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 febacbc8a69c2799316b1b484ff3c04c2060e743..2865fc6b6a6db82181177f2bf635da7677714f50 100644 |
--- a/webrtc/modules/audio_device/audio_device_buffer.cc |
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc |
@@ -406,12 +406,6 @@ int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer, |
return -1; |
} |
- if (nSamples != _recSamples) |
henrika_webrtc
2015/06/10 08:11:35
Not sure why these lines are removed. Care to elab
Peter Kasting
2015/06/11 04:31:41
On line 401 above we unconditionally set |_recSamp
|
- { |
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of recorded samples (%d)", nSamples); |
- return -1; |
- } |
- |
if (_recChannel == AudioDeviceModule::kChannelBoth) |
{ |
// (default) copy the complete input buffer to the local buffer |
@@ -576,8 +570,9 @@ int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer) |
if (_playSize > kMaxBufferSizeBytes) |
{ |
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "_playSize %i exceeds " |
- "kMaxBufferSizeBytes in AudioDeviceBuffer::GetPlayoutData", _playSize); |
+ WEBRTC_TRACE(kTraceError, kTraceUtility, _id, |
+ "_playSize %i exceeds kMaxBufferSizeBytes in " |
+ "AudioDeviceBuffer::GetPlayoutData", _playSize); |
assert(false); |
return -1; |
} |