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 79a0ebc3b69cbecb0a837b64e9f4495c0e7deaef..063c65f4fc120febd6da7db6fc800aaf23f4775f 100644 |
--- a/webrtc/modules/audio_device/audio_device_buffer.cc |
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc |
@@ -376,8 +376,8 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) { |
} |
// Update playout stats which is used as base for periodic logging of the |
// audio output state. |
- UpdatePlayStats(max_abs, num_samples_out); |
- return static_cast<int32_t>(num_samples_out); |
+ UpdatePlayStats(max_abs, num_samples_out / play_channels_); |
+ return static_cast<int32_t>(num_samples_out / play_channels_); |
} |
int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) { |