1 /* 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ 12 #define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ 13 14 #include "api/array_view.h" 15 #include "rtc_base/buffer.h" 16 17 namespace webrtc { 18 19 class AudioDeviceBuffer; 20 21 // FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with 16-bit PCM 22 // audio samples corresponding to 10ms of data. It then allows for this data 23 // to be pulled in a finer or coarser granularity. I.e. interacting with this 24 // class instead of directly with the AudioDeviceBuffer one can ask for any 25 // number of audio data samples. This class also ensures that audio data can be 26 // delivered to the ADB in 10ms chunks when the size of the provided audio 27 // buffers differs from 10ms. 28 // As an example: calling DeliverRecordedData() with 5ms buffers will deliver 29 // accumulated 10ms worth of data to the ADB every second call. 30 class FineAudioBuffer { 31 public: 32 // |device_buffer| is a buffer that provides 10ms of audio data. 33 FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer); 34 ~FineAudioBuffer(); 35 36 // Clears buffers and counters dealing with playout and/or recording. 37 void ResetPlayout(); 38 void ResetRecord(); 39 40 // Utility methods which returns true if valid parameters are acquired at 41 // constructions. 42 bool IsReadyForPlayout() const; 43 bool IsReadyForRecord() const; 44 45 // Copies audio samples into |audio_buffer| where number of requested 46 // elements is specified by |audio_buffer.size()|. The producer will always 47 // fill up the audio buffer and if no audio exists, the buffer will contain 48 // silence instead. The provided delay estimate in |playout_delay_ms| should 49 // contain an estimate of the latency between when an audio frame is read from 50 // WebRTC and when it is played out on the speaker. 51 void GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer, 52 int playout_delay_ms); 53 54 // Consumes the audio data in |audio_buffer| and sends it to the WebRTC layer 55 // in chunks of 10ms. The sum of the provided delay estimate in 56 // |record_delay_ms| and the latest |playout_delay_ms| in GetPlayoutData() 57 // are given to the AEC in the audio processing module. 58 // They can be fixed values on most platforms and they are ignored if an 59 // external (hardware/built-in) AEC is used. 60 // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores 61 // 5ms of data and sends a total of 10ms to WebRTC and clears the internal 62 // cache. Call #3 restarts the scheme above. 63 void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer, 64 int record_delay_ms); 65 66 private: 67 // Device buffer that works with 10ms chunks of data both for playout and 68 // for recording. I.e., the WebRTC side will always be asked for audio to be 69 // played out in 10ms chunks and recorded audio will be sent to WebRTC in 70 // 10ms chunks as well. This raw pointer is owned by the constructor of this 71 // class and the owner must ensure that the pointer is valid during the life- 72 // time of this object. 73 AudioDeviceBuffer* const audio_device_buffer_; 74 // Number of audio samples per channel per 10ms. Set once at construction 75 // based on parameters in |audio_device_buffer|. 76 const size_t playout_samples_per_channel_10ms_; 77 const size_t record_samples_per_channel_10ms_; 78 // Number of audio channels. Set once at construction based on parameters in 79 // |audio_device_buffer|. 80 const size_t playout_channels_; 81 const size_t record_channels_; 82 // Storage for output samples from which a consumer can read audio buffers 83 // in any size using GetPlayoutData(). 84 rtc::BufferT<int16_t> playout_buffer_; 85 // Storage for input samples that are about to be delivered to the WebRTC 86 // ADB or remains from the last successful delivery of a 10ms audio buffer. 87 rtc::BufferT<int16_t> record_buffer_; 88 // Contains latest delay estimate given to GetPlayoutData(). 89 int playout_delay_ms_ = 0; 90 }; 91 92 } // namespace webrtc 93 94 #endif // MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ 95