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Reverting rev 929 due to failing assert on Linux.

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Failing at: audio_buffer.cc:159

TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/270008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@935 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2011-11-11 13:26:05 +00:00
parent 9b18ed6220
commit b351d6a8d8
4 changed files with 160 additions and 183 deletions
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2
src/voice_engine/main/source/channel.cc
View File

@ -5898,7 +5898,7 @@ Channel::EncodeAndSend()
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::EncodeAndSend()");
assert(_audioFrame._audioChannel <= 2);
assert(_audioFrame._audioChannel == 1);
if (_audioFrame._payloadDataLengthInSamples == 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),

9
src/voice_engine/main/source/transmit_mixer.cc
View File

@ -1158,13 +1158,9 @@ TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
"samplesPerSec=%u, mixingFrequency=%u)",
nSamples, samplesPerSec, mixingFrequency);
ResamplerType resampType = (nChannels == 1)?
kResamplerSynchronous : kResamplerSynchronousStereo;
if (_audioResampler.ResetIfNeeded(samplesPerSec,
mixingFrequency,
resampType) != 0)
kResamplerSynchronous) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
"TransmitMixer::GenerateAudioFrame() unable to resample");
@ -1172,7 +1168,7 @@ TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
}
if (_audioResampler.Push(
(WebRtc_Word16*) audioSamples,
nSamples * nChannels,
nSamples,
_audioFrame._payloadData,
AudioFrame::kMaxAudioFrameSizeSamples,
(int&) _audioFrame._payloadDataLengthInSamples) == -1)
@ -1182,7 +1178,6 @@ TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
return -1;
}
_audioFrame._payloadDataLengthInSamples /= nChannels;
_audioFrame._id = _instanceId;
_audioFrame._timeStamp = -1;
_audioFrame._frequencyInHz = mixingFrequency;

298
src/voice_engine/main/source/voe_base_impl.cc
View File

@ -159,6 +159,21 @@ WebRtc_Word32 VoEBaseImpl::RecordedDataIsAvailable(
assert(_transmitMixerPtr != NULL);
assert(_audioDevicePtr != NULL);
// Always use mono representation within VoE
if (nChannels == 2)
{
WebRtc_Word16* audio16ptr = (WebRtc_Word16*) audioSamples;
WebRtc_Word32 audio32;
for (WebRtc_UWord32 i = 0; i < nSamples; i++)
{
// y(i) = (1/2)*(x(2i) + x(2i+1)) => (1/2)*(left(i) + right(i))
audio32 = audio16ptr[2 * i];
audio32 += audio16ptr[2 * i + 1];
audio32 >>= 1;
audio16ptr[i] = static_cast<WebRtc_Word16> (audio32);
}
}
bool isAnalogAGC(false);
WebRtc_UWord32 maxVolume(0);
WebRtc_UWord16 currentVoEMicLevel(0);
@ -204,9 +219,12 @@ WebRtc_Word32 VoEBaseImpl::RecordedDataIsAvailable(
currentVoEMicLevel = (WebRtc_UWord16) _oldVoEMicLevel;
}
// Sending side only supports mono
const WebRtc_UWord8 nAudioChannels(1);
// Perform channel-independent operations
// (APM, mix with file, record to file, mute, etc.)
_transmitMixerPtr->PrepareDemux(audioSamples, nSamples, nChannels,
_transmitMixerPtr->PrepareDemux(audioSamples, nSamples, nAudioChannels,
samplesPerSec,
(WebRtc_UWord16) totalDelayMS, clockDrift,
currentVoEMicLevel);
@ -365,6 +383,129 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm)
}
}
// Create the AudioProcessing Module if it does not exist.
if (_audioProcessingModulePtr == NULL)
{
_audioProcessingModulePtr = AudioProcessing::Create(
VoEId(_instanceId, -1));
if (_audioProcessingModulePtr == NULL)
{
_engineStatistics.SetLastError(VE_NO_MEMORY, kTraceCritical,
"Init() failed to create the AP module");
return -1;
}
voe::Utility::TraceModuleVersion(VoEId(_instanceId, -1),
*_audioProcessingModulePtr);
// Ensure that mixers in both directions has access to the created APM
_transmitMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr);
_outputMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr);
if (_audioProcessingModulePtr->echo_cancellation()->
set_device_sample_rate_hz(
kVoiceEngineAudioProcessingDeviceSampleRateHz))
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set the device sample rate to 48K for AP "
" module");
return -1;
}
// Using 8 kHz as inital Fs. Might be changed already at first call.
if (_audioProcessingModulePtr->set_sample_rate_hz(8000))
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set the sample rate to 8K for AP module");
return -1;
}
if (_audioProcessingModulePtr->set_num_channels(1, 1) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
"Init() failed to set channels for the primary audio stream");
return -1;
}
if (_audioProcessingModulePtr->set_num_reverse_channels(1) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
"Init() failed to set channels for the primary audio stream");
return -1;
}
// high-pass filter
if (_audioProcessingModulePtr->high_pass_filter()->Enable(
WEBRTC_VOICE_ENGINE_HP_DEFAULT_STATE) != 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set the high-pass filter for AP module");
return -1;
}
// Echo Cancellation
if (_audioProcessingModulePtr->echo_cancellation()->
enable_drift_compensation(false) != 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set drift compensation for AP module");
return -1;
}
if (_audioProcessingModulePtr->echo_cancellation()->Enable(
WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE))
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set echo cancellation state for AP module");
return -1;
}
// Noise Reduction
if (_audioProcessingModulePtr->noise_suppression()->set_level(
(NoiseSuppression::Level) WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE)
!= 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set noise reduction level for AP module");
return -1;
}
if (_audioProcessingModulePtr->noise_suppression()->Enable(
WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE) != 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set noise reduction state for AP module");
return -1;
}
// Automatic Gain control
if (_audioProcessingModulePtr->gain_control()->set_analog_level_limits(
kMinVolumeLevel,kMaxVolumeLevel) != 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set AGC analog level for AP module");
return -1;
}
if (_audioProcessingModulePtr->gain_control()->set_mode(
(GainControl::Mode) WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE)
!= 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set AGC mode for AP module");
return -1;
}
if (_audioProcessingModulePtr->gain_control()->Enable(
WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE)
!= 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set AGC state for AP module");
return -1;
}
// VAD
if (_audioProcessingModulePtr->voice_detection()->Enable(
WEBRTC_VOICE_ENGINE_VAD_DEFAULT_STATE)
!= 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set VAD state for AP module");
return -1;
}
}
// Create and internal ADM if the user has not added and external
// ADM implementation as input to Init().
if (external_adm == NULL)
@ -471,148 +612,6 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm)
"Init() failed to initialize the microphone");
}
// Set number of channels
_audioDevicePtr->StereoPlayoutIsAvailable(&available);
if (_audioDevicePtr->SetStereoPlayout(available) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set mono/stereo playout mode");
}
_audioDevicePtr->StereoRecordingIsAvailable(&available);
if (_audioDevicePtr->SetStereoRecording(available) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set mono/stereo recording mode");
}
int recordingChannels = available ? 2 : 1;
// APM initialization done after sound card since we need
// to know if we support stereo recording or not.
// Create the AudioProcessing Module if it does not exist.
if (_audioProcessingModulePtr == NULL)
{
_audioProcessingModulePtr = AudioProcessing::Create(
VoEId(_instanceId, -1));
if (_audioProcessingModulePtr == NULL)
{
_engineStatistics.SetLastError(VE_NO_MEMORY, kTraceCritical,
"Init() failed to create the AP module");
return -1;
}
voe::Utility::TraceModuleVersion(VoEId(_instanceId, -1),
*_audioProcessingModulePtr);
// Ensure that mixers in both directions has access to the created APM
_transmitMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr);
_outputMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr);
if (_audioProcessingModulePtr->echo_cancellation()->
set_device_sample_rate_hz(
kVoiceEngineAudioProcessingDeviceSampleRateHz))
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set the device sample rate to 48K for AP "
" module");
return -1;
}
// Using 8 kHz as inital Fs. Might be changed already at first call.
if (_audioProcessingModulePtr->set_sample_rate_hz(8000))
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set the sample rate to 8K for AP module");
return -1;
}
// Assume mono sending until a send codec is set.
if (_audioProcessingModulePtr->set_num_channels(recordingChannels, 1) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
"Init() failed to set channels for the primary audio stream");
return -1;
}
if (_audioProcessingModulePtr->set_num_reverse_channels(1) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
"Init() failed to set channels for the primary audio stream");
return -1;
}
// high-pass filter
if (_audioProcessingModulePtr->high_pass_filter()->Enable(
WEBRTC_VOICE_ENGINE_HP_DEFAULT_STATE) != 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set the high-pass filter for AP module");
return -1;
}
// Echo Cancellation
if (_audioProcessingModulePtr->echo_cancellation()->
enable_drift_compensation(false) != 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set drift compensation for AP module");
return -1;
}
if (_audioProcessingModulePtr->echo_cancellation()->Enable(
WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE))
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set echo cancellation state for AP module");
return -1;
}
// Noise Reduction
if (_audioProcessingModulePtr->noise_suppression()->set_level(
(NoiseSuppression::Level) WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE)
!= 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set noise reduction level for AP module");
return -1;
}
if (_audioProcessingModulePtr->noise_suppression()->Enable(
WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE) != 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set noise reduction state for AP module");
return -1;
}
// Automatic Gain control
if (_audioProcessingModulePtr->gain_control()->set_analog_level_limits(
kMinVolumeLevel,kMaxVolumeLevel) != 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set AGC analog level for AP module");
return -1;
}
if (_audioProcessingModulePtr->gain_control()->set_mode(
(GainControl::Mode) WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE)
!= 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set AGC mode for AP module");
return -1;
}
if (_audioProcessingModulePtr->gain_control()->Enable(
WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE)
!= 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set AGC state for AP module");
return -1;
}
// VAD
if (_audioProcessingModulePtr->voice_detection()->Enable(
WEBRTC_VOICE_ENGINE_VAD_DEFAULT_STATE)
!= 0)
{
_engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
"Init() failed to set VAD state for AP module");
return -1;
}
}
// Set default AGC mode for the ADM
#ifdef WEBRTC_VOICE_ENGINE_AGC
bool enable(false);
@ -628,6 +627,19 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm)
}
}
#endif
// Set number of channels
_audioDevicePtr->StereoPlayoutIsAvailable(&available);
if (_audioDevicePtr->SetStereoPlayout(available) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set mono/stereo playout mode");
}
_audioDevicePtr->StereoRecordingIsAvailable(&available);
if (_audioDevicePtr->SetStereoRecording(available) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
"Init() failed to set mono/stereo recording mode");
}
return _engineStatistics.SetInitialized();
}

32
src/voice_engine/main/source/voe_codec_impl.cc
View File

@ -135,7 +135,7 @@ int VoECodecImpl::SetSendCodec(int channel, const CodecInst& codec)
"SetSendCodec() invalid codec name");
return -1;
}
if ((copyCodec.channels != 1) && (copyCodec.channels != 2))
if (copyCodec.channels != 1)
{
_engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"SetSendCodec() invalid number of "
@ -167,36 +167,6 @@ int VoECodecImpl::SetSendCodec(int channel, const CodecInst& codec)
return -1;
}
// Need to check if we should change APM settings for mono/stereo.
// We'll check all channels (sending or not), so we don't have to
// check this again when starting/stopping sending.
voe::ScopedChannel sc2(_channelManager);
void* iterator(NULL);
channelPtr = sc2.GetFirstChannel(iterator);
int maxNumChannels = 1;
while (channelPtr != NULL)
{
CodecInst tmpCdc;
channelPtr->GetSendCodec(tmpCdc);
if (tmpCdc.channels > maxNumChannels)
maxNumChannels = tmpCdc.channels;
channelPtr = sc2.GetNextChannel(iterator);
}
bool available(false);
_audioDevicePtr->StereoRecordingIsAvailable(&available);
int recordingChannels = available ? 2 : 1;
if (_audioProcessingModulePtr->set_num_channels(recordingChannels, maxNumChannels) != 0)
{
_engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
"Init() failed to set APM channels for the send audio stream");
return -1;
}
return 0;
}