95 lines
4.3 KiB
C++
95 lines
4.3 KiB
C++
/*
|
||
* 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.
|
||
*/
|
||
|
||
#ifndef MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
|
||
#define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
|
||
|
||
#include "api/array_view.h"
|
||
#include "rtc_base/buffer.h"
|
||
|
||
namespace webrtc {
|
||
|
||
class AudioDeviceBuffer;
|
||
|
||
// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with 16-bit PCM
|
||
// audio samples 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.
|
||
class FineAudioBuffer {
|
||
public:
|
||
// |device_buffer| is a buffer that provides 10ms of audio data.
|
||
FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer);
|
||
~FineAudioBuffer();
|
||
|
||
// Clears buffers and counters dealing with playout and/or recording.
|
||
void ResetPlayout();
|
||
void ResetRecord();
|
||
|
||
// Utility methods which returns true if valid parameters are acquired at
|
||
// constructions.
|
||
bool IsReadyForPlayout() const;
|
||
bool IsReadyForRecord() const;
|
||
|
||
// 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. The provided delay estimate in |playout_delay_ms| should
|
||
// contain an estimate of the latency between when an audio frame is read from
|
||
// WebRTC and when it is played out on the speaker.
|
||
void GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
|
||
int playout_delay_ms);
|
||
|
||
// Consumes the audio data in |audio_buffer| and sends it to the WebRTC layer
|
||
// in chunks of 10ms. The sum of the provided delay estimate in
|
||
// |record_delay_ms| and the latest |playout_delay_ms| in GetPlayoutData()
|
||
// 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 internal
|
||
// cache. Call #3 restarts the scheme above.
|
||
void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
|
||
int record_delay_ms);
|
||
|
||
private:
|
||
// 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 raw 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 audio_device_buffer_;
|
||
// Number of audio samples per channel per 10ms. Set once at construction
|
||
// based on parameters in |audio_device_buffer|.
|
||
const size_t playout_samples_per_channel_10ms_;
|
||
const size_t record_samples_per_channel_10ms_;
|
||
// Number of audio channels. Set once at construction based on parameters in
|
||
// |audio_device_buffer|.
|
||
const size_t playout_channels_;
|
||
const size_t record_channels_;
|
||
// Storage for output samples from which a consumer can read audio buffers
|
||
// in any size using GetPlayoutData().
|
||
rtc::BufferT<int16_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<int16_t> record_buffer_;
|
||
// Contains latest delay estimate given to GetPlayoutData().
|
||
int playout_delay_ms_ = 0;
|
||
};
|
||
|
||
} // namespace webrtc
|
||
|
||
#endif // MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
|