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 WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ 12 #define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ 13 14 #include "webrtc/base/scoped_ptr.h" 15 #include "webrtc/typedefs.h" 16 17 namespace webrtc { 18 19 class AudioDeviceBuffer; 20 21 // FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data 22 // corresponding to 10ms of data. It then allows for this data to be pulled in 23 // a finer or coarser granularity. I.e. interacting with this class instead of 24 // directly with the AudioDeviceBuffer one can ask for any number of audio data 25 // samples. This class also ensures that audio data can be delivered to the ADB 26 // in 10ms chunks when the size of the provided audio buffers differs from 10ms. 27 // As an example: calling DeliverRecordedData() with 5ms buffers will deliver 28 // accumulated 10ms worth of data to the ADB every second call. 29 class FineAudioBuffer { 30 public: 31 // |device_buffer| is a buffer that provides 10ms of audio data. 32 // |desired_frame_size_bytes| is the number of bytes of audio data 33 // GetPlayoutData() should return on success. It is also the required size of 34 // each recorded buffer used in DeliverRecordedData() calls. 35 // |sample_rate| is the sample rate of the audio data. This is needed because 36 // |device_buffer| delivers 10ms of data. Given the sample rate the number 37 // of samples can be calculated. 38 FineAudioBuffer(AudioDeviceBuffer* device_buffer, 39 size_t desired_frame_size_bytes, 40 int sample_rate); 41 ~FineAudioBuffer(); 42 43 // Returns the required size of |buffer| when calling GetPlayoutData(). If 44 // the buffer is smaller memory trampling will happen. 45 size_t RequiredPlayoutBufferSizeBytes(); 46 47 // Clears buffers and counters dealing with playour and/or recording. 48 void ResetPlayout(); 49 void ResetRecord(); 50 51 // |buffer| must be of equal or greater size than what is returned by 52 // RequiredBufferSize(). This is to avoid unnecessary memcpy. 53 void GetPlayoutData(int8_t* buffer); 54 55 // Consumes the audio data in |buffer| and sends it to the WebRTC layer in 56 // chunks of 10ms. The provided delay estimates in |playout_delay_ms| and 57 // |record_delay_ms| 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 // The size of |buffer| is given by |size_in_bytes| and must be equal to 61 // |desired_frame_size_bytes_|. A RTC_CHECK will be hit if this is not the 62 // case. 63 // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores 64 // 5ms of data and sends a total of 10ms to WebRTC and clears the intenal 65 // cache. Call #3 restarts the scheme above. 66 void DeliverRecordedData(const int8_t* buffer, 67 size_t size_in_bytes, 68 int playout_delay_ms, 69 int record_delay_ms); 70 71 private: 72 // Device buffer that works with 10ms chunks of data both for playout and 73 // for recording. I.e., the WebRTC side will always be asked for audio to be 74 // played out in 10ms chunks and recorded audio will be sent to WebRTC in 75 // 10ms chunks as well. This pointer is owned by the constructor of this 76 // class and the owner must ensure that the pointer is valid during the life- 77 // time of this object. 78 AudioDeviceBuffer* const device_buffer_; 79 // Number of bytes delivered by GetPlayoutData() call and provided to 80 // DeliverRecordedData(). 81 const size_t desired_frame_size_bytes_; 82 // Sample rate in Hertz. 83 const int sample_rate_; 84 // Number of audio samples per 10ms. 85 const size_t samples_per_10_ms_; 86 // Number of audio bytes per 10ms. 87 const size_t bytes_per_10_ms_; 88 // Storage for output samples that are not yet asked for. 89 rtc::scoped_ptr<int8_t[]> playout_cache_buffer_; 90 // Location of first unread output sample. 91 size_t playout_cached_buffer_start_; 92 // Number of bytes stored in output (contain samples to be played out) cache. 93 size_t playout_cached_bytes_; 94 // Storage for input samples that are about to be delivered to the WebRTC 95 // ADB or remains from the last successful delivery of a 10ms audio buffer. 96 rtc::scoped_ptr<int8_t[]> record_cache_buffer_; 97 // Required (max) size in bytes of the |record_cache_buffer_|. 98 const size_t required_record_buffer_size_bytes_; 99 // Number of bytes in input (contains recorded samples) cache. 100 size_t record_cached_bytes_; 101 // Read and write pointers used in the buffering scheme on the recording side. 102 size_t record_read_pos_; 103 size_t record_write_pos_; 104 }; 105 106 } // namespace webrtc 107 108 #endif // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ 109