最近公司的iPad項目中一個功能點涉及到了VOIP通訊中的錄音,需要在已有的WebRTC引擎中增加錄音功能,錄制通話雙方的聲音
參考了往上一位兄弟的博文(鏈接在此 http://blog.csdn.net/darkinger/article/details/13627479),代碼實現基本問題不大,就是由於WebRTC本身版本更新導致部分代碼要改動下結構;
但是那位兄台的代碼存在兩個問題
1 在一處地方沒有描述清楚,導致混音不可用.后面仔細跟蹤了下,調整下順序即可.
2 錄制出來的文件默認是PCM16K的裸數據,而不是通話使用的編碼方式,在這里走了一天彎路(還得怪自己懶,其實去看下代碼就知道了)
OK,以下是我做的全盤修改:
//////////////voe_file.h///////////////
基類增加兩個虛函數接口,用於啟停錄音調用
//DECWANG_4RECORD 20140814
virtual int StartRecordingPlayoutAndMic(const char* fileNameUTF8,
CodecInst* compression = NULL,
int maxSizeBytes = -1) = 0;
virtual int StopRecordingPlayoutAndMic() = 0;
/**/
//////////////voe_file_impl.h///////////////
子類定義增加兩個虛函數接口,用於啟停錄音調用
virtual int StartRecordingPlayoutAndMic(const char* fileNameUTF8,
CodecInst* compression = NULL,
int maxSizeBytes = -1);
virtual int StopRecordingPlayoutAndMic();
//////////////voe_file_impl.cc///////////////
int VoEFileImpl::StartRecordingPlayoutAndMic(const char* fileNameUTF8, CodecInst* compression,int maxSizeBytes)
{
//DECWANG_4RECORD,用於啟動錄音進程
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(),-1),
"StartRecordingPlayoutAndMic(fileNameUTF8=%s, "
"compression, maxSizeBytes=%d)",
fileNameUTF8, maxSizeBytes);
assert(1024 == FileWrapper::kMaxFileNameSize);
if (!_shared->statics()->Initialized())
{
_shared->statics()->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
_shared->outputall_mixer()->StartRecordingPlayout(fileNameUTF8, compression);
return 0;
}
int VoEFileImpl::StopRecordingPlayoutAndMic()
{
//DECWANG_4RECORD ,用於停止錄音進程
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(),-1),
"StopRecordingPlayoutAndMic");
if (!_shared->statics()->Initialized())
{
_shared->statics()->SetLastError(VE_NOT_INITED, kTraceError);
return -1;
}
return _shared->outputall_mixer()->StopRecordingPlayout();
}
//////////////share_data.h///////////////
增加一個處理雙向數據的混音對象,並增加對應的對象訪問接口
public:
//DECWANG_4RECORD
OutputMixer* outputall_mixer() { return _outputAllMixerPtr; }
Statistics* statics() {return &_engineStatistics;}
protected:
//DECWANG_4RECORD
OutputMixer* _outputAllMixerPtr; //for Record speaker+mic
//////////////share_data.cc///////////////
初始化
SharedData::SharedData() :
_instanceId(++_gInstanceCounter),
_apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
_channelManager(_gInstanceCounter),
_engineStatistics(_gInstanceCounter),
_audioDevicePtr(NULL),
audioproc_(NULL),
_moduleProcessThreadPtr(ProcessThread::CreateProcessThread()),
_externalRecording(false),
_externalPlayout(false)
{
Trace::CreateTrace();
if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
{
_outputMixerPtr->SetEngineInformation(_engineStatistics);
}
if (TransmitMixer::Create(_transmitMixerPtr, _gInstanceCounter) == 0)
{
_transmitMixerPtr->SetEngineInformation(*_moduleProcessThreadPtr,
_engineStatistics,
_channelManager);
}
//DECWANG_4RECORD
if (OutputMixer::Create(_outputAllMixerPtr, _gInstanceCounter) == 0)
{
_outputAllMixerPtr->SetEngineInformation(_engineStatistics);
}
/**/
_audioDeviceLayer = AudioDeviceModule::kPlatformDefaultAudio;
}
//////////////audio_conference_mixer_defines.h///////////////
增加一個處理混音的數據類
//DECWANG_4RECORD
//for Record speaker+mic
//record mic or playout signal from OutputMixer output
class AudioFrameMixerPart:public MixerParticipant
{
public:
AudioFrameMixerPart();
void SetAudioFrame(AudioFrame &audioFrame);
uint16_t GetPayloadDataLengthInSamples();
public:
// From MixerParticipant
int32_t GetAudioFrame(const int32_t id,AudioFrame& audioFrame);
int32_t NeededFrequency(const int32_t id);
private:
AudioFrame _audioFrame;
};
//////////////audio_conference_mixer_impl.cc///////////////
實現混音數據類的必備接口,原有代碼是在.h中實現,為了干凈,干脆直接全部移到cc中
AudioFrameMixerPart::AudioFrameMixerPart()
{
}
void AudioFrameMixerPart::SetAudioFrame(AudioFrame &audioFrame)
{
_audioFrame.CopyFrom(audioFrame);
}
uint16_t AudioFrameMixerPart::GetPayloadDataLengthInSamples()
{
return _audioFrame.samples_per_channel_;
}
int32_t AudioFrameMixerPart::GetAudioFrame(const int32_t id,AudioFrame& audioFrame)
{
if (_audioFrame.samples_per_channel_ <= 0)
return -1;
audioFrame.CopyFrom(_audioFrame);
return 0;
};
int32_t AudioFrameMixerPart::NeededFrequency(const int32_t id)
{
return _audioFrame.sample_rate_hz_;
}
//////////////output_mixer.h///////////////
增加一個共有成員函數,用於返回數據幀
public:
AudioFrame* GetAudioFrame();
//////////////output_mixer.cc///////////////
//DECWANG_4RECORD
AudioFrame* OutputMixer::GetAudioFrame()
{
return &_audioFrame;
}
//////////////transmit_mixer.h///////////////
增加成員函數
public:
AudioFrame* GetAudioFrame();
//////////////transmit_mixer.cc///////////////
AudioFrame* TransmitMixer::GetAudioFrame()
{
return &_audioFrame;
}
//////////////voe_base_impl.h///////////////
頭文件引用,增加
#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
私有成員
private:
AudioFrameMixerPart _afmTransmitMixer;
AudioFrameMixerPart _afmOutputMixer;
//////////////voe_base_impl.cc///////////////
int32_t VoEBaseImpl::RecordedDataIsAvailable(
const void* audioSamples,
uint32_t nSamples,
uint8_t nBytesPerSample,
uint8_t nChannels,
uint32_t samplesPerSec,
uint32_t totalDelayMS,
int32_t clockDrift,
uint32_t currentMicLevel,
bool keyPressed,
uint32_t& newMicLevel)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
"VoEBaseImpl::RecordedDataIsAvailable(nSamples=%u, "
"nBytesPerSample=%u, nChannels=%u, samplesPerSec=%u, "
"totalDelayMS=%u, clockDrift=%d, currentMicLevel=%u)",
nSamples, nBytesPerSample, nChannels, samplesPerSec,
totalDelayMS, clockDrift, currentMicLevel);
newMicLevel = static_cast<uint32_t>(ProcessRecordedDataWithAPM(
NULL, 0, audioSamples, samplesPerSec, nChannels, nSamples,
totalDelayMS, clockDrift, currentMicLevel, keyPressed));
//for Record speaker+mic
//DECWANG_4RECORD,用於拷貝已有的音頻幀,用於下一步的混音
_afmTransmitMixer.SetAudioFrame(*(_shared->transmit_mixer()->GetAudioFrame()));
return 0;
}
int32_t VoEBaseImpl::NeedMorePlayData(
uint32_t nSamples,
uint8_t nBytesPerSample,
uint8_t nChannels,
uint32_t samplesPerSec,
void* audioSamples,
uint32_t& nSamplesOut)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
"VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
"nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)",
nSamples, nBytesPerSample, nChannels, samplesPerSec);
assert(_shared->output_mixer() != NULL);
//DECWANG_4RECORD,獲取音頻幀,與資料出處的區別就在於此.
//for Record speaker+mic
_afmOutputMixer.SetAudioFrame(*(_shared->output_mixer()->GetAudioFrame()));
/**/
// TODO(andrew): if the device is running in mono, we should tell the mixer
// here so that it will only request mono from AudioCodingModule.
// Perform mixing of all active participants (channel-based mixing)
_shared->output_mixer()->MixActiveChannels();
// Additional operations on the combined signal
_shared->output_mixer()->DoOperationsOnCombinedSignal();
// Retrieve the final output mix (resampled to match the ADM)
_shared->output_mixer()->GetMixedAudio(samplesPerSec, nChannels,
&_audioFrame);
assert(static_cast<int>(nSamples) == _audioFrame.samples_per_channel_);
assert(samplesPerSec ==
static_cast<uint32_t>(_audioFrame.sample_rate_hz_));
// Deliver audio (PCM) samples to the ADM
memcpy(
(int16_t*) audioSamples,
(const int16_t*) _audioFrame.data_,
sizeof(int16_t) * (_audioFrame.samples_per_channel_
* _audioFrame.num_channels_));
nSamplesOut = _audioFrame.samples_per_channel_;
//DECWANG_4RECORD,用於實際混音動作
//for Record speaker+mic
if (_afmOutputMixer.GetPayloadDataLengthInSamples() == _afmTransmitMixer.GetPayloadDataLengthInSamples())
{
AudioFrame audioFrameX;
_shared->outputall_mixer()->MixActiveChannels();
_shared->outputall_mixer()->DoOperationsOnCombinedSignal();
_shared->outputall_mixer()->GetMixedAudio(samplesPerSec, nChannels, &audioFrameX);
}
/**/
return 0;
}
int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
AudioProcessing* audioproc)
{
WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
"Init(external_adm=0x%p)", external_adm);
CriticalSectionScoped cs(_shared->crit_sec());
WebRtcSpl_Init();
if (_shared->statistics().Initialized())
{
return 0;
}
if (_shared->process_thread())
{
if (_shared->process_thread()->Start() != 0)
{
_shared->SetLastError(VE_THREAD_ERROR, kTraceError,
"Init() failed to start module process thread");
return -1;
}
}
// Create an internal ADM if the user has not added an external
// ADM implementation as input to Init().
if (external_adm == NULL)
{
// Create the internal ADM implementation.
_shared->set_audio_device(AudioDeviceModuleImpl::Create(
VoEId(_shared->instance_id(), -1), _shared->audio_device_layer()));
if (_shared->audio_device() == NULL)
{
_shared->SetLastError(VE_NO_MEMORY, kTraceCritical,
"Init() failed to create the ADM");
return -1;
}
}
else
{
// Use the already existing external ADM implementation.
_shared->set_audio_device(external_adm);
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
"An external ADM implementation will be used in VoiceEngine");
}
// Register the ADM to the process thread, which will drive the error
// callback mechanism
if (_shared->process_thread() &&
_shared->process_thread()->RegisterModule(_shared->audio_device()) != 0)
{
_shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
"Init() failed to register the ADM");
return -1;
}
bool available(false);
// --------------------
// Reinitialize the ADM
// Register the AudioObserver implementation
if (_shared->audio_device()->RegisterEventObserver(this) != 0) {
_shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
"Init() failed to register event observer for the ADM");
}
// Register the AudioTransport implementation
if (_shared->audio_device()->RegisterAudioCallback(this) != 0) {
_shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
"Init() failed to register audio callback for the ADM");
}
// ADM initialization
if (_shared->audio_device()->Init() != 0)
{
_shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
"Init() failed to initialize the ADM");
return -1;
}
// Initialize the default speaker
if (_shared->audio_device()->SetPlayoutDevice(
WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0)
{
_shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo,
"Init() failed to set the default output device");
}
if (_shared->audio_device()->SpeakerIsAvailable(&available) != 0)
{
_shared->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
"Init() failed to check speaker availability, trying to "
"initialize speaker anyway");
}
else if (!available)
{
_shared->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
"Init() speaker not available, trying to initialize speaker "
"anyway");
}
if (_shared->audio_device()->InitSpeaker() != 0)
{
_shared->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
"Init() failed to initialize the speaker");
}
// Initialize the default microphone
if (_shared->audio_device()->SetRecordingDevice(
WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0)
{
_shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo,
"Init() failed to set the default input device");
}
if (_shared->audio_device()->MicrophoneIsAvailable(&available) != 0)
{
_shared->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
"Init() failed to check microphone availability, trying to "
"initialize microphone anyway");
}
else if (!available)
{
_shared->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
"Init() microphone not available, trying to initialize "
"microphone anyway");
}
if (_shared->audio_device()->InitMicrophone() != 0)
{
_shared->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
"Init() failed to initialize the microphone");
}
// Set number of channels
if (_shared->audio_device()->StereoPlayoutIsAvailable(&available) != 0) {
_shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to query stereo playout mode");
}
if (_shared->audio_device()->SetStereoPlayout(available) != 0)
{
_shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set mono/stereo playout mode");
}
// TODO(andrew): These functions don't tell us whether stereo recording
// is truly available. We simply set the AudioProcessing input to stereo
// here, because we have to wait until receiving the first frame to
// determine the actual number of channels anyway.
//
// These functions may be changed; tracked here:
// http://code.google.com/p/webrtc/issues/detail?id=204
_shared->audio_device()->StereoRecordingIsAvailable(&available);
if (_shared->audio_device()->SetStereoRecording(available) != 0)
{
_shared->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set mono/stereo recording mode");
}
if (!audioproc) {
audioproc = AudioProcessing::Create(VoEId(_shared->instance_id(), -1));
if (!audioproc) {
LOG(LS_ERROR) << "Failed to create AudioProcessing.";
_shared->SetLastError(VE_NO_MEMORY);
return -1;
}
}
_shared->set_audio_processing(audioproc);
/*DECWANG_4RECORD設置混音模塊調用指針
*/
_shared->outputall_mixer()->SetAudioProcessingModule(_shared->audio_processing());
_shared->outputall_mixer()->SetMixabilityStatus(_afmTransmitMixer, true);
_shared->outputall_mixer()->SetMixabilityStatus(_afmOutputMixer, true);
/**/
// Set the error state for any failures in this block.
_shared->SetLastError(VE_APM_ERROR);
if (audioproc->echo_cancellation()->set_device_sample_rate_hz(48000)) {
LOG_FERR1(LS_ERROR, set_device_sample_rate_hz, 48000);
return -1;
}
//DECWANG_ADD 20110620 FOR RECORD SYNC ,
// Assume 16 kHz mono until the audio frames are received from the capture
// device, at which point this can be updated.
if (audioproc->set_sample_rate_hz(16000)) {
LOG_FERR1(LS_ERROR, set_sample_rate_hz, 16000);
return -1;
}
if (audioproc->set_num_channels(1, 1) != 0) {
LOG_FERR2(LS_ERROR, set_num_channels, 1, 1);
return -1;
}
if (audioproc->set_num_reverse_channels(1) != 0) {
LOG_FERR1(LS_ERROR, set_num_reverse_channels, 1);
return -1;
}
// Configure AudioProcessing components. All are disabled by default.
if (audioproc->high_pass_filter()->Enable(true) != 0) {
LOG_FERR1(LS_ERROR, high_pass_filter()->Enable, true);
return -1;
}
if (audioproc->echo_cancellation()->enable_drift_compensation(false) != 0) {
LOG_FERR1(LS_ERROR, enable_drift_compensation, false);
return -1;
}
if (audioproc->noise_suppression()->set_level(kDefaultNsMode) != 0) {
LOG_FERR1(LS_ERROR, noise_suppression()->set_level, kDefaultNsMode);
return -1;
}
GainControl* agc = audioproc->gain_control();
if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
LOG_FERR2(LS_ERROR, agc->set_analog_level_limits, kMinVolumeLevel,
kMaxVolumeLevel);
return -1;
}
if (agc->set_mode(kDefaultAgcMode) != 0) {
LOG_FERR1(LS_ERROR, agc->set_mode, kDefaultAgcMode);
return -1;
}
if (agc->Enable(kDefaultAgcState) != 0) {
LOG_FERR1(LS_ERROR, agc->Enable, kDefaultAgcState);
return -1;
}
_shared->SetLastError(0); // Clear error state.
#ifdef WEBRTC_VOICE_ENGINE_AGC
bool agc_enabled = agc->mode() == GainControl::kAdaptiveAnalog &&
agc->is_enabled();
if (_shared->audio_device()->SetAGC(agc_enabled) != 0) {
LOG_FERR1(LS_ERROR, audio_device()->SetAGC, agc_enabled);
_shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR);
// TODO(ajm): No error return here due to
// https://code.google.com/p/webrtc/issues/detail?id=1464
}
#endif
return _shared->statistics().SetInitialized();
}
//////////////.h///////////////
//////////////.h///////////////
至此,結束,進行通話時就可以進行錄音了.下一篇將介紹錄制的文件如何提高語音質量和格式轉換的問題了.See you Next.
轉載請注明出處 decwang@qq.com