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      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