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Simplifies FineAudioBuffer by using rtc::Buffer

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BUG=NONE

Review-Url: https://codereview.webrtc.org/2715963002
Cr-Commit-Position: refs/heads/master@{#16864}
This commit is contained in:
henrika
2017-02-27 05:14:17 -08:00
committed by Commit bot
parent 686aa37382
commit b3ebc1aa59
5 changed files with 38 additions and 88 deletions
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14
webrtc/modules/audio_device/fine_audio_buffer.h
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@ -42,10 +42,6 @@ class FineAudioBuffer {
int sample_rate);
~FineAudioBuffer();
// Returns the required size of |buffer| when calling GetPlayoutData(). If
// the buffer is smaller memory trampling will happen.
size_t RequiredPlayoutBufferSizeBytes();
// Clears buffers and counters dealing with playour and/or recording.
void ResetPlayout();
void ResetRecord();
@ -60,8 +56,7 @@ class FineAudioBuffer {
// They can be fixed values on most platforms and they are ignored if an
// external (hardware/built-in) AEC is used.
// The size of |buffer| is given by |size_in_bytes| and must be equal to
// |desired_frame_size_bytes_|. A RTC_CHECK will be hit if this is not the
// case.
// |desired_frame_size_bytes_|.
// Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
// 5ms of data and sends a total of 10ms to WebRTC and clears the intenal
// cache. Call #3 restarts the scheme above.
@ -87,12 +82,7 @@ class FineAudioBuffer {
const size_t samples_per_10_ms_;
// Number of audio bytes per 10ms.
const size_t bytes_per_10_ms_;
// Storage for output samples that are not yet asked for.
std::unique_ptr<int8_t[]> playout_cache_buffer_;
// Location of first unread output sample.
size_t playout_cached_buffer_start_;
// Number of bytes stored in output (contain samples to be played out) cache.
size_t playout_cached_bytes_;
rtc::BufferT<int8_t> playout_buffer_;
// Storage for input samples that are about to be delivered to the WebRTC
// ADB or remains from the last successful delivery of a 10ms audio buffer.
rtc::BufferT<int8_t> record_buffer_;