bb6f7524ba
This change: Reduces complexity for audio playout by removing a redundant memcopy in the output audio path. Adds support for iOS simulator for playout since we now allow the audio layer to ask for different sizes of audio buffers at each callback. Real iOS devices always asks for the same size, simulators does not. This change comes without any new cost for real devices. BUG=b/37580746 Review-Url: https://codereview.webrtc.org/2894873002 Cr-Commit-Position: refs/heads/master@{#18321}
92 lines
4.1 KiB
C++
92 lines
4.1 KiB
C++
/*
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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 WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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#include <memory>
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#include "webrtc/base/array_view.h"
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#include "webrtc/base/buffer.h"
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#include "webrtc/typedefs.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 audio data
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// corresponding to 10ms of data. It then allows for this data to be pulled in
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// a finer or coarser granularity. I.e. interacting with this class instead of
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// directly with the AudioDeviceBuffer one can ask for any number of audio data
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// samples. This class also ensures that audio data can be delivered to the ADB
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// in 10ms chunks when the size of the provided audio 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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// TODO(henrika): add support for stereo when mobile platforms need it.
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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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// |sample_rate| is the sample rate of the audio data. This is needed because
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// |device_buffer| delivers 10ms of data. Given the sample rate the number
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// of samples can be calculated. The |capacity| ensures that the buffer size
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// can be increased to at least capacity without further reallocation.
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FineAudioBuffer(AudioDeviceBuffer* device_buffer,
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int sample_rate,
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size_t capacity);
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~FineAudioBuffer();
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// Clears buffers and counters dealing with playour and/or recording.
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void ResetPlayout();
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void ResetRecord();
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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.
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void GetPlayoutData(rtc::ArrayView<int8_t> audio_buffer);
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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 provided delay estimates in |playout_delay_ms| and
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// |record_delay_ms| 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 intenal
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// cache. Call #3 restarts the scheme above.
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void DeliverRecordedData(rtc::ArrayView<const int8_t> audio_buffer,
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int playout_delay_ms,
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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 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 device_buffer_;
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// Sample rate in Hertz.
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const int sample_rate_;
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// Number of audio samples per 10ms.
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const size_t samples_per_10_ms_;
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// Number of audio bytes per 10ms.
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const size_t bytes_per_10_ms_;
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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<int8_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<int8_t> record_buffer_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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