Chromium Code Reviews| 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; |
| } |