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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <memory>
#include "webrtc/api/array_view.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioDeviceBuffer;
// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data
// corresponding to 10ms of data. It then allows for this data to be pulled in
// a finer or coarser granularity. I.e. interacting with this class instead of
// directly with the AudioDeviceBuffer one can ask for any number of audio data
// samples. This class also ensures that audio data can be delivered to the ADB
// in 10ms chunks when the size of the provided audio buffers differs from 10ms.
// As an example: calling DeliverRecordedData() with 5ms buffers will deliver
// accumulated 10ms worth of data to the ADB every second call.
// TODO(henrika): add support for stereo when mobile platforms need it.
class FineAudioBuffer {
// |device_buffer| is a buffer that provides 10ms of audio data.
// |sample_rate| is the sample rate of the audio data. This is needed because
// |device_buffer| delivers 10ms of data. Given the sample rate the number
// of samples can be calculated. The |capacity| ensures that the buffer size
// can be increased to at least capacity without further reallocation.
FineAudioBuffer(AudioDeviceBuffer* device_buffer,
int sample_rate,
size_t capacity);
// Clears buffers and counters dealing with playour and/or recording.
void ResetPlayout();
void ResetRecord();
// Copies audio samples into |audio_buffer| where number of requested
// elements is specified by |audio_buffer.size()|. The producer will always
// fill up the audio buffer and if no audio exists, the buffer will contain
// silence instead.
void GetPlayoutData(rtc::ArrayView<int8_t> audio_buffer);
// Consumes the audio data in |audio_buffer| and sends it to the WebRTC layer
// in chunks of 10ms. The provided delay estimates in |playout_delay_ms| and
// |record_delay_ms| are given to the AEC in the audio processing module.
// They can be fixed values on most platforms and they are ignored if an
// external (hardware/built-in) AEC is used.
// 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.
void DeliverRecordedData(rtc::ArrayView<const int8_t> audio_buffer,
int playout_delay_ms,
int record_delay_ms);
// Device buffer that works with 10ms chunks of data both for playout and
// for recording. I.e., the WebRTC side will always be asked for audio to be
// played out in 10ms chunks and recorded audio will be sent to WebRTC in
// 10ms chunks as well. This pointer is owned by the constructor of this
// class and the owner must ensure that the pointer is valid during the life-
// time of this object.
AudioDeviceBuffer* const device_buffer_;
// Sample rate in Hertz.
const int sample_rate_;
// Number of audio samples per 10ms.
const size_t samples_per_10_ms_;
// Number of audio bytes per 10ms.
const size_t bytes_per_10_ms_;
// Storage for output samples from which a consumer can read audio buffers
// in any size using GetPlayoutData().
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_;
} // namespace webrtc