generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Handle 96 kHz when downmixing the capture path.

Browse Source 
BUG=issue721
TEST=96 kHz capture on Windows works.

Review URL: https://webrtc-codereview.appspot.com/722004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2558 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2012-08-03 18:03:02 +00:00
parent c0348fb349
commit 07ebdb9432
1 changed files with 12 additions and 9 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

21
src/voice_engine/transmit_mixer.cc
View File

@ -28,7 +28,9 @@ namespace webrtc {
namespace voe { namespace voe {
// Used for downmixing before resampling. // Used for downmixing before resampling.
static const int kMaxMonoDeviceDataSizeSamples = 480; // 10 ms, 48 kHz, mono. // TODO(andrew): audio_device should advertise the maximum sample rate it can
// provide.
static const int kMaxMonoDeviceDataSizeSamples = 960; // 10 ms, 96 kHz, mono.
void void
TransmitMixer::OnPeriodicProcess() TransmitMixer::OnPeriodicProcess()
@ -61,7 +63,7 @@ TransmitMixer::OnPeriodicProcess()
"TransmitMixer::OnPeriodicProcess() =>" "TransmitMixer::OnPeriodicProcess() =>"
" CallbackOnError(VE_SATURATION_WARNING)"); " CallbackOnError(VE_SATURATION_WARNING)");
_voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING); _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
} }
_saturationWarning = 0; _saturationWarning = 0;
} }
@ -89,7 +91,7 @@ void TransmitMixer::PlayNotification(const WebRtc_Word32 id,
// Not implement yet // Not implement yet
} }
void TransmitMixer::RecordNotification(const WebRtc_Word32 id, void TransmitMixer::RecordNotification(const WebRtc_Word32 id,
const WebRtc_UWord32 durationMs) const WebRtc_UWord32 durationMs)
{ {
@ -115,7 +117,7 @@ void TransmitMixer::PlayFileEnded(const WebRtc_Word32 id)
"file player module is shutdown"); "file player module is shutdown");
} }
void void
TransmitMixer::RecordFileEnded(const WebRtc_Word32 id) TransmitMixer::RecordFileEnded(const WebRtc_Word32 id)
{ {
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
@ -487,7 +489,7 @@ TransmitMixer::UpdateMuteMicrophoneTime(const WebRtc_UWord32 lengthMs)
_remainingMuteMicTimeMs = lengthMs; _remainingMuteMicTimeMs = lengthMs;
} }
WebRtc_Word32 WebRtc_Word32
TransmitMixer::StopSend() TransmitMixer::StopSend()
{ {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
@ -578,7 +580,7 @@ int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
"TransmitMixer::StartPlayingFileAsMicrophone(format=%d," "TransmitMixer::StartPlayingFileAsMicrophone(format=%d,"
" volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
format, volumeScaling, startPosition, stopPosition); format, volumeScaling, startPosition, stopPosition);
if (stream == NULL) if (stream == NULL)
{ {
_engineStatisticsPtr->SetLastError( _engineStatisticsPtr->SetLastError(
@ -1043,7 +1045,7 @@ int TransmitMixer::StartRecordingCall(OutStream* stream,
_fileCallRecorderPtr = NULL; _fileCallRecorderPtr = NULL;
return -1; return -1;
} }
_fileCallRecorderPtr->RegisterModuleFileCallback(this); _fileCallRecorderPtr->RegisterModuleFileCallback(this);
_fileCallRecording = true; _fileCallRecording = true;
@ -1080,7 +1082,7 @@ int TransmitMixer::StopRecordingCall()
return 0; return 0;
} }
void void
TransmitMixer::SetMixWithMicStatus(bool mix) TransmitMixer::SetMixWithMicStatus(bool mix)
{ {
_mixFileWithMicrophone = mix; _mixFileWithMicrophone = mix;
@ -1149,7 +1151,7 @@ bool TransmitMixer::IsRecordingMic()
return _fileRecording; return _fileRecording;
} }
// TODO(andrew): use RemixAndResample for this.
int TransmitMixer::GenerateAudioFrame(const int16_t audio[], int TransmitMixer::GenerateAudioFrame(const int16_t audio[],
int samples_per_channel, int samples_per_channel,
int num_channels, int num_channels,
@ -1157,6 +1159,7 @@ int TransmitMixer::GenerateAudioFrame(const int16_t audio[],
{ {
const int16_t* audio_ptr = audio; const int16_t* audio_ptr = audio;
int16_t mono_audio[kMaxMonoDeviceDataSizeSamples]; int16_t mono_audio[kMaxMonoDeviceDataSizeSamples];
assert(samples_per_channel <= kMaxMonoDeviceDataSizeSamples);
// If no stereo codecs are in use, we downmix a stereo stream from the // If no stereo codecs are in use, we downmix a stereo stream from the
// device early in the chain, before resampling. // device early in the chain, before resampling.
if (num_channels == 2 && !stereo_codec_) { if (num_channels == 2 && !stereo_codec_) {