Index: webrtc/modules/audio_device/audio_device_buffer.cc |
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc |
index 12b28b3ab3f7cafaa9ce9b568864d6f69430d66c..3cfbc7d09c4da6d7f9fb88850e76ae9db104af05 100644 |
--- a/webrtc/modules/audio_device/audio_device_buffer.cc |
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc |
@@ -130,8 +130,6 @@ int32_t AudioDeviceBuffer::InitRecording() |
int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) |
{ |
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingSampleRate(fsHz=%u)", fsHz); |
- |
CriticalSectionScoped lock(&_critSect); |
_recSampleRate = fsHz; |
return 0; |
@@ -143,8 +141,6 @@ int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) |
int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) |
{ |
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutSampleRate(fsHz=%u)", fsHz); |
- |
CriticalSectionScoped lock(&_critSect); |
_playSampleRate = fsHz; |
return 0; |
@@ -174,8 +170,6 @@ int32_t AudioDeviceBuffer::PlayoutSampleRate() const |
int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels) |
{ |
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingChannels(channels=%u)", channels); |
- |
CriticalSectionScoped lock(&_critSect); |
_recChannels = channels; |
_recBytesPerSample = 2*channels; // 16 bits per sample in mono, 32 bits in stereo |
@@ -188,8 +182,6 @@ int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels) |
int32_t AudioDeviceBuffer::SetPlayoutChannels(uint8_t channels) |
{ |
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutChannels(channels=%u)", channels); |
- |
CriticalSectionScoped lock(&_critSect); |
_playChannels = channels; |
// 16 bits per sample in mono, 32 bits in stereo |