| /* |
| * 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 "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h" |
| #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h" |
| #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "webrtc/modules/utility/interface/audio_frame_operations.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| struct ParticipantFramePair { |
| MixerParticipant* participant; |
| AudioFrame* audioFrame; |
| }; |
| |
| typedef std::list<ParticipantFramePair*> ParticipantFramePairList; |
| |
| // Mix |frame| into |mixed_frame|, with saturation protection and upmixing. |
| // These effects are applied to |frame| itself prior to mixing. Assumes that |
| // |mixed_frame| always has at least as many channels as |frame|. Supports |
| // stereo at most. |
| // |
| // TODO(andrew): consider not modifying |frame| here. |
| void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) { |
| assert(mixed_frame->num_channels_ >= frame->num_channels_); |
| if (use_limiter) { |
| // Divide by two to avoid saturation in the mixing. |
| // This is only meaningful if the limiter will be used. |
| *frame >>= 1; |
| } |
| if (mixed_frame->num_channels_ > frame->num_channels_) { |
| // We only support mono-to-stereo. |
| assert(mixed_frame->num_channels_ == 2 && |
| frame->num_channels_ == 1); |
| AudioFrameOperations::MonoToStereo(frame); |
| } |
| |
| *mixed_frame += *frame; |
| } |
| |
| // Return the max number of channels from a |list| composed of AudioFrames. |
| int MaxNumChannels(const AudioFrameList* list) { |
| int max_num_channels = 1; |
| for (AudioFrameList::const_iterator iter = list->begin(); |
| iter != list->end(); |
| ++iter) { |
| max_num_channels = std::max(max_num_channels, (*iter)->num_channels_); |
| } |
| return max_num_channels; |
| } |
| |
| } // namespace |
| |
| MixerParticipant::MixerParticipant() |
| : _mixHistory(new MixHistory()) { |
| } |
| |
| MixerParticipant::~MixerParticipant() { |
| delete _mixHistory; |
| } |
| |
| int32_t MixerParticipant::IsMixed(bool& mixed) const { |
| return _mixHistory->IsMixed(mixed); |
| } |
| |
| MixHistory::MixHistory() |
| : _isMixed(0) { |
| } |
| |
| MixHistory::~MixHistory() { |
| } |
| |
| int32_t MixHistory::IsMixed(bool& mixed) const { |
| mixed = _isMixed; |
| return 0; |
| } |
| |
| int32_t MixHistory::WasMixed(bool& wasMixed) const { |
| // Was mixed is the same as is mixed depending on perspective. This function |
| // is for the perspective of AudioConferenceMixerImpl. |
| return IsMixed(wasMixed); |
| } |
| |
| int32_t MixHistory::SetIsMixed(const bool mixed) { |
| _isMixed = mixed; |
| return 0; |
| } |
| |
| void MixHistory::ResetMixedStatus() { |
| _isMixed = false; |
| } |
| |
| AudioConferenceMixer* AudioConferenceMixer::Create(int id) { |
| AudioConferenceMixerImpl* mixer = new AudioConferenceMixerImpl(id); |
| if(!mixer->Init()) { |
| delete mixer; |
| return NULL; |
| } |
| return mixer; |
| } |
| |
| AudioConferenceMixerImpl::AudioConferenceMixerImpl(int id) |
| : _id(id), |
| _minimumMixingFreq(kLowestPossible), |
| _mixReceiver(NULL), |
| _outputFrequency(kDefaultFrequency), |
| _sampleSize(0), |
| _audioFramePool(NULL), |
| _participantList(), |
| _additionalParticipantList(), |
| _numMixedParticipants(0), |
| use_limiter_(true), |
| _timeStamp(0), |
| _timeScheduler(kProcessPeriodicityInMs), |
| _processCalls(0) {} |
| |
| bool AudioConferenceMixerImpl::Init() { |
| _crit.reset(CriticalSectionWrapper::CreateCriticalSection()); |
| if (_crit.get() == NULL) |
| return false; |
| |
| _cbCrit.reset(CriticalSectionWrapper::CreateCriticalSection()); |
| if(_cbCrit.get() == NULL) |
| return false; |
| |
| Config config; |
| config.Set<ExperimentalAgc>(new ExperimentalAgc(false)); |
| _limiter.reset(AudioProcessing::Create(config)); |
| if(!_limiter.get()) |
| return false; |
| |
| MemoryPool<AudioFrame>::CreateMemoryPool(_audioFramePool, |
| DEFAULT_AUDIO_FRAME_POOLSIZE); |
| if(_audioFramePool == NULL) |
| return false; |
| |
| if(SetOutputFrequency(kDefaultFrequency) == -1) |
| return false; |
| |
| if(_limiter->gain_control()->set_mode(GainControl::kFixedDigital) != |
| _limiter->kNoError) |
| return false; |
| |
| // We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the |
| // divide-by-2 but -7 is used instead to give a bit of headroom since the |
| // AGC is not a hard limiter. |
| if(_limiter->gain_control()->set_target_level_dbfs(7) != _limiter->kNoError) |
| return false; |
| |
| if(_limiter->gain_control()->set_compression_gain_db(0) |
| != _limiter->kNoError) |
| return false; |
| |
| if(_limiter->gain_control()->enable_limiter(true) != _limiter->kNoError) |
| return false; |
| |
| if(_limiter->gain_control()->Enable(true) != _limiter->kNoError) |
| return false; |
| |
| return true; |
| } |
| |
| AudioConferenceMixerImpl::~AudioConferenceMixerImpl() { |
| MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool); |
| assert(_audioFramePool == NULL); |
| } |
| |
| // Process should be called every kProcessPeriodicityInMs ms |
| int64_t AudioConferenceMixerImpl::TimeUntilNextProcess() { |
| int64_t timeUntilNextProcess = 0; |
| CriticalSectionScoped cs(_crit.get()); |
| if(_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0) { |
| WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, |
| "failed in TimeToNextUpdate() call"); |
| // Sanity check |
| assert(false); |
| return -1; |
| } |
| return timeUntilNextProcess; |
| } |
| |
| int32_t AudioConferenceMixerImpl::Process() { |
| size_t remainingParticipantsAllowedToMix = |
| kMaximumAmountOfMixedParticipants; |
| { |
| CriticalSectionScoped cs(_crit.get()); |
| assert(_processCalls == 0); |
| _processCalls++; |
| |
| // Let the scheduler know that we are running one iteration. |
| _timeScheduler.UpdateScheduler(); |
| } |
| |
| AudioFrameList mixList; |
| AudioFrameList rampOutList; |
| AudioFrameList additionalFramesList; |
| std::map<int, MixerParticipant*> mixedParticipantsMap; |
| { |
| CriticalSectionScoped cs(_cbCrit.get()); |
| |
| int32_t lowFreq = GetLowestMixingFrequency(); |
| // SILK can run in 12 kHz and 24 kHz. These frequencies are not |
| // supported so use the closest higher frequency to not lose any |
| // information. |
| // TODO(henrike): this is probably more appropriate to do in |
| // GetLowestMixingFrequency(). |
| if (lowFreq == 12000) { |
| lowFreq = 16000; |
| } else if (lowFreq == 24000) { |
| lowFreq = 32000; |
| } |
| if(lowFreq <= 0) { |
| CriticalSectionScoped cs(_crit.get()); |
| _processCalls--; |
| return 0; |
| } else { |
| switch(lowFreq) { |
| case 8000: |
| if(OutputFrequency() != kNbInHz) { |
| SetOutputFrequency(kNbInHz); |
| } |
| break; |
| case 16000: |
| if(OutputFrequency() != kWbInHz) { |
| SetOutputFrequency(kWbInHz); |
| } |
| break; |
| case 32000: |
| if(OutputFrequency() != kSwbInHz) { |
| SetOutputFrequency(kSwbInHz); |
| } |
| break; |
| case 48000: |
| if(OutputFrequency() != kFbInHz) { |
| SetOutputFrequency(kFbInHz); |
| } |
| break; |
| default: |
| assert(false); |
| |
| CriticalSectionScoped cs(_crit.get()); |
| _processCalls--; |
| return -1; |
| } |
| } |
| |
| UpdateToMix(&mixList, &rampOutList, &mixedParticipantsMap, |
| remainingParticipantsAllowedToMix); |
| |
| GetAdditionalAudio(&additionalFramesList); |
| UpdateMixedStatus(mixedParticipantsMap); |
| } |
| |
| // Get an AudioFrame for mixing from the memory pool. |
| AudioFrame* mixedAudio = NULL; |
| if(_audioFramePool->PopMemory(mixedAudio) == -1) { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id, |
| "failed PopMemory() call"); |
| assert(false); |
| return -1; |
| } |
| |
| int retval = 0; |
| { |
| CriticalSectionScoped cs(_crit.get()); |
| |
| // TODO(henrike): it might be better to decide the number of channels |
| // with an API instead of dynamically. |
| |
| // Find the max channels over all mixing lists. |
| const int num_mixed_channels = std::max(MaxNumChannels(&mixList), |
| std::max(MaxNumChannels(&additionalFramesList), |
| MaxNumChannels(&rampOutList))); |
| |
| mixedAudio->UpdateFrame(-1, _timeStamp, NULL, 0, _outputFrequency, |
| AudioFrame::kNormalSpeech, |
| AudioFrame::kVadPassive, num_mixed_channels); |
| |
| _timeStamp += _sampleSize; |
| |
| // We only use the limiter if it supports the output sample rate and |
| // we're actually mixing multiple streams. |
| use_limiter_ = _numMixedParticipants > 1 && |
| _outputFrequency <= kAudioProcMaxNativeSampleRateHz; |
| |
| MixFromList(*mixedAudio, &mixList); |
| MixAnonomouslyFromList(*mixedAudio, &additionalFramesList); |
| MixAnonomouslyFromList(*mixedAudio, &rampOutList); |
| |
| if(mixedAudio->samples_per_channel_ == 0) { |
| // Nothing was mixed, set the audio samples to silence. |
| mixedAudio->samples_per_channel_ = _sampleSize; |
| mixedAudio->Mute(); |
| } else { |
| // Only call the limiter if we have something to mix. |
| if(!LimitMixedAudio(*mixedAudio)) |
| retval = -1; |
| } |
| } |
| |
| { |
| CriticalSectionScoped cs(_cbCrit.get()); |
| if(_mixReceiver != NULL) { |
| const AudioFrame** dummy = NULL; |
| _mixReceiver->NewMixedAudio( |
| _id, |
| *mixedAudio, |
| dummy, |
| 0); |
| } |
| } |
| |
| // Reclaim all outstanding memory. |
| _audioFramePool->PushMemory(mixedAudio); |
| ClearAudioFrameList(&mixList); |
| ClearAudioFrameList(&rampOutList); |
| ClearAudioFrameList(&additionalFramesList); |
| { |
| CriticalSectionScoped cs(_crit.get()); |
| _processCalls--; |
| } |
| return retval; |
| } |
| |
| int32_t AudioConferenceMixerImpl::RegisterMixedStreamCallback( |
| AudioMixerOutputReceiver& mixReceiver) { |
| CriticalSectionScoped cs(_cbCrit.get()); |
| if(_mixReceiver != NULL) { |
| return -1; |
| } |
| _mixReceiver = &mixReceiver; |
| return 0; |
| } |
| |
| int32_t AudioConferenceMixerImpl::UnRegisterMixedStreamCallback() { |
| CriticalSectionScoped cs(_cbCrit.get()); |
| if(_mixReceiver == NULL) { |
| return -1; |
| } |
| _mixReceiver = NULL; |
| return 0; |
| } |
| |
| int32_t AudioConferenceMixerImpl::SetOutputFrequency( |
| const Frequency frequency) { |
| CriticalSectionScoped cs(_crit.get()); |
| |
| _outputFrequency = frequency; |
| _sampleSize = (_outputFrequency*kProcessPeriodicityInMs) / 1000; |
| |
| return 0; |
| } |
| |
| AudioConferenceMixer::Frequency |
| AudioConferenceMixerImpl::OutputFrequency() const { |
| CriticalSectionScoped cs(_crit.get()); |
| return _outputFrequency; |
| } |
| |
| int32_t AudioConferenceMixerImpl::SetMixabilityStatus( |
| MixerParticipant& participant, |
| bool mixable) { |
| if (!mixable) { |
| // Anonymous participants are in a separate list. Make sure that the |
| // participant is in the _participantList if it is being mixed. |
| SetAnonymousMixabilityStatus(participant, false); |
| } |
| size_t numMixedParticipants; |
| { |
| CriticalSectionScoped cs(_cbCrit.get()); |
| const bool isMixed = |
| IsParticipantInList(participant, &_participantList); |
| // API must be called with a new state. |
| if(!(mixable ^ isMixed)) { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, |
| "Mixable is aready %s", |
| isMixed ? "ON" : "off"); |
| return -1; |
| } |
| bool success = false; |
| if(mixable) { |
| success = AddParticipantToList(participant, &_participantList); |
| } else { |
| success = RemoveParticipantFromList(participant, &_participantList); |
| } |
| if(!success) { |
| WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, |
| "failed to %s participant", |
| mixable ? "add" : "remove"); |
| assert(false); |
| return -1; |
| } |
| |
| size_t numMixedNonAnonymous = _participantList.size(); |
| if (numMixedNonAnonymous > kMaximumAmountOfMixedParticipants) { |
| numMixedNonAnonymous = kMaximumAmountOfMixedParticipants; |
| } |
| numMixedParticipants = |
| numMixedNonAnonymous + _additionalParticipantList.size(); |
| } |
| // A MixerParticipant was added or removed. Make sure the scratch |
| // buffer is updated if necessary. |
| // Note: The scratch buffer may only be updated in Process(). |
| CriticalSectionScoped cs(_crit.get()); |
| _numMixedParticipants = numMixedParticipants; |
| return 0; |
| } |
| |
| int32_t AudioConferenceMixerImpl::MixabilityStatus( |
| MixerParticipant& participant, |
| bool& mixable) { |
| CriticalSectionScoped cs(_cbCrit.get()); |
| mixable = IsParticipantInList(participant, &_participantList); |
| return 0; |
| } |
| |
| int32_t AudioConferenceMixerImpl::SetAnonymousMixabilityStatus( |
| MixerParticipant& participant, const bool anonymous) { |
| CriticalSectionScoped cs(_cbCrit.get()); |
| if(IsParticipantInList(participant, &_additionalParticipantList)) { |
| if(anonymous) { |
| return 0; |
| } |
| if(!RemoveParticipantFromList(participant, |
| &_additionalParticipantList)) { |
| WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, |
| "unable to remove participant from anonymous list"); |
| assert(false); |
| return -1; |
| } |
| return AddParticipantToList(participant, &_participantList) ? 0 : -1; |
| } |
| if(!anonymous) { |
| return 0; |
| } |
| const bool mixable = RemoveParticipantFromList(participant, |
| &_participantList); |
| if(!mixable) { |
| WEBRTC_TRACE( |
| kTraceWarning, |
| kTraceAudioMixerServer, |
| _id, |
| "participant must be registered before turning it into anonymous"); |
| // Setting anonymous status is only possible if MixerParticipant is |
| // already registered. |
| return -1; |
| } |
| return AddParticipantToList(participant, &_additionalParticipantList) ? |
| 0 : -1; |
| } |
| |
| int32_t AudioConferenceMixerImpl::AnonymousMixabilityStatus( |
| MixerParticipant& participant, bool& mixable) { |
| CriticalSectionScoped cs(_cbCrit.get()); |
| mixable = IsParticipantInList(participant, |
| &_additionalParticipantList); |
| return 0; |
| } |
| |
| int32_t AudioConferenceMixerImpl::SetMinimumMixingFrequency( |
| Frequency freq) { |
| // Make sure that only allowed sampling frequencies are used. Use closest |
| // higher sampling frequency to avoid losing information. |
| if (static_cast<int>(freq) == 12000) { |
| freq = kWbInHz; |
| } else if (static_cast<int>(freq) == 24000) { |
| freq = kSwbInHz; |
| } |
| |
| if((freq == kNbInHz) || (freq == kWbInHz) || (freq == kSwbInHz) || |
| (freq == kLowestPossible)) { |
| _minimumMixingFreq=freq; |
| return 0; |
| } else { |
| WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, |
| "SetMinimumMixingFrequency incorrect frequency: %i",freq); |
| assert(false); |
| return -1; |
| } |
| } |
| |
| // Check all AudioFrames that are to be mixed. The highest sampling frequency |
| // found is the lowest that can be used without losing information. |
| int32_t AudioConferenceMixerImpl::GetLowestMixingFrequency() { |
| const int participantListFrequency = |
| GetLowestMixingFrequencyFromList(&_participantList); |
| const int anonymousListFrequency = |
| GetLowestMixingFrequencyFromList(&_additionalParticipantList); |
| const int highestFreq = |
| (participantListFrequency > anonymousListFrequency) ? |
| participantListFrequency : anonymousListFrequency; |
| // Check if the user specified a lowest mixing frequency. |
| if(_minimumMixingFreq != kLowestPossible) { |
| if(_minimumMixingFreq > highestFreq) { |
| return _minimumMixingFreq; |
| } |
| } |
| return highestFreq; |
| } |
| |
| int32_t AudioConferenceMixerImpl::GetLowestMixingFrequencyFromList( |
| MixerParticipantList* mixList) { |
| int32_t highestFreq = 8000; |
| for (MixerParticipantList::iterator iter = mixList->begin(); |
| iter != mixList->end(); |
| ++iter) { |
| const int32_t neededFrequency = (*iter)->NeededFrequency(_id); |
| if(neededFrequency > highestFreq) { |
| highestFreq = neededFrequency; |
| } |
| } |
| return highestFreq; |
| } |
| |
| void AudioConferenceMixerImpl::UpdateToMix( |
| AudioFrameList* mixList, |
| AudioFrameList* rampOutList, |
| std::map<int, MixerParticipant*>* mixParticipantList, |
| size_t& maxAudioFrameCounter) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "UpdateToMix(mixList,rampOutList,mixParticipantList,%d)", |
| maxAudioFrameCounter); |
| const size_t mixListStartSize = mixList->size(); |
| AudioFrameList activeList; |
| // Struct needed by the passive lists to keep track of which AudioFrame |
| // belongs to which MixerParticipant. |
| ParticipantFramePairList passiveWasNotMixedList; |
| ParticipantFramePairList passiveWasMixedList; |
| for (MixerParticipantList::iterator participant = _participantList.begin(); |
| participant != _participantList.end(); |
| ++participant) { |
| // Stop keeping track of passive participants if there are already |
| // enough participants available (they wont be mixed anyway). |
| bool mustAddToPassiveList = (maxAudioFrameCounter > |
| (activeList.size() + |
| passiveWasMixedList.size() + |
| passiveWasNotMixedList.size())); |
| |
| bool wasMixed = false; |
| (*participant)->_mixHistory->WasMixed(wasMixed); |
| AudioFrame* audioFrame = NULL; |
| if(_audioFramePool->PopMemory(audioFrame) == -1) { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id, |
| "failed PopMemory() call"); |
| assert(false); |
| return; |
| } |
| audioFrame->sample_rate_hz_ = _outputFrequency; |
| |
| if((*participant)->GetAudioFrame(_id,*audioFrame) != 0) { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, |
| "failed to GetAudioFrame() from participant"); |
| _audioFramePool->PushMemory(audioFrame); |
| continue; |
| } |
| if (_participantList.size() != 1) { |
| // TODO(wu): Issue 3390, add support for multiple participants case. |
| audioFrame->ntp_time_ms_ = -1; |
| } |
| |
| // TODO(henrike): this assert triggers in some test cases where SRTP is |
| // used which prevents NetEQ from making a VAD. Temporarily disable this |
| // assert until the problem is fixed on a higher level. |
| // assert(audioFrame->vad_activity_ != AudioFrame::kVadUnknown); |
| if (audioFrame->vad_activity_ == AudioFrame::kVadUnknown) { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, |
| "invalid VAD state from participant"); |
| } |
| |
| if(audioFrame->vad_activity_ == AudioFrame::kVadActive) { |
| if(!wasMixed) { |
| RampIn(*audioFrame); |
| } |
| |
| if(activeList.size() >= maxAudioFrameCounter) { |
| // There are already more active participants than should be |
| // mixed. Only keep the ones with the highest energy. |
| AudioFrameList::iterator replaceItem; |
| CalculateEnergy(*audioFrame); |
| uint32_t lowestEnergy = audioFrame->energy_; |
| |
| bool found_replace_item = false; |
| for (AudioFrameList::iterator iter = activeList.begin(); |
| iter != activeList.end(); |
| ++iter) { |
| CalculateEnergy(**iter); |
| if((*iter)->energy_ < lowestEnergy) { |
| replaceItem = iter; |
| lowestEnergy = (*iter)->energy_; |
| found_replace_item = true; |
| } |
| } |
| if(found_replace_item) { |
| AudioFrame* replaceFrame = *replaceItem; |
| |
| bool replaceWasMixed = false; |
| std::map<int, MixerParticipant*>::iterator it = |
| mixParticipantList->find(replaceFrame->id_); |
| |
| // When a frame is pushed to |activeList| it is also pushed |
| // to mixParticipantList with the frame's id. This means |
| // that the Find call above should never fail. |
| assert(it != mixParticipantList->end()); |
| it->second->_mixHistory->WasMixed(replaceWasMixed); |
| |
| mixParticipantList->erase(replaceFrame->id_); |
| activeList.erase(replaceItem); |
| |
| activeList.push_front(audioFrame); |
| (*mixParticipantList)[audioFrame->id_] = *participant; |
| assert(mixParticipantList->size() <= |
| kMaximumAmountOfMixedParticipants); |
| |
| if (replaceWasMixed) { |
| RampOut(*replaceFrame); |
| rampOutList->push_back(replaceFrame); |
| assert(rampOutList->size() <= |
| kMaximumAmountOfMixedParticipants); |
| } else { |
| _audioFramePool->PushMemory(replaceFrame); |
| } |
| } else { |
| if(wasMixed) { |
| RampOut(*audioFrame); |
| rampOutList->push_back(audioFrame); |
| assert(rampOutList->size() <= |
| kMaximumAmountOfMixedParticipants); |
| } else { |
| _audioFramePool->PushMemory(audioFrame); |
| } |
| } |
| } else { |
| activeList.push_front(audioFrame); |
| (*mixParticipantList)[audioFrame->id_] = *participant; |
| assert(mixParticipantList->size() <= |
| kMaximumAmountOfMixedParticipants); |
| } |
| } else { |
| if(wasMixed) { |
| ParticipantFramePair* pair = new ParticipantFramePair; |
| pair->audioFrame = audioFrame; |
| pair->participant = *participant; |
| passiveWasMixedList.push_back(pair); |
| } else if(mustAddToPassiveList) { |
| RampIn(*audioFrame); |
| ParticipantFramePair* pair = new ParticipantFramePair; |
| pair->audioFrame = audioFrame; |
| pair->participant = *participant; |
| passiveWasNotMixedList.push_back(pair); |
| } else { |
| _audioFramePool->PushMemory(audioFrame); |
| } |
| } |
| } |
| assert(activeList.size() <= maxAudioFrameCounter); |
| // At this point it is known which participants should be mixed. Transfer |
| // this information to this functions output parameters. |
| for (AudioFrameList::iterator iter = activeList.begin(); |
| iter != activeList.end(); |
| ++iter) { |
| mixList->push_back(*iter); |
| } |
| activeList.clear(); |
| // Always mix a constant number of AudioFrames. If there aren't enough |
| // active participants mix passive ones. Starting with those that was mixed |
| // last iteration. |
| for (ParticipantFramePairList::iterator iter = passiveWasMixedList.begin(); |
| iter != passiveWasMixedList.end(); |
| ++iter) { |
| if(mixList->size() < maxAudioFrameCounter + mixListStartSize) { |
| mixList->push_back((*iter)->audioFrame); |
| (*mixParticipantList)[(*iter)->audioFrame->id_] = |
| (*iter)->participant; |
| assert(mixParticipantList->size() <= |
| kMaximumAmountOfMixedParticipants); |
| } else { |
| _audioFramePool->PushMemory((*iter)->audioFrame); |
| } |
| delete *iter; |
| } |
| // And finally the ones that have not been mixed for a while. |
| for (ParticipantFramePairList::iterator iter = |
| passiveWasNotMixedList.begin(); |
| iter != passiveWasNotMixedList.end(); |
| ++iter) { |
| if(mixList->size() < maxAudioFrameCounter + mixListStartSize) { |
| mixList->push_back((*iter)->audioFrame); |
| (*mixParticipantList)[(*iter)->audioFrame->id_] = |
| (*iter)->participant; |
| assert(mixParticipantList->size() <= |
| kMaximumAmountOfMixedParticipants); |
| } else { |
| _audioFramePool->PushMemory((*iter)->audioFrame); |
| } |
| delete *iter; |
| } |
| assert(maxAudioFrameCounter + mixListStartSize >= mixList->size()); |
| maxAudioFrameCounter += mixListStartSize - mixList->size(); |
| } |
| |
| void AudioConferenceMixerImpl::GetAdditionalAudio( |
| AudioFrameList* additionalFramesList) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "GetAdditionalAudio(additionalFramesList)"); |
| // The GetAudioFrame() callback may result in the participant being removed |
| // from additionalParticipantList_. If that happens it will invalidate any |
| // iterators. Create a copy of the participants list such that the list of |
| // participants can be traversed safely. |
| MixerParticipantList additionalParticipantList; |
| additionalParticipantList.insert(additionalParticipantList.begin(), |
| _additionalParticipantList.begin(), |
| _additionalParticipantList.end()); |
| |
| for (MixerParticipantList::iterator participant = |
| additionalParticipantList.begin(); |
| participant != additionalParticipantList.end(); |
| ++participant) { |
| AudioFrame* audioFrame = NULL; |
| if(_audioFramePool->PopMemory(audioFrame) == -1) { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id, |
| "failed PopMemory() call"); |
| assert(false); |
| return; |
| } |
| audioFrame->sample_rate_hz_ = _outputFrequency; |
| if((*participant)->GetAudioFrame(_id, *audioFrame) != 0) { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id, |
| "failed to GetAudioFrame() from participant"); |
| _audioFramePool->PushMemory(audioFrame); |
| continue; |
| } |
| if(audioFrame->samples_per_channel_ == 0) { |
| // Empty frame. Don't use it. |
| _audioFramePool->PushMemory(audioFrame); |
| continue; |
| } |
| additionalFramesList->push_back(audioFrame); |
| } |
| } |
| |
| void AudioConferenceMixerImpl::UpdateMixedStatus( |
| std::map<int, MixerParticipant*>& mixedParticipantsMap) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "UpdateMixedStatus(mixedParticipantsMap)"); |
| assert(mixedParticipantsMap.size() <= kMaximumAmountOfMixedParticipants); |
| |
| // Loop through all participants. If they are in the mix map they |
| // were mixed. |
| for (MixerParticipantList::iterator participant = _participantList.begin(); |
| participant != _participantList.end(); |
| ++participant) { |
| bool isMixed = false; |
| for (std::map<int, MixerParticipant*>::iterator it = |
| mixedParticipantsMap.begin(); |
| it != mixedParticipantsMap.end(); |
| ++it) { |
| if (it->second == *participant) { |
| isMixed = true; |
| break; |
| } |
| } |
| (*participant)->_mixHistory->SetIsMixed(isMixed); |
| } |
| } |
| |
| void AudioConferenceMixerImpl::ClearAudioFrameList( |
| AudioFrameList* audioFrameList) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "ClearAudioFrameList(audioFrameList)"); |
| for (AudioFrameList::iterator iter = audioFrameList->begin(); |
| iter != audioFrameList->end(); |
| ++iter) { |
| _audioFramePool->PushMemory(*iter); |
| } |
| audioFrameList->clear(); |
| } |
| |
| void AudioConferenceMixerImpl::UpdateVADPositiveParticipants( |
| AudioFrameList* mixList) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "UpdateVADPositiveParticipants(mixList)"); |
| |
| for (AudioFrameList::iterator iter = mixList->begin(); |
| iter != mixList->end(); |
| ++iter) { |
| CalculateEnergy(**iter); |
| } |
| } |
| |
| bool AudioConferenceMixerImpl::IsParticipantInList( |
| MixerParticipant& participant, |
| MixerParticipantList* participantList) const { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "IsParticipantInList(participant,participantList)"); |
| for (MixerParticipantList::const_iterator iter = participantList->begin(); |
| iter != participantList->end(); |
| ++iter) { |
| if(&participant == *iter) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool AudioConferenceMixerImpl::AddParticipantToList( |
| MixerParticipant& participant, |
| MixerParticipantList* participantList) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "AddParticipantToList(participant, participantList)"); |
| participantList->push_back(&participant); |
| // Make sure that the mixed status is correct for new MixerParticipant. |
| participant._mixHistory->ResetMixedStatus(); |
| return true; |
| } |
| |
| bool AudioConferenceMixerImpl::RemoveParticipantFromList( |
| MixerParticipant& participant, |
| MixerParticipantList* participantList) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "RemoveParticipantFromList(participant, participantList)"); |
| for (MixerParticipantList::iterator iter = participantList->begin(); |
| iter != participantList->end(); |
| ++iter) { |
| if(*iter == &participant) { |
| participantList->erase(iter); |
| // Participant is no longer mixed, reset to default. |
| participant._mixHistory->ResetMixedStatus(); |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| int32_t AudioConferenceMixerImpl::MixFromList( |
| AudioFrame& mixedAudio, |
| const AudioFrameList* audioFrameList) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "MixFromList(mixedAudio, audioFrameList)"); |
| if(audioFrameList->empty()) return 0; |
| |
| uint32_t position = 0; |
| |
| if (_numMixedParticipants == 1) { |
| mixedAudio.timestamp_ = audioFrameList->front()->timestamp_; |
| mixedAudio.elapsed_time_ms_ = audioFrameList->front()->elapsed_time_ms_; |
| } else { |
| // TODO(wu): Issue 3390. |
| // Audio frame timestamp is only supported in one channel case. |
| mixedAudio.timestamp_ = 0; |
| mixedAudio.elapsed_time_ms_ = -1; |
| } |
| |
| for (AudioFrameList::const_iterator iter = audioFrameList->begin(); |
| iter != audioFrameList->end(); |
| ++iter) { |
| if(position >= kMaximumAmountOfMixedParticipants) { |
| WEBRTC_TRACE( |
| kTraceMemory, |
| kTraceAudioMixerServer, |
| _id, |
| "Trying to mix more than max amount of mixed participants:%d!", |
| kMaximumAmountOfMixedParticipants); |
| // Assert and avoid crash |
| assert(false); |
| position = 0; |
| } |
| MixFrames(&mixedAudio, (*iter), use_limiter_); |
| |
| position++; |
| } |
| |
| return 0; |
| } |
| |
| // TODO(andrew): consolidate this function with MixFromList. |
| int32_t AudioConferenceMixerImpl::MixAnonomouslyFromList( |
| AudioFrame& mixedAudio, |
| const AudioFrameList* audioFrameList) { |
| WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id, |
| "MixAnonomouslyFromList(mixedAudio, audioFrameList)"); |
| |
| if(audioFrameList->empty()) return 0; |
| |
| for (AudioFrameList::const_iterator iter = audioFrameList->begin(); |
| iter != audioFrameList->end(); |
| ++iter) { |
| MixFrames(&mixedAudio, *iter, use_limiter_); |
| } |
| return 0; |
| } |
| |
| bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio) { |
| if (!use_limiter_) { |
| return true; |
| } |
| |
| // Smoothly limit the mixed frame. |
| const int error = _limiter->ProcessStream(&mixedAudio); |
| |
| // And now we can safely restore the level. This procedure results in |
| // some loss of resolution, deemed acceptable. |
| // |
| // It's possible to apply the gain in the AGC (with a target level of 0 dbFS |
| // and compression gain of 6 dB). However, in the transition frame when this |
| // is enabled (moving from one to two participants) it has the potential to |
| // create discontinuities in the mixed frame. |
| // |
| // Instead we double the frame (with addition since left-shifting a |
| // negative value is undefined). |
| mixedAudio += mixedAudio; |
| |
| if(error != _limiter->kNoError) { |
| WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, |
| "Error from AudioProcessing: %d", error); |
| assert(false); |
| return false; |
| } |
| return true; |
| } |
| } // namespace webrtc |
