
// Not implement yet
}
void OutputMixer::PlayFileEnded(const WebRtc_Word32 id)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::PlayFileEnded(id=%d)", id);
// not needed
}
void OutputMixer::RecordFileEnded(const WebRtc_Word32 id)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::RecordFileEnded(id=%d)", id);
assert(id == _instanceId);
CriticalSectionScoped cs(&_fileCritSect);
_outputFileRecording = false;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::RecordFileEnded() =>"
"output file recorder module is shutdown");
}
WebRtc_Word32
OutputMixer::Create(OutputMixer*& mixer, const WebRtc_UWord32 instanceId)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
"OutputMixer::Create(instanceId=%d)", instanceId);
mixer = new OutputMixer(instanceId);
if (mixer == NULL)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
"OutputMixer::Create() unable to allocate memory for"
"mixer");
return -1;
}
return 0;
}
OutputMixer::OutputMixer(const WebRtc_UWord32 instanceId) :
_callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_mixerModule(*AudioConferenceMixer::Create(instanceId)),
_audioLevel(),
_dtmfGenerator(instanceId),
_instanceId(instanceId),
_externalMediaCallbackPtr(NULL),
_externalMedia(false),
_panLeft(1.0f),
_panRight(1.0f),
_mixingFrequencyHz(8000),
_outputFileRecorderPtr(NULL),
_outputFileRecording(false)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::OutputMixer() - ctor");
if ((_mixerModule.RegisterMixedStreamCallback(*this) == -1) ||
(_mixerModule.RegisterMixerStatusCallback(*this, 100) == -1))
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::OutputMixer() failed to register mixer"
"callbacks");
}
_dtmfGenerator.Init();
}
void
OutputMixer::Destroy(OutputMixer*& mixer)
{
if (mixer)
{
delete mixer;
mixer = NULL;
}
}
OutputMixer::~OutputMixer()
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::~OutputMixer() - dtor");
if (_externalMedia)
{
DeRegisterExternalMediaProcessing();
}
{
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFileRecorderPtr)
{
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
_outputFileRecorderPtr->StopRecording();
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
}
}
_mixerModule.UnRegisterMixerStatusCallback();
_mixerModule.UnRegisterMixedStreamCallback();
delete &_mixerModule;
delete &_callbackCritSect;
delete &_fileCritSect;
}
WebRtc_Word32
OutputMixer::SetEngineInformation(voe::Statistics& engineStatistics)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::SetEngineInformation()");
_engineStatisticsPtr = &engineStatistics;
return 0;
}
WebRtc_Word32
OutputMixer::SetAudioProcessingModule(
AudioProcessing* audioProcessingModule)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::SetAudioProcessingModule("
"audioProcessingModule=0x%x)", audioProcessingModule);
_audioProcessingModulePtr = audioProcessingModule;
return 0;
}
int OutputMixer::RegisterExternalMediaProcessing(
VoEMediaProcess& proccess_object)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::RegisterExternalMediaProcessing()");
CriticalSectionScoped cs(&_callbackCritSect);
_externalMediaCallbackPtr = &proccess_object;
_externalMedia = true;
return 0;
}
int OutputMixer::DeRegisterExternalMediaProcessing()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::DeRegisterExternalMediaProcessing()");
CriticalSectionScoped cs(&_callbackCritSect);
_externalMedia = false;
_externalMediaCallbackPtr = NULL;
return 0;
}
int OutputMixer::PlayDtmfTone(WebRtc_UWord8 eventCode, int lengthMs,
int attenuationDb)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
"OutputMixer::PlayDtmfTone()");
if (_dtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb) != 0)
{
_engineStatisticsPtr->SetLastError(VE_STILL_PLAYING_PREV_DTMF,
kTraceError,
"OutputMixer::PlayDtmfTone()");
return -1;
}
return 0;
}
int OutputMixer::StartPlayingDtmfTone(WebRtc_UWord8 eventCode,
int attenuationDb)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
"OutputMixer::StartPlayingDtmfTone()");
if (_dtmfGenerator.StartTone(eventCode, attenuationDb) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_STILL_PLAYING_PREV_DTMF,
kTraceError,
"OutputMixer::StartPlayingDtmfTone())");
return -1;
}
return 0;
}
int OutputMixer::StopPlayingDtmfTone()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
"OutputMixer::StopPlayingDtmfTone()");
return (_dtmfGenerator.StopTone());
}
WebRtc_Word32
OutputMixer::SetMixabilityStatus(MixerParticipant& participant,
const bool mixable)
{
return _mixerModule.SetMixabilityStatus(participant, mixable);
}
WebRtc_Word32
OutputMixer::SetAnonymousMixabilityStatus(MixerParticipant& participant,
const bool mixable)
{
return _mixerModule.SetAnonymousMixabilityStatus(participant,mixable);
}
WebRtc_Word32
OutputMixer::MixActiveChannels()
{
return _mixerModule.Process();
}
int
OutputMixer::GetSpeechOutputLevel(WebRtc_UWord32& level)
{
WebRtc_Word8 currentLevel = _audioLevel.Level();
level = static_cast<WebRtc_UWord32> (currentLevel);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
"GetSpeechOutputLevel() => level=%u", level);
return 0;
}
int
OutputMixer::GetSpeechOutputLevelFullRange(WebRtc_UWord32& level)
{
WebRtc_Word16 currentLevel = _audioLevel.LevelFullRange();
level = static_cast<WebRtc_UWord32> (currentLevel);
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
"GetSpeechOutputLevelFullRange() => level=%u", level);
return 0;
}
int
OutputMixer::SetOutputVolumePan(float left, float right)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::SetOutputVolumePan()");
_panLeft = left;
_panRight = right;
return 0;
}
int
OutputMixer::GetOutputVolumePan(float& left, float& right)
{
left = _panLeft;
right = _panRight;
WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
"GetOutputVolumePan() => left=%2.1f, right=%2.1f",
left, right);
return 0;
}
int OutputMixer::StartRecordingPlayout(const char* fileName,
const CodecInst* codecInst)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::StartRecordingPlayout(fileName=%s)", fileName);
if (_outputFileRecording)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
"StartRecordingPlayout() is already recording");
return 0;
}
FileFormats format;
const WebRtc_UWord32 notificationTime(0);
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;
}
CriticalSectionScoped cs(&_fileCritSect);
// Destroy the old instance
if (_outputFileRecorderPtr)
{
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
}
_outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
_instanceId,
(const FileFormats)format);
if (_outputFileRecorderPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartRecordingPlayout() fileRecorder format isnot correct");
return -1;
}
if (_outputFileRecorderPtr->StartRecordingAudioFile(
fileName,
(const CodecInst&)*codecInst,
notificationTime) != 0)
{
_engineStatisticsPtr->SetLastError(
VE_BAD_FILE, kTraceError,
"StartRecordingAudioFile() failed to start file recording");
_outputFileRecorderPtr->StopRecording();
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
return -1;
}
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
_outputFileRecording = true;
return 0;
}
int OutputMixer::StartRecordingPlayout(OutStream* stream,
const CodecInst* codecInst)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::StartRecordingPlayout()");
if (_outputFileRecording)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
"StartRecordingPlayout() is already recording");
return 0;
}
FileFormats format;
const WebRtc_UWord32 notificationTime(0);
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;
}
CriticalSectionScoped cs(&_fileCritSect);
// Destroy the old instance
if (_outputFileRecorderPtr)
{
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
}
_outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
_instanceId,
(const FileFormats)format);
if (_outputFileRecorderPtr == NULL)
{
_engineStatisticsPtr->SetLastError(
VE_INVALID_ARGUMENT, kTraceError,
"StartRecordingPlayout() fileRecorder format isnot correct");
return -1;
}
if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream,
*codecInst,
notificationTime) != 0)
{
_engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
"StartRecordingAudioFile() failed to start file recording");
_outputFileRecorderPtr->StopRecording();
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
return -1;
}
_outputFileRecorderPtr->RegisterModuleFileCallback(this);
_outputFileRecording = true;
return 0;
}
int OutputMixer::StopRecordingPlayout()
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::StopRecordingPlayout()");
if (!_outputFileRecording)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
"StopRecordingPlayout() file isnot recording");
return -1;
}
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFileRecorderPtr->StopRecording() != 0)
{
_engineStatisticsPtr->SetLastError(
VE_STOP_RECORDING_FAILED, kTraceError,
"StopRecording(), could not stop recording");
return -1;
}
_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
_outputFileRecorderPtr = NULL;
_outputFileRecording = false;
return 0;
}
int OutputMixer::GetMixedAudio(int sample_rate_hz,
int num_channels,
AudioFrame* frame) {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%d)",
sample_rate_hz, num_channels);
// --- Record playout if enabled
{
CriticalSectionScoped cs(&_fileCritSect);
if (_outputFileRecording && _outputFileRecorderPtr)
_outputFileRecorderPtr->RecordAudioToFile(_audioFrame);
}
frame->num_channels_ = num_channels;
frame->sample_rate_hz_ = sample_rate_hz;
// TODO(andrew): Ideally the downmixing would occur much earlier, in
// AudioCodingModule.
return RemixAndResample(_audioFrame, &_resampler, frame);
}
WebRtc_Word32
OutputMixer::DoOperationsOnCombinedSignal()
{
if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
"OutputMixer::DoOperationsOnCombinedSignal() => "
"mixing frequency = %d", _audioFrame.sample_rate_hz_);
_mixingFrequencyHz = _audioFrame.sample_rate_hz_;
}
// --- Insert inband Dtmf tone
if (_dtmfGenerator.IsAddingTone())
{
InsertInbandDtmfTone();
}
// Scale left and/or right channel(s) if balance is active
if (_panLeft != 1.0 || _panRight != 1.0)
{
if (_audioFrame.num_channels_ == 1)
{
AudioFrameOperations::MonoToStereo(&_audioFrame);
}
else
{
// Pure stereo mode (we are receiving a stereo signal).
}
assert(_audioFrame.num_channels_ == 2);
AudioFrameOperations::Scale(_panLeft, _panRight, _audioFrame);
}
// --- Far-end Voice Quality Enhancement (AudioProcessing Module)
APMAnalyzeReverseStream();
// --- External media processing
if (_externalMedia)
{
CriticalSectionScoped cs(&_callbackCritSect);
const bool isStereo = (_audioFrame.num_channels_ == 2);
if (_externalMediaCallbackPtr)
{
_externalMediaCallbackPtr->Process(
-1,
kPlaybackAllChannelsMixed,
(WebRtc_Word16*)_audioFrame.data_,
_audioFrame.samples_per_channel_,
_audioFrame.sample_rate_hz_,
isStereo);
}
}
// --- Measure audio level (0-9) for the combined signal
_audioLevel.ComputeLevel(_audioFrame);
return 0;
}
// ----------------------------------------------------------------------------
// Private methods
// ----------------------------------------------------------------------------
void OutputMixer::APMAnalyzeReverseStream() {
// Convert from mixing to AudioProcessing sample rate, determined by the send
// side. Downmix to mono.
AudioFrame frame;
frame.num_channels_ = 1;
frame.sample_rate_hz_ = _audioProcessingModulePtr->sample_rate_hz();
if (RemixAndResample(_audioFrame, &_apmResampler, &frame) == -1)
return;
if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) == -1) {
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
"AudioProcessingModule::AnalyzeReverseStream() => error");
}
}
int
OutputMixer::InsertInbandDtmfTone()
{
WebRtc_UWord16 sampleRate(0);
_dtmfGenerator.GetSampleRate(sampleRate);
if (sampleRate != _audioFrame.sample_rate_hz_)
{
// Update sample rate of Dtmf tone since the mixing frequency changed.
_dtmfGenerator.SetSampleRate(
(WebRtc_UWord16)(_audioFrame.sample_rate_hz_));
// Reset the tone to be added taking the new sample rate into account.
_dtmfGenerator.ResetTone();
}
WebRtc_Word16 toneBuffer[320];
WebRtc_UWord16 toneSamples(0);
if (_dtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"OutputMixer::InsertInbandDtmfTone() inserting Dtmf"
"tone failed");
return -1;
}
// replace mixed audio with Dtmf tone
if (_audioFrame.num_channels_ == 1)
{
// mono
memcpy(_audioFrame.data_, toneBuffer, sizeof(WebRtc_Word16)
* toneSamples);
} else
{
// stereo
for (int i = 0; i < _audioFrame.samples_per_channel_; i++)
{
_audioFrame.data_[2 * i] = toneBuffer[i];
_audioFrame.data_[2 * i + 1] = 0;
}
}
assert(_audioFrame.samples_per_channel_ == toneSamples);
return 0;
}
} // namespace voe
} // namespace webrtc
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