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7120742701
Now that we have moved WebRTC from src/webrtc to src/, common_types.h and typedefs.h are triggering a cpplint error. The cpplint complaint is: Include the directory when naming .h files [build/include] [4] This CL disables the error but we have to remove these two headers from the root directory. NOPRESUBMIT=true Bug: webrtc:5876 Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333 Reviewed-on: https://webrtc-review.googlesource.com/1577 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19859}
91 lines
4.1 KiB
C++
91 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 MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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#define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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#include <memory>
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#include "api/array_view.h"
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#include "rtc_base/buffer.h"
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#include "typedefs.h" // NOLINT(build/include)
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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 // MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
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