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_ANDROID_FINE_AUDIO_BUFFER_H_ 12 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_ 13 14 #include "webrtc/system_wrappers/interface/scoped_ptr.h" 15 #include "webrtc/typedefs.h" 16 17 namespace webrtc { 18 19 class AudioDeviceBuffer; 20 21 // FineAudioBuffer takes an AudioDeviceBuffer which delivers 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. 26 class FineAudioBuffer { 27 public: 28 // |device_buffer| is a buffer that provides 10ms of audio data. 29 // |desired_frame_size_bytes| is the number of bytes of audio data 30 // (not samples) |GetBufferData| should return on success. 31 // |sample_rate| is the sample rate of the audio data. This is needed because 32 // |device_buffer| delivers 10ms of data. Given the sample rate the number 33 // of samples can be calculated. 34 FineAudioBuffer(AudioDeviceBuffer* device_buffer, 35 int desired_frame_size_bytes, 36 int sample_rate); 37 ~FineAudioBuffer(); 38 39 // Returns the required size of |buffer| when calling GetBufferData. If the 40 // buffer is smaller memory trampling will happen. 41 // |desired_frame_size_bytes| and |samples_rate| are as described in the 42 // constructor. 43 int RequiredBufferSizeBytes(); 44 45 // |buffer| must be of equal or greater size than what is returned by 46 // RequiredBufferSize. This is to avoid unnecessary memcpy. 47 void GetBufferData(int8_t* buffer); 48 49 private: 50 // Device buffer that provides 10ms chunks of data. 51 AudioDeviceBuffer* device_buffer_; 52 int desired_frame_size_bytes_; // Number of bytes delivered per GetBufferData 53 int sample_rate_; 54 int samples_per_10_ms_; 55 // Convenience parameter to avoid converting from samples 56 int bytes_per_10_ms_; 57 58 // Storage for samples that are not yet asked for. 59 scoped_ptr<int8_t[]> cache_buffer_; 60 int cached_buffer_start_; // Location of first unread sample. 61 int cached_bytes_; // Number of bytes stored in cache. 62 }; 63 64 } // namespace webrtc 65 66 #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_FINE_AUDIO_BUFFER_H_ 67
