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/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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#define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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#include "api/array_view.h"
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#include "rtc_base/buffer.h"
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namespace webrtc {
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class AudioDeviceBuffer;
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// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with 16-bit PCM
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// audio samples corresponding to 10ms of data. It then allows for this data
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// to be pulled in a finer or coarser granularity. I.e. interacting with this
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// class instead of directly with the AudioDeviceBuffer one can ask for any
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// number of audio data samples. This class also ensures that audio data can be
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// delivered to the ADB in 10ms chunks when the size of the provided audio
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// buffers differs from 10ms.
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// As an example: calling DeliverRecordedData() with 5ms buffers will deliver
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// accumulated 10ms worth of data to the ADB every second call.
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class FineAudioBuffer {
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public:
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// |device_buffer| is a buffer that provides 10ms of audio data.
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FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer);
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~FineAudioBuffer();
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// Clears buffers and counters dealing with playout and/or recording.
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void ResetPlayout();
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void ResetRecord();
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// Utility methods which returns true if valid parameters are acquired at
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// constructions.
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bool IsReadyForPlayout() const;
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bool IsReadyForRecord() const;
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// Copies audio samples into |audio_buffer| where number of requested
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// elements is specified by |audio_buffer.size()|. The producer will always
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// fill up the audio buffer and if no audio exists, the buffer will contain
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// silence instead. The provided delay estimate in |playout_delay_ms| should
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// contain an estimate of the latency between when an audio frame is read from
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// WebRTC and when it is played out on the speaker.
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void GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
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int playout_delay_ms);
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// Consumes the audio data in |audio_buffer| and sends it to the WebRTC layer
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// in chunks of 10ms. The sum of the provided delay estimate in
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// |record_delay_ms| and the latest |playout_delay_ms| in GetPlayoutData()
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// are given to the AEC in the audio processing module.
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// They can be fixed values on most platforms and they are ignored if an
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// external (hardware/built-in) AEC is used.
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// Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
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// 5ms of data and sends a total of 10ms to WebRTC and clears the internal
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// cache. Call #3 restarts the scheme above.
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void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
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int record_delay_ms);
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private:
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// Device buffer that works with 10ms chunks of data both for playout and
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// for recording. I.e., the WebRTC side will always be asked for audio to be
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// played out in 10ms chunks and recorded audio will be sent to WebRTC in
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// 10ms chunks as well. This raw pointer is owned by the constructor of this
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// class and the owner must ensure that the pointer is valid during the life-
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// time of this object.
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AudioDeviceBuffer* const audio_device_buffer_;
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// Number of audio samples per channel per 10ms. Set once at construction
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// based on parameters in |audio_device_buffer|.
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const size_t playout_samples_per_channel_10ms_;
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const size_t record_samples_per_channel_10ms_;
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// Number of audio channels. Set once at construction based on parameters in
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// |audio_device_buffer|.
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const size_t playout_channels_;
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const size_t record_channels_;
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// Storage for output samples from which a consumer can read audio buffers
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// in any size using GetPlayoutData().
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rtc::BufferT<int16_t> playout_buffer_;
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// Storage for input samples that are about to be delivered to the WebRTC
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// ADB or remains from the last successful delivery of a 10ms audio buffer.
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rtc::BufferT<int16_t> record_buffer_;
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// Contains latest delay estimate given to GetPlayoutData().
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int playout_delay_ms_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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