Index: webrtc/modules/audio_device/audio_device_buffer.h |
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h |
index f5f8f1d2207d0be59cb85a761081a70f10850b75..0e7a22dda5d9f51e08e50bfe474548df56d0e782 100644 |
--- a/webrtc/modules/audio_device/audio_device_buffer.h |
+++ b/webrtc/modules/audio_device/audio_device_buffer.h |
@@ -60,13 +60,13 @@ class AudioDeviceBuffer { |
int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const; |
virtual int32_t SetRecordedBuffer(const void* audio_buffer, |
- size_t num_samples); |
+ size_t samples_per_channel); |
int32_t SetCurrentMicLevel(uint32_t level); |
virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift); |
virtual int32_t DeliverRecordedData(); |
uint32_t NewMicLevel() const; |
- virtual int32_t RequestPlayoutData(size_t num_samples); |
+ virtual int32_t RequestPlayoutData(size_t samples_per_channel); |
virtual int32_t GetPlayoutData(void* audio_buffer); |
// TODO(henrika): these methods should not be used and does not contain any |
@@ -95,8 +95,8 @@ class AudioDeviceBuffer { |
// Updates counters in each play/record callback but does it on the task |
// queue to ensure that they can be read by LogStats() without any locks since |
// each task is serialized by the task queue. |
- void UpdateRecStats(int16_t max_abs, size_t num_samples); |
- void UpdatePlayStats(int16_t max_abs, size_t num_samples); |
+ void UpdateRecStats(int16_t max_abs, size_t samples_per_channel); |
+ void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel); |
// Clears all members tracking stats for recording and playout. |
// These methods both run on the task queue. |
@@ -154,12 +154,13 @@ class AudioDeviceBuffer { |
bool recording_ ACCESS_ON(main_thread_checker_); |
// Buffer used for audio samples to be played out. Size can be changed |
- // dynamically. |
- rtc::Buffer play_buffer_ ACCESS_ON(playout_thread_checker_); |
+ // dynamically. The 16-bit samples are interleaved, hence the size is |
+ // proportional to the number of channels. |
+ rtc::BufferT<int16_t> play_buffer_ ACCESS_ON(playout_thread_checker_); |
// Byte buffer used for recorded audio samples. Size can be changed |
// dynamically. |
- rtc::Buffer rec_buffer_ ACCESS_ON(recording_thread_checker_); |
+ rtc::BufferT<int16_t> rec_buffer_ ACCESS_ON(recording_thread_checker_); |
// AGC parameters. |
#if !defined(WEBRTC_WIN) |