tor-browser

fine_audio_buffer.h (4671B)

---

/*
*  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 <cstddef>
#include <cstdint>
#include <optional>

#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(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(ArrayView<const int16_t> audio_buffer,
                          int record_delay_ms) {
   DeliverRecordedData(audio_buffer, record_delay_ms, std::nullopt);
 }
 void DeliverRecordedData(ArrayView<const int16_t> audio_buffer,
                          int record_delay_ms,
                          std::optional<int64_t> capture_time_ns);

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().
 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.
 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_