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 #include "webrtc/modules/audio_device/android/fine_audio_buffer.h" 12 13 #include <memory.h> 14 #include <stdio.h> 15 #include <algorithm> 16 17 #include "webrtc/modules/audio_device/audio_device_buffer.h" 18 19 namespace webrtc { 20 21 FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer, 22 int desired_frame_size_bytes, 23 int sample_rate) 24 : device_buffer_(device_buffer), 25 desired_frame_size_bytes_(desired_frame_size_bytes), 26 sample_rate_(sample_rate), 27 samples_per_10_ms_(sample_rate_ * 10 / 1000), 28 bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)), 29 cached_buffer_start_(0), 30 cached_bytes_(0) { 31 cache_buffer_.reset(new int8_t[bytes_per_10_ms_]); 32 } 33 34 FineAudioBuffer::~FineAudioBuffer() { 35 } 36 37 int FineAudioBuffer::RequiredBufferSizeBytes() { 38 // It is possible that we store the desired frame size - 1 samples. Since new 39 // audio frames are pulled in chunks of 10ms we will need a buffer that can 40 // hold desired_frame_size - 1 + 10ms of data. We omit the - 1. 41 return desired_frame_size_bytes_ + bytes_per_10_ms_; 42 } 43 44 void FineAudioBuffer::GetBufferData(int8_t* buffer) { 45 if (desired_frame_size_bytes_ <= cached_bytes_) { 46 memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_], 47 desired_frame_size_bytes_); 48 cached_buffer_start_ += desired_frame_size_bytes_; 49 cached_bytes_ -= desired_frame_size_bytes_; 50 assert(cached_buffer_start_ + cached_bytes_ < bytes_per_10_ms_); 51 return; 52 } 53 memcpy(buffer, &cache_buffer_.get()[cached_buffer_start_], cached_bytes_); 54 // Push another n*10ms of audio to |buffer|. n > 1 if 55 // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we 56 // write the audio after the cached bytes copied earlier. 57 int8_t* unwritten_buffer = &buffer[cached_bytes_]; 58 int bytes_left = desired_frame_size_bytes_ - cached_bytes_; 59 // Ceiling of integer division: 1 + ((x - 1) / y) 60 int number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_); 61 for (int i = 0; i < number_of_requests; ++i) { 62 device_buffer_->RequestPlayoutData(samples_per_10_ms_); 63 int num_out = device_buffer_->GetPlayoutData(unwritten_buffer); 64 if (num_out != samples_per_10_ms_) { 65 assert(num_out == 0); 66 cached_bytes_ = 0; 67 return; 68 } 69 unwritten_buffer += bytes_per_10_ms_; 70 assert(bytes_left >= 0); 71 bytes_left -= bytes_per_10_ms_; 72 } 73 assert(bytes_left <= 0); 74 // Put the samples that were written to |buffer| but are not used in the 75 // cache. 76 int cache_location = desired_frame_size_bytes_; 77 int8_t* cache_ptr = &buffer[cache_location]; 78 cached_bytes_ = number_of_requests * bytes_per_10_ms_ - 79 (desired_frame_size_bytes_ - cached_bytes_); 80 // If cached_bytes_ is larger than the cache buffer, uninitialized memory 81 // will be read. 82 assert(cached_bytes_ <= bytes_per_10_ms_); 83 assert(-bytes_left == cached_bytes_); 84 cached_buffer_start_ = 0; 85 memcpy(cache_buffer_.get(), cache_ptr, cached_bytes_); 86 } 87 88 } // namespace webrtc 89
