Index: webrtc/modules/audio_device/dummy/file_audio_device.cc |
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc |
index ea432af203be16dfe905af421a72efe695a66878..ea61ce64a7072af25a7e881b26e491972b8f5c2d 100644 |
--- a/webrtc/modules/audio_device/dummy/file_audio_device.cc |
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc |
@@ -17,10 +17,10 @@ const int kRecordingFixedSampleRate = 48000; |
const int kRecordingNumChannels = 2; |
const int kPlayoutFixedSampleRate = 48000; |
const int kPlayoutNumChannels = 2; |
-const int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100 |
- * kPlayoutNumChannels * 2; |
-const int kRecordingBufferSize = kRecordingFixedSampleRate / 100 |
- * kRecordingNumChannels * 2; |
+const size_t kPlayoutBufferSize = |
+ kPlayoutFixedSampleRate / 100 * kPlayoutNumChannels * 2; |
+const size_t kRecordingBufferSize = |
+ kRecordingFixedSampleRate / 100 * kRecordingNumChannels * 2; |
FileAudioDevice::FileAudioDevice(const int32_t id, |
const char* inputFilename, |
@@ -194,9 +194,7 @@ int32_t FileAudioDevice::StartPlayout() { |
_playoutFramesLeft = 0; |
if (!_playoutBuffer) { |
- _playoutBuffer = new int8_t[2 * |
- kPlayoutNumChannels * |
- kPlayoutFixedSampleRate/100]; |
+ _playoutBuffer = new int8_t[kPlayoutBufferSize]; |
} |
if (!_playoutBuffer) { |
_playing = false; |