| /* |
| * 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 WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ |
| |
| #include <memory> |
| |
| #include "webrtc/rtc_base/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 { |
| public: |
| // |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); |
| ~FineAudioBuffer(); |
| |
| // 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); |
| |
| 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 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 |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ |