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authorandrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2014-04-03 21:56:01 +0000
committerandrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2014-04-03 21:56:01 +0000
commitf7c73b531c9f2aca2adb87044613a7b7fa94de84 (patch)
tree4e437ae32b38d2a07a05052b5d94fc6eb36710ac /voice_engine/transmit_mixer.cc
parentb5a182a9321ff847a24620c5e991e1749362549a (diff)
downloadwebrtc-f7c73b531c9f2aca2adb87044613a7b7fa94de84.tar.gz
Consolidate audio conversion from Channel and TransmitMixer.
Replace the two versions with a single DownConvertToCodecFormat. As mentioned in comments, this could be further consolidated with RemixAndResample but we should write a full audio converter class in that case. Along the way: - Fix the bug present in Channel::Demultiplex with mono input and a stereo codec. - Remove the 32 kHz max from the OnDataAvailable path. This avoids a 48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we get a straight pass-through to ACM. The 32 kHz conversion is still needed in the RecordedDataIsAvailable path until APM natively supports 48 kHz. - Merge resampler improvements from ACM1 to ACM2. This allows ACM to handle 44.1 kHz audio passed to VoE and was originally done here: https://webrtc-codereview.appspot.com/1590004 - Reuse the RemixAndResample unit tests for DownConvertToCodecFormat. - Remove unused functions from utility.cc. BUG=3155,3000,b/12867572 TESTED=voe_cmd_test using both the OnDataAvailable and RecordedDataIsAvailable paths, with a captured audio format of all combinations of {44.1,48} kHz and {1,2} channels, running through all codecs, and finally using both ACM1 and ACM2. R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11019005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'voice_engine/transmit_mixer.cc')
-rw-r--r--voice_engine/transmit_mixer.cc2797
1 files changed, 1379 insertions, 1418 deletions
diff --git a/voice_engine/transmit_mixer.cc b/voice_engine/transmit_mixer.cc
index 56bf2fa0..3893da36 100644
--- a/voice_engine/transmit_mixer.cc
+++ b/voice_engine/transmit_mixer.cc
@@ -1,1418 +1,1379 @@
-/*
- * 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/voice_engine/transmit_mixer.h"
-
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/voice_engine/channel.h"
-#include "webrtc/voice_engine/channel_manager.h"
-#include "webrtc/voice_engine/include/voe_external_media.h"
-#include "webrtc/voice_engine/statistics.h"
-#include "webrtc/voice_engine/utility.h"
-#include "webrtc/voice_engine/voe_base_impl.h"
-
-#define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a))
-
-namespace webrtc {
-
-namespace voe {
-
-// Used for downmixing before resampling.
-// TODO(ajm): audio_device should advertise the maximum sample rate it can
-// provide.
-static const int kMaxMonoDeviceDataSizeSamples = 1920; // 10 ms, 192 kHz, mono.
-
-// TODO(ajm): The thread safety of this is dubious...
-void
-TransmitMixer::OnPeriodicProcess()
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::OnPeriodicProcess()");
-
-#if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
- if (_typingNoiseWarningPending)
- {
- CriticalSectionScoped cs(&_callbackCritSect);
- if (_voiceEngineObserverPtr)
- {
- if (_typingNoiseDetected) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::OnPeriodicProcess() => "
- "CallbackOnError(VE_TYPING_NOISE_WARNING)");
- _voiceEngineObserverPtr->CallbackOnError(
- -1,
- VE_TYPING_NOISE_WARNING);
- } else {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::OnPeriodicProcess() => "
- "CallbackOnError(VE_TYPING_NOISE_OFF_WARNING)");
- _voiceEngineObserverPtr->CallbackOnError(
- -1,
- VE_TYPING_NOISE_OFF_WARNING);
- }
- }
- _typingNoiseWarningPending = false;
- }
-#endif
-
- bool saturationWarning = false;
- {
- // Modify |_saturationWarning| under lock to avoid conflict with write op
- // in ProcessAudio and also ensure that we don't hold the lock during the
- // callback.
- CriticalSectionScoped cs(&_critSect);
- saturationWarning = _saturationWarning;
- if (_saturationWarning)
- _saturationWarning = false;
- }
-
- if (saturationWarning)
- {
- CriticalSectionScoped cs(&_callbackCritSect);
- if (_voiceEngineObserverPtr)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::OnPeriodicProcess() =>"
- " CallbackOnError(VE_SATURATION_WARNING)");
- _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
- }
- }
-}
-
-
-void TransmitMixer::PlayNotification(int32_t id,
- uint32_t durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void TransmitMixer::RecordNotification(int32_t id,
- uint32_t durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::RecordNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void TransmitMixer::PlayFileEnded(int32_t id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayFileEnded(id=%d)", id);
-
- assert(id == _filePlayerId);
-
- CriticalSectionScoped cs(&_critSect);
-
- _filePlaying = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayFileEnded() =>"
- "file player module is shutdown");
-}
-
-void
-TransmitMixer::RecordFileEnded(int32_t id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded(id=%d)", id);
-
- if (id == _fileRecorderId)
- {
- CriticalSectionScoped cs(&_critSect);
- _fileRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded() => fileRecorder module"
- "is shutdown");
- } else if (id == _fileCallRecorderId)
- {
- CriticalSectionScoped cs(&_critSect);
- _fileCallRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded() => fileCallRecorder"
- "module is shutdown");
- }
-}
-
-int32_t
-TransmitMixer::Create(TransmitMixer*& mixer, uint32_t instanceId)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
- "TransmitMixer::Create(instanceId=%d)", instanceId);
- mixer = new TransmitMixer(instanceId);
- if (mixer == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
- "TransmitMixer::Create() unable to allocate memory"
- "for mixer");
- return -1;
- }
- return 0;
-}
-
-void
-TransmitMixer::Destroy(TransmitMixer*& mixer)
-{
- if (mixer)
- {
- delete mixer;
- mixer = NULL;
- }
-}
-
-TransmitMixer::TransmitMixer(uint32_t instanceId) :
- _engineStatisticsPtr(NULL),
- _channelManagerPtr(NULL),
- audioproc_(NULL),
- _voiceEngineObserverPtr(NULL),
- _processThreadPtr(NULL),
- _filePlayerPtr(NULL),
- _fileRecorderPtr(NULL),
- _fileCallRecorderPtr(NULL),
- // Avoid conflict with other channels by adding 1024 - 1026,
- // won't use as much as 1024 channels.
- _filePlayerId(instanceId + 1024),
- _fileRecorderId(instanceId + 1025),
- _fileCallRecorderId(instanceId + 1026),
- _filePlaying(false),
- _fileRecording(false),
- _fileCallRecording(false),
- _audioLevel(),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
- _typingNoiseWarningPending(false),
- _typingNoiseDetected(false),
-#endif
- _saturationWarning(false),
- _instanceId(instanceId),
- _mixFileWithMicrophone(false),
- _captureLevel(0),
- external_postproc_ptr_(NULL),
- external_preproc_ptr_(NULL),
- _mute(false),
- _remainingMuteMicTimeMs(0),
- stereo_codec_(false),
- swap_stereo_channels_(false)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::TransmitMixer() - ctor");
-}
-
-TransmitMixer::~TransmitMixer()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::~TransmitMixer() - dtor");
- _monitorModule.DeRegisterObserver();
- if (_processThreadPtr)
- {
- _processThreadPtr->DeRegisterModule(&_monitorModule);
- }
- DeRegisterExternalMediaProcessing(kRecordingAllChannelsMixed);
- DeRegisterExternalMediaProcessing(kRecordingPreprocessing);
- {
- CriticalSectionScoped cs(&_critSect);
- if (_fileRecorderPtr)
- {
- _fileRecorderPtr->RegisterModuleFileCallback(NULL);
- _fileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- }
- if (_fileCallRecorderPtr)
- {
- _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
- _fileCallRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- }
- if (_filePlayerPtr)
- {
- _filePlayerPtr->RegisterModuleFileCallback(NULL);
- _filePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- }
- }
- delete &_critSect;
- delete &_callbackCritSect;
-}
-
-int32_t
-TransmitMixer::SetEngineInformation(ProcessThread& processThread,
- Statistics& engineStatistics,
- ChannelManager& channelManager)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::SetEngineInformation()");
-
- _processThreadPtr = &processThread;
- _engineStatisticsPtr = &engineStatistics;
- _channelManagerPtr = &channelManager;
-
- if (_processThreadPtr->RegisterModule(&_monitorModule) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::SetEngineInformation() failed to"
- "register the monitor module");
- } else
- {
- _monitorModule.RegisterObserver(*this);
- }
-
- return 0;
-}
-
-int32_t
-TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RegisterVoiceEngineObserver()");
- CriticalSectionScoped cs(&_callbackCritSect);
-
- if (_voiceEngineObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "RegisterVoiceEngineObserver() observer already enabled");
- return -1;
- }
- _voiceEngineObserverPtr = &observer;
- return 0;
-}
-
-int32_t
-TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::SetAudioProcessingModule("
- "audioProcessingModule=0x%x)",
- audioProcessingModule);
- audioproc_ = audioProcessingModule;
- return 0;
-}
-
-void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) {
- *max_sample_rate = 8000;
- *max_channels = 1;
- for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
- it.Increment()) {
- Channel* channel = it.GetChannel();
- if (channel->Sending()) {
- CodecInst codec;
- channel->GetSendCodec(codec);
- // TODO(tlegrand): Remove the 32 kHz restriction once we have full 48 kHz
- // support in Audio Coding Module.
- *max_sample_rate = std::min(32000,
- std::max(*max_sample_rate, codec.plfreq));
- *max_channels = std::max(*max_channels, codec.channels);
- }
- }
-}
-
-int32_t
-TransmitMixer::PrepareDemux(const void* audioSamples,
- uint32_t nSamples,
- uint8_t nChannels,
- uint32_t samplesPerSec,
- uint16_t totalDelayMS,
- int32_t clockDrift,
- uint16_t currentMicLevel,
- bool keyPressed)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
- "samplesPerSec=%u, totalDelayMS=%u, clockDrift=%d,"
- "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec,
- totalDelayMS, clockDrift, currentMicLevel);
-
- // --- Resample input audio and create/store the initial audio frame
- if (GenerateAudioFrame(static_cast<const int16_t*>(audioSamples),
- nSamples,
- nChannels,
- samplesPerSec) == -1)
- {
- return -1;
- }
-
- {
- CriticalSectionScoped cs(&_callbackCritSect);
- if (external_preproc_ptr_) {
- external_preproc_ptr_->Process(-1, kRecordingPreprocessing,
- _audioFrame.data_,
- _audioFrame.samples_per_channel_,
- _audioFrame.sample_rate_hz_,
- _audioFrame.num_channels_ == 2);
- }
- }
-
- // --- Near-end audio processing.
- ProcessAudio(totalDelayMS, clockDrift, currentMicLevel, keyPressed);
-
- if (swap_stereo_channels_ && stereo_codec_)
- // Only bother swapping if we're using a stereo codec.
- AudioFrameOperations::SwapStereoChannels(&_audioFrame);
-
- // --- Annoying typing detection (utilizes the APM/VAD decision)
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
- TypingDetection(keyPressed);
-#endif
-
- // --- Mute during DTMF tone if direct feedback is enabled
- if (_remainingMuteMicTimeMs > 0)
- {
- AudioFrameOperations::Mute(_audioFrame);
- _remainingMuteMicTimeMs -= 10;
- if (_remainingMuteMicTimeMs < 0)
- {
- _remainingMuteMicTimeMs = 0;
- }
- }
-
- // --- Mute signal
- if (_mute)
- {
- AudioFrameOperations::Mute(_audioFrame);
- }
-
- // --- Mix with file (does not affect the mixing frequency)
- if (_filePlaying)
- {
- MixOrReplaceAudioWithFile(_audioFrame.sample_rate_hz_);
- }
-
- // --- Record to file
- bool file_recording = false;
- {
- CriticalSectionScoped cs(&_critSect);
- file_recording = _fileRecording;
- }
- if (file_recording)
- {
- RecordAudioToFile(_audioFrame.sample_rate_hz_);
- }
-
- {
- CriticalSectionScoped cs(&_callbackCritSect);
- if (external_postproc_ptr_) {
- external_postproc_ptr_->Process(-1, kRecordingAllChannelsMixed,
- _audioFrame.data_,
- _audioFrame.samples_per_channel_,
- _audioFrame.sample_rate_hz_,
- _audioFrame.num_channels_ == 2);
- }
- }
-
- // --- Measure audio level of speech after all processing.
- _audioLevel.ComputeLevel(_audioFrame);
- return 0;
-}
-
-int32_t
-TransmitMixer::DemuxAndMix()
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::DemuxAndMix()");
-
- for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
- it.Increment())
- {
- Channel* channelPtr = it.GetChannel();
- if (channelPtr->InputIsOnHold())
- {
- channelPtr->UpdateLocalTimeStamp();
- } else if (channelPtr->Sending())
- {
- // Demultiplex makes a copy of its input.
- channelPtr->Demultiplex(_audioFrame);
- channelPtr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_);
- }
- }
- return 0;
-}
-
-void TransmitMixer::DemuxAndMix(const int voe_channels[],
- int number_of_voe_channels) {
- for (int i = 0; i < number_of_voe_channels; ++i) {
- voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
- voe::Channel* channel_ptr = ch.channel();
- if (channel_ptr) {
- if (channel_ptr->InputIsOnHold()) {
- channel_ptr->UpdateLocalTimeStamp();
- } else if (channel_ptr->Sending()) {
- // Demultiplex makes a copy of its input.
- channel_ptr->Demultiplex(_audioFrame);
- channel_ptr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_);
- }
- }
- }
-}
-
-int32_t
-TransmitMixer::EncodeAndSend()
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::EncodeAndSend()");
-
- for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
- it.Increment())
- {
- Channel* channelPtr = it.GetChannel();
- if (channelPtr->Sending() && !channelPtr->InputIsOnHold())
- {
- channelPtr->EncodeAndSend();
- }
- }
- return 0;
-}
-
-void TransmitMixer::EncodeAndSend(const int voe_channels[],
- int number_of_voe_channels) {
- for (int i = 0; i < number_of_voe_channels; ++i) {
- voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
- voe::Channel* channel_ptr = ch.channel();
- if (channel_ptr && channel_ptr->Sending() && !channel_ptr->InputIsOnHold())
- channel_ptr->EncodeAndSend();
- }
-}
-
-uint32_t TransmitMixer::CaptureLevel() const
-{
- return _captureLevel;
-}
-
-void
-TransmitMixer::UpdateMuteMicrophoneTime(uint32_t lengthMs)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)",
- lengthMs);
- _remainingMuteMicTimeMs = lengthMs;
-}
-
-int32_t
-TransmitMixer::StopSend()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StopSend()");
- _audioLevel.Clear();
- return 0;
-}
-
-int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName,
- bool loop,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartPlayingFileAsMicrophone("
- "fileNameUTF8[]=%s,loop=%d, format=%d, volumeScaling=%5.3f,"
- " startPosition=%d, stopPosition=%d)", fileName, loop,
- format, volumeScaling, startPosition, stopPosition);
-
- if (_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is already playing");
- return 0;
- }
-
- CriticalSectionScoped cs(&_critSect);
-
- // Destroy the old instance
- if (_filePlayerPtr)
- {
- _filePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- }
-
- // Dynamically create the instance
- _filePlayerPtr
- = FilePlayer::CreateFilePlayer(_filePlayerId,
- (const FileFormats) format);
-
- if (_filePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (_filePlayerPtr->StartPlayingFile(
- fileName,
- loop,
- startPosition,
- volumeScaling,
- notificationTime,
- stopPosition,
- (const CodecInst*) codecInst) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- _filePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- return -1;
- }
-
- _filePlayerPtr->RegisterModuleFileCallback(this);
- _filePlaying = true;
-
- return 0;
-}
-
-int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
- FileFormats format,
- int startPosition,
- float volumeScaling,
- int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::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;
- }
-
- if (_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is already playing");
- return 0;
- }
-
- CriticalSectionScoped cs(&_critSect);
-
- // Destroy the old instance
- if (_filePlayerPtr)
- {
- _filePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- }
-
- // Dynamically create the instance
- _filePlayerPtr
- = FilePlayer::CreateFilePlayer(_filePlayerId,
- (const FileFormats) format);
-
- if (_filePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceWarning,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const uint32_t notificationTime(0);
-
- if (_filePlayerPtr->StartPlayingFile(
- (InStream&) *stream,
- startPosition,
- volumeScaling,
- notificationTime,
- stopPosition,
- (const CodecInst*) codecInst) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- _filePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- return -1;
- }
- _filePlayerPtr->RegisterModuleFileCallback(this);
- _filePlaying = true;
-
- return 0;
-}
-
-int TransmitMixer::StopPlayingFileAsMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::StopPlayingFileAsMicrophone()");
-
- if (!_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "StopPlayingFileAsMicrophone() isnot playing");
- return 0;
- }
-
- CriticalSectionScoped cs(&_critSect);
-
- if (_filePlayerPtr->StopPlayingFile() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_STOP_PLAYOUT, kTraceError,
- "StopPlayingFile() couldnot stop playing file");
- return -1;
- }
-
- _filePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- _filePlaying = false;
-
- return 0;
-}
-
-int TransmitMixer::IsPlayingFileAsMicrophone() const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::IsPlayingFileAsMicrophone()");
- return _filePlaying;
-}
-
-int TransmitMixer::ScaleFileAsMicrophonePlayout(float scale)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::ScaleFileAsMicrophonePlayout(scale=%5.3f)",
- scale);
-
- CriticalSectionScoped cs(&_critSect);
-
- if (!_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "ScaleFileAsMicrophonePlayout() isnot playing file");
- return -1;
- }
-
- if ((_filePlayerPtr == NULL) ||
- (_filePlayerPtr->SetAudioScaling(scale) != 0))
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "SetAudioScaling() failed to scale playout");
- return -1;
- }
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingMicrophone(const char* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingMicrophone(fileName=%s)",
- fileName);
-
- CriticalSectionScoped cs(&_critSect);
-
- if (_fileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingMicrophone() 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 < 0 || codecInst->channels > 2))
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() 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;
- }
-
- // Destroy the old instance
- if (_fileRecorderPtr)
- {
- _fileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- }
-
- _fileRecorderPtr =
- FileRecorder::CreateFileRecorder(_fileRecorderId,
- (const FileFormats) format);
- if (_fileRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_fileRecorderPtr->StartRecordingAudioFile(
- fileName,
- (const CodecInst&) *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _fileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- return -1;
- }
- _fileRecorderPtr->RegisterModuleFileCallback(this);
- _fileRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingMicrophone()");
-
- CriticalSectionScoped cs(&_critSect);
-
- if (_fileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingMicrophone() 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,
- "StartRecordingMicrophone() 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;
- }
-
- // Destroy the old instance
- if (_fileRecorderPtr)
- {
- _fileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- }
-
- _fileRecorderPtr =
- FileRecorder::CreateFileRecorder(_fileRecorderId,
- (const FileFormats) format);
- if (_fileRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_fileRecorderPtr->StartRecordingAudioFile(*stream,
- *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _fileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- return -1;
- }
-
- _fileRecorderPtr->RegisterModuleFileCallback(this);
- _fileRecording = true;
-
- return 0;
-}
-
-
-int TransmitMixer::StopRecordingMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StopRecordingMicrophone()");
-
- CriticalSectionScoped cs(&_critSect);
-
- if (!_fileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StopRecordingMicrophone() isnot recording");
- return 0;
- }
-
- if (_fileRecorderPtr->StopRecording() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording(), could not stop recording");
- return -1;
- }
- _fileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- _fileRecording = false;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingCall(const char* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingCall(fileName=%s)", fileName);
-
- if (_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingCall() 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,
- "StartRecordingCall() 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(&_critSect);
-
- // Destroy the old instance
- if (_fileCallRecorderPtr)
- {
- _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- }
-
- _fileCallRecorderPtr
- = FileRecorder::CreateFileRecorder(_fileCallRecorderId,
- (const FileFormats) format);
- if (_fileCallRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingCall() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_fileCallRecorderPtr->StartRecordingAudioFile(
- fileName,
- (const CodecInst&) *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _fileCallRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- return -1;
- }
- _fileCallRecorderPtr->RegisterModuleFileCallback(this);
- _fileCallRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingCall(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingCall()");
-
- if (_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingCall() 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,
- "StartRecordingCall() 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(&_critSect);
-
- // Destroy the old instance
- if (_fileCallRecorderPtr)
- {
- _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- }
-
- _fileCallRecorderPtr =
- FileRecorder::CreateFileRecorder(_fileCallRecorderId,
- (const FileFormats) format);
- if (_fileCallRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingCall() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_fileCallRecorderPtr->StartRecordingAudioFile(*stream,
- *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _fileCallRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- return -1;
- }
-
- _fileCallRecorderPtr->RegisterModuleFileCallback(this);
- _fileCallRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StopRecordingCall()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StopRecordingCall()");
-
- if (!_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StopRecordingCall() file isnot recording");
- return -1;
- }
-
- CriticalSectionScoped cs(&_critSect);
-
- if (_fileCallRecorderPtr->StopRecording() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording(), could not stop recording");
- return -1;
- }
-
- _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- _fileCallRecording = false;
-
- return 0;
-}
-
-void
-TransmitMixer::SetMixWithMicStatus(bool mix)
-{
- _mixFileWithMicrophone = mix;
-}
-
-int TransmitMixer::RegisterExternalMediaProcessing(
- VoEMediaProcess* object,
- ProcessingTypes type) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RegisterExternalMediaProcessing()");
-
- CriticalSectionScoped cs(&_callbackCritSect);
- if (!object) {
- return -1;
- }
-
- // Store the callback object according to the processing type.
- if (type == kRecordingAllChannelsMixed) {
- external_postproc_ptr_ = object;
- } else if (type == kRecordingPreprocessing) {
- external_preproc_ptr_ = object;
- } else {
- return -1;
- }
- return 0;
-}
-
-int TransmitMixer::DeRegisterExternalMediaProcessing(ProcessingTypes type) {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::DeRegisterExternalMediaProcessing()");
-
- CriticalSectionScoped cs(&_callbackCritSect);
- if (type == kRecordingAllChannelsMixed) {
- external_postproc_ptr_ = NULL;
- } else if (type == kRecordingPreprocessing) {
- external_preproc_ptr_ = NULL;
- } else {
- return -1;
- }
- return 0;
-}
-
-int
-TransmitMixer::SetMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::SetMute(enable=%d)", enable);
- _mute = enable;
- return 0;
-}
-
-bool
-TransmitMixer::Mute() const
-{
- return _mute;
-}
-
-int8_t TransmitMixer::AudioLevel() const
-{
- // Speech + file level [0,9]
- return _audioLevel.Level();
-}
-
-int16_t TransmitMixer::AudioLevelFullRange() const
-{
- // Speech + file level [0,32767]
- return _audioLevel.LevelFullRange();
-}
-
-bool TransmitMixer::IsRecordingCall()
-{
- return _fileCallRecording;
-}
-
-bool TransmitMixer::IsRecordingMic()
-{
- CriticalSectionScoped cs(&_critSect);
- return _fileRecording;
-}
-
-// TODO(andrew): use RemixAndResample for this.
-int TransmitMixer::GenerateAudioFrame(const int16_t audio[],
- int samples_per_channel,
- int num_channels,
- int sample_rate_hz) {
- int destination_rate;
- int num_codec_channels;
- GetSendCodecInfo(&destination_rate, &num_codec_channels);
-
- // Never upsample the capture signal here. This should be done at the
- // end of the send chain.
- destination_rate = std::min(destination_rate, sample_rate_hz);
- stereo_codec_ = num_codec_channels == 2;
-
- const int16_t* audio_ptr = audio;
- int16_t mono_audio[kMaxMonoDeviceDataSizeSamples];
- assert(samples_per_channel <= kMaxMonoDeviceDataSizeSamples);
- // If no stereo codecs are in use, we downmix a stereo stream from the
- // device early in the chain, before resampling.
- if (num_channels == 2 && !stereo_codec_) {
- AudioFrameOperations::StereoToMono(audio, samples_per_channel,
- mono_audio);
- audio_ptr = mono_audio;
- num_channels = 1;
- }
-
- if (resampler_.InitializeIfNeeded(sample_rate_hz,
- destination_rate,
- num_channels) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::GenerateAudioFrame() unable to resample");
- return -1;
- }
-
- int out_length = resampler_.Resample(audio_ptr,
- samples_per_channel * num_channels,
- _audioFrame.data_,
- AudioFrame::kMaxDataSizeSamples);
- if (out_length == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::GenerateAudioFrame() resampling failed");
- return -1;
- }
-
- _audioFrame.samples_per_channel_ = out_length / num_channels;
- _audioFrame.id_ = _instanceId;
- _audioFrame.timestamp_ = -1;
- _audioFrame.sample_rate_hz_ = destination_rate;
- _audioFrame.speech_type_ = AudioFrame::kNormalSpeech;
- _audioFrame.vad_activity_ = AudioFrame::kVadUnknown;
- _audioFrame.num_channels_ = num_channels;
-
- return 0;
-}
-
-int32_t TransmitMixer::RecordAudioToFile(
- uint32_t mixingFrequency)
-{
- CriticalSectionScoped cs(&_critSect);
- if (_fileRecorderPtr == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordAudioToFile() filerecorder doesnot"
- "exist");
- return -1;
- }
-
- if (_fileRecorderPtr->RecordAudioToFile(_audioFrame) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordAudioToFile() file recording"
- "failed");
- return -1;
- }
-
- return 0;
-}
-
-int32_t TransmitMixer::MixOrReplaceAudioWithFile(
- int mixingFrequency)
-{
- scoped_array<int16_t> fileBuffer(new int16_t[640]);
-
- int fileSamples(0);
- {
- CriticalSectionScoped cs(&_critSect);
- if (_filePlayerPtr == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, -1),
- "TransmitMixer::MixOrReplaceAudioWithFile()"
- "fileplayer doesnot exist");
- return -1;
- }
-
- if (_filePlayerPtr->Get10msAudioFromFile(fileBuffer.get(),
- fileSamples,
- mixingFrequency) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::MixOrReplaceAudioWithFile() file"
- " mixing failed");
- return -1;
- }
- }
-
- assert(_audioFrame.samples_per_channel_ == fileSamples);
-
- if (_mixFileWithMicrophone)
- {
- // 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);
- } else
- {
- // Replace ACM audio with file.
- // Currently file stream is always mono.
- // TODO(xians): Change the code when FilePlayer supports real stereo.
- _audioFrame.UpdateFrame(-1,
- -1,
- fileBuffer.get(),
- fileSamples,
- mixingFrequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadUnknown,
- 1);
- }
- return 0;
-}
-
-void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
- int current_mic_level, bool key_pressed) {
- if (audioproc_->set_stream_delay_ms(delay_ms) != 0) {
- // A redundant warning is reported in AudioDevice, which we've throttled
- // to avoid flooding the logs. Relegate this one to LS_VERBOSE to avoid
- // repeating the problem here.
- LOG_FERR1(LS_VERBOSE, set_stream_delay_ms, delay_ms);
- }
-
- GainControl* agc = audioproc_->gain_control();
- if (agc->set_stream_analog_level(current_mic_level) != 0) {
- LOG_FERR1(LS_ERROR, set_stream_analog_level, current_mic_level);
- assert(false);
- }
-
- EchoCancellation* aec = audioproc_->echo_cancellation();
- if (aec->is_drift_compensation_enabled()) {
- aec->set_stream_drift_samples(clock_drift);
- }
-
- audioproc_->set_stream_key_pressed(key_pressed);
-
- int err = audioproc_->ProcessStream(&_audioFrame);
- if (err != 0) {
- LOG(LS_ERROR) << "ProcessStream() error: " << err;
- assert(false);
- }
-
- // Store new capture level. Only updated when analog AGC is enabled.
- _captureLevel = agc->stream_analog_level();
-
- CriticalSectionScoped cs(&_critSect);
- // Triggers a callback in OnPeriodicProcess().
- _saturationWarning |= agc->stream_is_saturated();
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-void TransmitMixer::TypingDetection(bool keyPressed)
-{
- // We let the VAD determine if we're using this feature or not.
- if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) {
- return;
- }
-
- bool vadActive = _audioFrame.vad_activity_ == AudioFrame::kVadActive;
- if (_typingDetection.Process(keyPressed, vadActive)) {
- _typingNoiseWarningPending = true;
- _typingNoiseDetected = true;
- } else {
- // If there is already a warning pending, do not change the state.
- // Otherwise set a warning pending if last callback was for noise detected.
- if (!_typingNoiseWarningPending && _typingNoiseDetected) {
- _typingNoiseWarningPending = true;
- _typingNoiseDetected = false;
- }
- }
-}
-#endif
-
-int TransmitMixer::GetMixingFrequency()
-{
- assert(_audioFrame.sample_rate_hz_ != 0);
- return _audioFrame.sample_rate_hz_;
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-int TransmitMixer::TimeSinceLastTyping(int &seconds)
-{
- // We check in VoEAudioProcessingImpl that this is only called when
- // typing detection is active.
- seconds = _typingDetection.TimeSinceLastDetectionInSeconds();
- return 0;
-}
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-int TransmitMixer::SetTypingDetectionParameters(int timeWindow,
- int costPerTyping,
- int reportingThreshold,
- int penaltyDecay,
- int typeEventDelay)
-{
- _typingDetection.SetParameters(timeWindow,
- costPerTyping,
- reportingThreshold,
- penaltyDecay,
- typeEventDelay,
- 0);
- return 0;
-}
-#endif
-
-void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
- swap_stereo_channels_ = enable;
-}
-
-bool TransmitMixer::IsStereoChannelSwappingEnabled() {
- return swap_stereo_channels_;
-}
-
-} // namespace voe
-
-} // namespace webrtc
+/*
+ * 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/voice_engine/transmit_mixer.h"
+
+#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/voice_engine/channel.h"
+#include "webrtc/voice_engine/channel_manager.h"
+#include "webrtc/voice_engine/include/voe_external_media.h"
+#include "webrtc/voice_engine/statistics.h"
+#include "webrtc/voice_engine/utility.h"
+#include "webrtc/voice_engine/voe_base_impl.h"
+
+#define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a))
+
+namespace webrtc {
+namespace voe {
+
+// TODO(ajm): The thread safety of this is dubious...
+void
+TransmitMixer::OnPeriodicProcess()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::OnPeriodicProcess()");
+
+#if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
+ if (_typingNoiseWarningPending)
+ {
+ CriticalSectionScoped cs(&_callbackCritSect);
+ if (_voiceEngineObserverPtr)
+ {
+ if (_typingNoiseDetected) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::OnPeriodicProcess() => "
+ "CallbackOnError(VE_TYPING_NOISE_WARNING)");
+ _voiceEngineObserverPtr->CallbackOnError(
+ -1,
+ VE_TYPING_NOISE_WARNING);
+ } else {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::OnPeriodicProcess() => "
+ "CallbackOnError(VE_TYPING_NOISE_OFF_WARNING)");
+ _voiceEngineObserverPtr->CallbackOnError(
+ -1,
+ VE_TYPING_NOISE_OFF_WARNING);
+ }
+ }
+ _typingNoiseWarningPending = false;
+ }
+#endif
+
+ bool saturationWarning = false;
+ {
+ // Modify |_saturationWarning| under lock to avoid conflict with write op
+ // in ProcessAudio and also ensure that we don't hold the lock during the
+ // callback.
+ CriticalSectionScoped cs(&_critSect);
+ saturationWarning = _saturationWarning;
+ if (_saturationWarning)
+ _saturationWarning = false;
+ }
+
+ if (saturationWarning)
+ {
+ CriticalSectionScoped cs(&_callbackCritSect);
+ if (_voiceEngineObserverPtr)
+ {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::OnPeriodicProcess() =>"
+ " CallbackOnError(VE_SATURATION_WARNING)");
+ _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
+ }
+ }
+}
+
+
+void TransmitMixer::PlayNotification(int32_t id,
+ uint32_t durationMs)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::PlayNotification(id=%d, durationMs=%d)",
+ id, durationMs);
+
+ // Not implement yet
+}
+
+void TransmitMixer::RecordNotification(int32_t id,
+ uint32_t durationMs)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
+ "TransmitMixer::RecordNotification(id=%d, durationMs=%d)",
+ id, durationMs);
+
+ // Not implement yet
+}
+
+void TransmitMixer::PlayFileEnded(int32_t id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::PlayFileEnded(id=%d)", id);
+
+ assert(id == _filePlayerId);
+
+ CriticalSectionScoped cs(&_critSect);
+
+ _filePlaying = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::PlayFileEnded() =>"
+ "file player module is shutdown");
+}
+
+void
+TransmitMixer::RecordFileEnded(int32_t id)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordFileEnded(id=%d)", id);
+
+ if (id == _fileRecorderId)
+ {
+ CriticalSectionScoped cs(&_critSect);
+ _fileRecording = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordFileEnded() => fileRecorder module"
+ "is shutdown");
+ } else if (id == _fileCallRecorderId)
+ {
+ CriticalSectionScoped cs(&_critSect);
+ _fileCallRecording = false;
+ WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordFileEnded() => fileCallRecorder"
+ "module is shutdown");
+ }
+}
+
+int32_t
+TransmitMixer::Create(TransmitMixer*& mixer, uint32_t instanceId)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
+ "TransmitMixer::Create(instanceId=%d)", instanceId);
+ mixer = new TransmitMixer(instanceId);
+ if (mixer == NULL)
+ {
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
+ "TransmitMixer::Create() unable to allocate memory"
+ "for mixer");
+ return -1;
+ }
+ return 0;
+}
+
+void
+TransmitMixer::Destroy(TransmitMixer*& mixer)
+{
+ if (mixer)
+ {
+ delete mixer;
+ mixer = NULL;
+ }
+}
+
+TransmitMixer::TransmitMixer(uint32_t instanceId) :
+ _engineStatisticsPtr(NULL),
+ _channelManagerPtr(NULL),
+ audioproc_(NULL),
+ _voiceEngineObserverPtr(NULL),
+ _processThreadPtr(NULL),
+ _filePlayerPtr(NULL),
+ _fileRecorderPtr(NULL),
+ _fileCallRecorderPtr(NULL),
+ // Avoid conflict with other channels by adding 1024 - 1026,
+ // won't use as much as 1024 channels.
+ _filePlayerId(instanceId + 1024),
+ _fileRecorderId(instanceId + 1025),
+ _fileCallRecorderId(instanceId + 1026),
+ _filePlaying(false),
+ _fileRecording(false),
+ _fileCallRecording(false),
+ _audioLevel(),
+ _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+ _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+ _typingNoiseWarningPending(false),
+ _typingNoiseDetected(false),
+#endif
+ _saturationWarning(false),
+ _instanceId(instanceId),
+ _mixFileWithMicrophone(false),
+ _captureLevel(0),
+ external_postproc_ptr_(NULL),
+ external_preproc_ptr_(NULL),
+ _mute(false),
+ _remainingMuteMicTimeMs(0),
+ stereo_codec_(false),
+ swap_stereo_channels_(false)
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::TransmitMixer() - ctor");
+}
+
+TransmitMixer::~TransmitMixer()
+{
+ WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::~TransmitMixer() - dtor");
+ _monitorModule.DeRegisterObserver();
+ if (_processThreadPtr)
+ {
+ _processThreadPtr->DeRegisterModule(&_monitorModule);
+ }
+ DeRegisterExternalMediaProcessing(kRecordingAllChannelsMixed);
+ DeRegisterExternalMediaProcessing(kRecordingPreprocessing);
+ {
+ CriticalSectionScoped cs(&_critSect);
+ if (_fileRecorderPtr)
+ {
+ _fileRecorderPtr->RegisterModuleFileCallback(NULL);
+ _fileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ }
+ if (_fileCallRecorderPtr)
+ {
+ _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
+ _fileCallRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ }
+ if (_filePlayerPtr)
+ {
+ _filePlayerPtr->RegisterModuleFileCallback(NULL);
+ _filePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ }
+ }
+ delete &_critSect;
+ delete &_callbackCritSect;
+}
+
+int32_t
+TransmitMixer::SetEngineInformation(ProcessThread& processThread,
+ Statistics& engineStatistics,
+ ChannelManager& channelManager)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::SetEngineInformation()");
+
+ _processThreadPtr = &processThread;
+ _engineStatisticsPtr = &engineStatistics;
+ _channelManagerPtr = &channelManager;
+
+ if (_processThreadPtr->RegisterModule(&_monitorModule) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::SetEngineInformation() failed to"
+ "register the monitor module");
+ } else
+ {
+ _monitorModule.RegisterObserver(*this);
+ }
+
+ return 0;
+}
+
+int32_t
+TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RegisterVoiceEngineObserver()");
+ CriticalSectionScoped cs(&_callbackCritSect);
+
+ if (_voiceEngineObserverPtr)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "RegisterVoiceEngineObserver() observer already enabled");
+ return -1;
+ }
+ _voiceEngineObserverPtr = &observer;
+ return 0;
+}
+
+int32_t
+TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::SetAudioProcessingModule("
+ "audioProcessingModule=0x%x)",
+ audioProcessingModule);
+ audioproc_ = audioProcessingModule;
+ return 0;
+}
+
+void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) {
+ *max_sample_rate = 8000;
+ *max_channels = 1;
+ for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
+ it.Increment()) {
+ Channel* channel = it.GetChannel();
+ if (channel->Sending()) {
+ CodecInst codec;
+ channel->GetSendCodec(codec);
+ *max_sample_rate = std::max(*max_sample_rate, codec.plfreq);
+ *max_channels = std::max(*max_channels, codec.channels);
+ }
+ }
+}
+
+int32_t
+TransmitMixer::PrepareDemux(const void* audioSamples,
+ uint32_t nSamples,
+ uint8_t nChannels,
+ uint32_t samplesPerSec,
+ uint16_t totalDelayMS,
+ int32_t clockDrift,
+ uint16_t currentMicLevel,
+ bool keyPressed)
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
+ "samplesPerSec=%u, totalDelayMS=%u, clockDrift=%d,"
+ "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec,
+ totalDelayMS, clockDrift, currentMicLevel);
+
+ // --- Resample input audio and create/store the initial audio frame
+ GenerateAudioFrame(static_cast<const int16_t*>(audioSamples),
+ nSamples,
+ nChannels,
+ samplesPerSec);
+
+ {
+ CriticalSectionScoped cs(&_callbackCritSect);
+ if (external_preproc_ptr_) {
+ external_preproc_ptr_->Process(-1, kRecordingPreprocessing,
+ _audioFrame.data_,
+ _audioFrame.samples_per_channel_,
+ _audioFrame.sample_rate_hz_,
+ _audioFrame.num_channels_ == 2);
+ }
+ }
+
+ // --- Near-end audio processing.
+ ProcessAudio(totalDelayMS, clockDrift, currentMicLevel, keyPressed);
+
+ if (swap_stereo_channels_ && stereo_codec_)
+ // Only bother swapping if we're using a stereo codec.
+ AudioFrameOperations::SwapStereoChannels(&_audioFrame);
+
+ // --- Annoying typing detection (utilizes the APM/VAD decision)
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+ TypingDetection(keyPressed);
+#endif
+
+ // --- Mute during DTMF tone if direct feedback is enabled
+ if (_remainingMuteMicTimeMs > 0)
+ {
+ AudioFrameOperations::Mute(_audioFrame);
+ _remainingMuteMicTimeMs -= 10;
+ if (_remainingMuteMicTimeMs < 0)
+ {
+ _remainingMuteMicTimeMs = 0;
+ }
+ }
+
+ // --- Mute signal
+ if (_mute)
+ {
+ AudioFrameOperations::Mute(_audioFrame);
+ }
+
+ // --- Mix with file (does not affect the mixing frequency)
+ if (_filePlaying)
+ {
+ MixOrReplaceAudioWithFile(_audioFrame.sample_rate_hz_);
+ }
+
+ // --- Record to file
+ bool file_recording = false;
+ {
+ CriticalSectionScoped cs(&_critSect);
+ file_recording = _fileRecording;
+ }
+ if (file_recording)
+ {
+ RecordAudioToFile(_audioFrame.sample_rate_hz_);
+ }
+
+ {
+ CriticalSectionScoped cs(&_callbackCritSect);
+ if (external_postproc_ptr_) {
+ external_postproc_ptr_->Process(-1, kRecordingAllChannelsMixed,
+ _audioFrame.data_,
+ _audioFrame.samples_per_channel_,
+ _audioFrame.sample_rate_hz_,
+ _audioFrame.num_channels_ == 2);
+ }
+ }
+
+ // --- Measure audio level of speech after all processing.
+ _audioLevel.ComputeLevel(_audioFrame);
+ return 0;
+}
+
+int32_t
+TransmitMixer::DemuxAndMix()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::DemuxAndMix()");
+
+ for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
+ it.Increment())
+ {
+ Channel* channelPtr = it.GetChannel();
+ if (channelPtr->InputIsOnHold())
+ {
+ channelPtr->UpdateLocalTimeStamp();
+ } else if (channelPtr->Sending())
+ {
+ // Demultiplex makes a copy of its input.
+ channelPtr->Demultiplex(_audioFrame);
+ channelPtr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_);
+ }
+ }
+ return 0;
+}
+
+void TransmitMixer::DemuxAndMix(const int voe_channels[],
+ int number_of_voe_channels) {
+ for (int i = 0; i < number_of_voe_channels; ++i) {
+ voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
+ voe::Channel* channel_ptr = ch.channel();
+ if (channel_ptr) {
+ if (channel_ptr->InputIsOnHold()) {
+ channel_ptr->UpdateLocalTimeStamp();
+ } else if (channel_ptr->Sending()) {
+ // Demultiplex makes a copy of its input.
+ channel_ptr->Demultiplex(_audioFrame);
+ channel_ptr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_);
+ }
+ }
+ }
+}
+
+int32_t
+TransmitMixer::EncodeAndSend()
+{
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::EncodeAndSend()");
+
+ for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
+ it.Increment())
+ {
+ Channel* channelPtr = it.GetChannel();
+ if (channelPtr->Sending() && !channelPtr->InputIsOnHold())
+ {
+ channelPtr->EncodeAndSend();
+ }
+ }
+ return 0;
+}
+
+void TransmitMixer::EncodeAndSend(const int voe_channels[],
+ int number_of_voe_channels) {
+ for (int i = 0; i < number_of_voe_channels; ++i) {
+ voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]);
+ voe::Channel* channel_ptr = ch.channel();
+ if (channel_ptr && channel_ptr->Sending() && !channel_ptr->InputIsOnHold())
+ channel_ptr->EncodeAndSend();
+ }
+}
+
+uint32_t TransmitMixer::CaptureLevel() const
+{
+ return _captureLevel;
+}
+
+void
+TransmitMixer::UpdateMuteMicrophoneTime(uint32_t lengthMs)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)",
+ lengthMs);
+ _remainingMuteMicTimeMs = lengthMs;
+}
+
+int32_t
+TransmitMixer::StopSend()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StopSend()");
+ _audioLevel.Clear();
+ return 0;
+}
+
+int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName,
+ bool loop,
+ FileFormats format,
+ int startPosition,
+ float volumeScaling,
+ int stopPosition,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartPlayingFileAsMicrophone("
+ "fileNameUTF8[]=%s,loop=%d, format=%d, volumeScaling=%5.3f,"
+ " startPosition=%d, stopPosition=%d)", fileName, loop,
+ format, volumeScaling, startPosition, stopPosition);
+
+ if (_filePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceWarning,
+ "StartPlayingFileAsMicrophone() is already playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(&_critSect);
+
+ // Destroy the old instance
+ if (_filePlayerPtr)
+ {
+ _filePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ }
+
+ // Dynamically create the instance
+ _filePlayerPtr
+ = FilePlayer::CreateFilePlayer(_filePlayerId,
+ (const FileFormats) format);
+
+ if (_filePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
+ return -1;
+ }
+
+ const uint32_t notificationTime(0);
+
+ if (_filePlayerPtr->StartPlayingFile(
+ fileName,
+ loop,
+ startPosition,
+ volumeScaling,
+ notificationTime,
+ stopPosition,
+ (const CodecInst*) codecInst) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFile() failed to start file playout");
+ _filePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ return -1;
+ }
+
+ _filePlayerPtr->RegisterModuleFileCallback(this);
+ _filePlaying = true;
+
+ return 0;
+}
+
+int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
+ FileFormats format,
+ int startPosition,
+ float volumeScaling,
+ int stopPosition,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "TransmitMixer::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;
+ }
+
+ if (_filePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_ALREADY_PLAYING, kTraceWarning,
+ "StartPlayingFileAsMicrophone() is already playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(&_critSect);
+
+ // Destroy the old instance
+ if (_filePlayerPtr)
+ {
+ _filePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ }
+
+ // Dynamically create the instance
+ _filePlayerPtr
+ = FilePlayer::CreateFilePlayer(_filePlayerId,
+ (const FileFormats) format);
+
+ if (_filePlayerPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceWarning,
+ "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
+ return -1;
+ }
+
+ const uint32_t notificationTime(0);
+
+ if (_filePlayerPtr->StartPlayingFile(
+ (InStream&) *stream,
+ startPosition,
+ volumeScaling,
+ notificationTime,
+ stopPosition,
+ (const CodecInst*) codecInst) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartPlayingFile() failed to start file playout");
+ _filePlayerPtr->StopPlayingFile();
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ return -1;
+ }
+ _filePlayerPtr->RegisterModuleFileCallback(this);
+ _filePlaying = true;
+
+ return 0;
+}
+
+int TransmitMixer::StopPlayingFileAsMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
+ "TransmitMixer::StopPlayingFileAsMicrophone()");
+
+ if (!_filePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceWarning,
+ "StopPlayingFileAsMicrophone() isnot playing");
+ return 0;
+ }
+
+ CriticalSectionScoped cs(&_critSect);
+
+ if (_filePlayerPtr->StopPlayingFile() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_CANNOT_STOP_PLAYOUT, kTraceError,
+ "StopPlayingFile() couldnot stop playing file");
+ return -1;
+ }
+
+ _filePlayerPtr->RegisterModuleFileCallback(NULL);
+ FilePlayer::DestroyFilePlayer(_filePlayerPtr);
+ _filePlayerPtr = NULL;
+ _filePlaying = false;
+
+ return 0;
+}
+
+int TransmitMixer::IsPlayingFileAsMicrophone() const
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::IsPlayingFileAsMicrophone()");
+ return _filePlaying;
+}
+
+int TransmitMixer::ScaleFileAsMicrophonePlayout(float scale)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::ScaleFileAsMicrophonePlayout(scale=%5.3f)",
+ scale);
+
+ CriticalSectionScoped cs(&_critSect);
+
+ if (!_filePlaying)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_OPERATION, kTraceError,
+ "ScaleFileAsMicrophonePlayout() isnot playing file");
+ return -1;
+ }
+
+ if ((_filePlayerPtr == NULL) ||
+ (_filePlayerPtr->SetAudioScaling(scale) != 0))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "SetAudioScaling() failed to scale playout");
+ return -1;
+ }
+
+ return 0;
+}
+
+int TransmitMixer::StartRecordingMicrophone(const char* fileName,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartRecordingMicrophone(fileName=%s)",
+ fileName);
+
+ CriticalSectionScoped cs(&_critSect);
+
+ if (_fileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StartRecordingMicrophone() 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 < 0 || codecInst->channels > 2))
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_ARGUMENT, kTraceError,
+ "StartRecordingMicrophone() 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;
+ }
+
+ // Destroy the old instance
+ if (_fileRecorderPtr)
+ {
+ _fileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ }
+
+ _fileRecorderPtr =
+ FileRecorder::CreateFileRecorder(_fileRecorderId,
+ (const FileFormats) format);
+ if (_fileRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingMicrophone() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_fileRecorderPtr->StartRecordingAudioFile(
+ fileName,
+ (const CodecInst&) *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _fileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ return -1;
+ }
+ _fileRecorderPtr->RegisterModuleFileCallback(this);
+ _fileRecording = true;
+
+ return 0;
+}
+
+int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartRecordingMicrophone()");
+
+ CriticalSectionScoped cs(&_critSect);
+
+ if (_fileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StartRecordingMicrophone() 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,
+ "StartRecordingMicrophone() 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;
+ }
+
+ // Destroy the old instance
+ if (_fileRecorderPtr)
+ {
+ _fileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ }
+
+ _fileRecorderPtr =
+ FileRecorder::CreateFileRecorder(_fileRecorderId,
+ (const FileFormats) format);
+ if (_fileRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingMicrophone() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_fileRecorderPtr->StartRecordingAudioFile(*stream,
+ *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _fileRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ return -1;
+ }
+
+ _fileRecorderPtr->RegisterModuleFileCallback(this);
+ _fileRecording = true;
+
+ return 0;
+}
+
+
+int TransmitMixer::StopRecordingMicrophone()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StopRecordingMicrophone()");
+
+ CriticalSectionScoped cs(&_critSect);
+
+ if (!_fileRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StopRecordingMicrophone() isnot recording");
+ return 0;
+ }
+
+ if (_fileRecorderPtr->StopRecording() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STOP_RECORDING_FAILED, kTraceError,
+ "StopRecording(), could not stop recording");
+ return -1;
+ }
+ _fileRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
+ _fileRecorderPtr = NULL;
+ _fileRecording = false;
+
+ return 0;
+}
+
+int TransmitMixer::StartRecordingCall(const char* fileName,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartRecordingCall(fileName=%s)", fileName);
+
+ if (_fileCallRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StartRecordingCall() 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,
+ "StartRecordingCall() 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(&_critSect);
+
+ // Destroy the old instance
+ if (_fileCallRecorderPtr)
+ {
+ _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ }
+
+ _fileCallRecorderPtr
+ = FileRecorder::CreateFileRecorder(_fileCallRecorderId,
+ (const FileFormats) format);
+ if (_fileCallRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingCall() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_fileCallRecorderPtr->StartRecordingAudioFile(
+ fileName,
+ (const CodecInst&) *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _fileCallRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ return -1;
+ }
+ _fileCallRecorderPtr->RegisterModuleFileCallback(this);
+ _fileCallRecording = true;
+
+ return 0;
+}
+
+int TransmitMixer::StartRecordingCall(OutStream* stream,
+ const CodecInst* codecInst)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StartRecordingCall()");
+
+ if (_fileCallRecording)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "StartRecordingCall() 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,
+ "StartRecordingCall() 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(&_critSect);
+
+ // Destroy the old instance
+ if (_fileCallRecorderPtr)
+ {
+ _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ }
+
+ _fileCallRecorderPtr =
+ FileRecorder::CreateFileRecorder(_fileCallRecorderId,
+ (const FileFormats) format);
+ if (_fileCallRecorderPtr == NULL)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_INVALID_ARGUMENT, kTraceError,
+ "StartRecordingCall() fileRecorder format isnot correct");
+ return -1;
+ }
+
+ if (_fileCallRecorderPtr->StartRecordingAudioFile(*stream,
+ *codecInst,
+ notificationTime) != 0)
+ {
+ _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
+ "StartRecordingAudioFile() failed to start file recording");
+ _fileCallRecorderPtr->StopRecording();
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ return -1;
+ }
+
+ _fileCallRecorderPtr->RegisterModuleFileCallback(this);
+ _fileCallRecording = true;
+
+ return 0;
+}
+
+int TransmitMixer::StopRecordingCall()
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::StopRecordingCall()");
+
+ if (!_fileCallRecording)
+ {
+ WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
+ "StopRecordingCall() file isnot recording");
+ return -1;
+ }
+
+ CriticalSectionScoped cs(&_critSect);
+
+ if (_fileCallRecorderPtr->StopRecording() != 0)
+ {
+ _engineStatisticsPtr->SetLastError(
+ VE_STOP_RECORDING_FAILED, kTraceError,
+ "StopRecording(), could not stop recording");
+ return -1;
+ }
+
+ _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
+ FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
+ _fileCallRecorderPtr = NULL;
+ _fileCallRecording = false;
+
+ return 0;
+}
+
+void
+TransmitMixer::SetMixWithMicStatus(bool mix)
+{
+ _mixFileWithMicrophone = mix;
+}
+
+int TransmitMixer::RegisterExternalMediaProcessing(
+ VoEMediaProcess* object,
+ ProcessingTypes type) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RegisterExternalMediaProcessing()");
+
+ CriticalSectionScoped cs(&_callbackCritSect);
+ if (!object) {
+ return -1;
+ }
+
+ // Store the callback object according to the processing type.
+ if (type == kRecordingAllChannelsMixed) {
+ external_postproc_ptr_ = object;
+ } else if (type == kRecordingPreprocessing) {
+ external_preproc_ptr_ = object;
+ } else {
+ return -1;
+ }
+ return 0;
+}
+
+int TransmitMixer::DeRegisterExternalMediaProcessing(ProcessingTypes type) {
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::DeRegisterExternalMediaProcessing()");
+
+ CriticalSectionScoped cs(&_callbackCritSect);
+ if (type == kRecordingAllChannelsMixed) {
+ external_postproc_ptr_ = NULL;
+ } else if (type == kRecordingPreprocessing) {
+ external_preproc_ptr_ = NULL;
+ } else {
+ return -1;
+ }
+ return 0;
+}
+
+int
+TransmitMixer::SetMute(bool enable)
+{
+ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::SetMute(enable=%d)", enable);
+ _mute = enable;
+ return 0;
+}
+
+bool
+TransmitMixer::Mute() const
+{
+ return _mute;
+}
+
+int8_t TransmitMixer::AudioLevel() const
+{
+ // Speech + file level [0,9]
+ return _audioLevel.Level();
+}
+
+int16_t TransmitMixer::AudioLevelFullRange() const
+{
+ // Speech + file level [0,32767]
+ return _audioLevel.LevelFullRange();
+}
+
+bool TransmitMixer::IsRecordingCall()
+{
+ return _fileCallRecording;
+}
+
+bool TransmitMixer::IsRecordingMic()
+{
+ CriticalSectionScoped cs(&_critSect);
+ return _fileRecording;
+}
+
+void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
+ int samples_per_channel,
+ int num_channels,
+ int sample_rate_hz) {
+ int codec_rate;
+ int num_codec_channels;
+ GetSendCodecInfo(&codec_rate, &num_codec_channels);
+ // TODO(ajm): This currently restricts the sample rate to 32 kHz.
+ // See: https://code.google.com/p/webrtc/issues/detail?id=3146
+ // When 48 kHz is supported natively by AudioProcessing, this will have
+ // to be changed to handle 44.1 kHz.
+ codec_rate = std::min(codec_rate, kAudioProcMaxNativeSampleRateHz);
+ stereo_codec_ = num_codec_channels == 2;
+
+ if (!mono_buffer_.get()) {
+ // Temporary space for DownConvertToCodecFormat.
+ mono_buffer_.reset(new int16_t[kMaxMonoDataSizeSamples]);
+ }
+ DownConvertToCodecFormat(audio,
+ samples_per_channel,
+ num_channels,
+ sample_rate_hz,
+ num_codec_channels,
+ codec_rate,
+ mono_buffer_.get(),
+ &resampler_,
+ &_audioFrame);
+}
+
+int32_t TransmitMixer::RecordAudioToFile(
+ uint32_t mixingFrequency)
+{
+ CriticalSectionScoped cs(&_critSect);
+ if (_fileRecorderPtr == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordAudioToFile() filerecorder doesnot"
+ "exist");
+ return -1;
+ }
+
+ if (_fileRecorderPtr->RecordAudioToFile(_audioFrame) != 0)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::RecordAudioToFile() file recording"
+ "failed");
+ return -1;
+ }
+
+ return 0;
+}
+
+int32_t TransmitMixer::MixOrReplaceAudioWithFile(
+ int mixingFrequency)
+{
+ scoped_array<int16_t> fileBuffer(new int16_t[640]);
+
+ int fileSamples(0);
+ {
+ CriticalSectionScoped cs(&_critSect);
+ if (_filePlayerPtr == NULL)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice,
+ VoEId(_instanceId, -1),
+ "TransmitMixer::MixOrReplaceAudioWithFile()"
+ "fileplayer doesnot exist");
+ return -1;
+ }
+
+ if (_filePlayerPtr->Get10msAudioFromFile(fileBuffer.get(),
+ fileSamples,
+ mixingFrequency) == -1)
+ {
+ WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
+ "TransmitMixer::MixOrReplaceAudioWithFile() file"
+ " mixing failed");
+ return -1;
+ }
+ }
+
+ assert(_audioFrame.samples_per_channel_ == fileSamples);
+
+ if (_mixFileWithMicrophone)
+ {
+ // Currently file stream is always mono.
+ // TODO(xians): Change the code when FilePlayer supports real stereo.
+ MixWithSat(_audioFrame.data_,
+ _audioFrame.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.
+ _audioFrame.UpdateFrame(-1,
+ -1,
+ fileBuffer.get(),
+ fileSamples,
+ mixingFrequency,
+ AudioFrame::kNormalSpeech,
+ AudioFrame::kVadUnknown,
+ 1);
+ }
+ return 0;
+}
+
+void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
+ int current_mic_level, bool key_pressed) {
+ if (audioproc_->set_stream_delay_ms(delay_ms) != 0) {
+ // A redundant warning is reported in AudioDevice, which we've throttled
+ // to avoid flooding the logs. Relegate this one to LS_VERBOSE to avoid
+ // repeating the problem here.
+ LOG_FERR1(LS_VERBOSE, set_stream_delay_ms, delay_ms);
+ }
+
+ GainControl* agc = audioproc_->gain_control();
+ if (agc->set_stream_analog_level(current_mic_level) != 0) {
+ LOG_FERR1(LS_ERROR, set_stream_analog_level, current_mic_level);
+ assert(false);
+ }
+
+ EchoCancellation* aec = audioproc_->echo_cancellation();
+ if (aec->is_drift_compensation_enabled()) {
+ aec->set_stream_drift_samples(clock_drift);
+ }
+
+ audioproc_->set_stream_key_pressed(key_pressed);
+
+ int err = audioproc_->ProcessStream(&_audioFrame);
+ if (err != 0) {
+ LOG(LS_ERROR) << "ProcessStream() error: " << err;
+ assert(false);
+ }
+
+ // Store new capture level. Only updated when analog AGC is enabled.
+ _captureLevel = agc->stream_analog_level();
+
+ CriticalSectionScoped cs(&_critSect);
+ // Triggers a callback in OnPeriodicProcess().
+ _saturationWarning |= agc->stream_is_saturated();
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+void TransmitMixer::TypingDetection(bool keyPressed)
+{
+ // We let the VAD determine if we're using this feature or not.
+ if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) {
+ return;
+ }
+
+ bool vadActive = _audioFrame.vad_activity_ == AudioFrame::kVadActive;
+ if (_typingDetection.Process(keyPressed, vadActive)) {
+ _typingNoiseWarningPending = true;
+ _typingNoiseDetected = true;
+ } else {
+ // If there is already a warning pending, do not change the state.
+ // Otherwise set a warning pending if last callback was for noise detected.
+ if (!_typingNoiseWarningPending && _typingNoiseDetected) {
+ _typingNoiseWarningPending = true;
+ _typingNoiseDetected = false;
+ }
+ }
+}
+#endif
+
+int TransmitMixer::GetMixingFrequency()
+{
+ assert(_audioFrame.sample_rate_hz_ != 0);
+ return _audioFrame.sample_rate_hz_;
+}
+
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+int TransmitMixer::TimeSinceLastTyping(int &seconds)
+{
+ // We check in VoEAudioProcessingImpl that this is only called when
+ // typing detection is active.
+ seconds = _typingDetection.TimeSinceLastDetectionInSeconds();
+ return 0;
+}
+#endif
+
+#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
+int TransmitMixer::SetTypingDetectionParameters(int timeWindow,
+ int costPerTyping,
+ int reportingThreshold,
+ int penaltyDecay,
+ int typeEventDelay)
+{
+ _typingDetection.SetParameters(timeWindow,
+ costPerTyping,
+ reportingThreshold,
+ penaltyDecay,
+ typeEventDelay,
+ 0);
+ return 0;
+}
+#endif
+
+void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
+ swap_stereo_channels_ = enable;
+}
+
+bool TransmitMixer::IsStereoChannelSwappingEnabled() {
+ return swap_stereo_channels_;
+}
+
+} // namespace voe
+} // namespace webrtc
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-25 14:24:29 +0000