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Handle 96 kHz when downmixing the capture path.

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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
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9
src/voice_engine/transmit_mixer.cc
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@@ -28,7 +28,9 @@ namespace webrtc {
namespace voe {
// 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
TransmitMixer::OnPeriodicProcess()
@@ -61,7 +63,7 @@ TransmitMixer::OnPeriodicProcess()
"TransmitMixer::OnPeriodicProcess() =>"
" CallbackOnError(VE_SATURATION_WARNING)");
_voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
}
}
_saturationWarning = 0;
}
@@ -1149,7 +1151,7 @@ bool TransmitMixer::IsRecordingMic()
return _fileRecording;
}
// TODO(andrew): use RemixAndResample for this.
int TransmitMixer::GenerateAudioFrame(const int16_t audio[],
int samples_per_channel,
int num_channels,
@@ -1157,6 +1159,7 @@ int TransmitMixer::GenerateAudioFrame(const int16_t audio[],
{
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_) {