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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 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
     12 
     13 #include <memory.h>
     14 #include <stdio.h>
     15 #include <algorithm>
     16 
     17 #include "webrtc/base/checks.h"
     18 #include "webrtc/base/logging.h"
     19 #include "webrtc/modules/audio_device/audio_device_buffer.h"
     20 
     21 namespace webrtc {
     22 
     23 FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
     24                                  size_t desired_frame_size_bytes,
     25                                  int sample_rate)
     26     : device_buffer_(device_buffer),
     27       desired_frame_size_bytes_(desired_frame_size_bytes),
     28       sample_rate_(sample_rate),
     29       samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
     30       bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
     31       playout_cached_buffer_start_(0),
     32       playout_cached_bytes_(0),
     33       // Allocate extra space on the recording side to reduce the number of
     34       // memmove() calls.
     35       required_record_buffer_size_bytes_(
     36           5 * (desired_frame_size_bytes + bytes_per_10_ms_)),
     37       record_cached_bytes_(0),
     38       record_read_pos_(0),
     39       record_write_pos_(0) {
     40   playout_cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
     41   record_cache_buffer_.reset(new int8_t[required_record_buffer_size_bytes_]);
     42   memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
     43 }
     44 
     45 FineAudioBuffer::~FineAudioBuffer() {}
     46 
     47 size_t FineAudioBuffer::RequiredPlayoutBufferSizeBytes() {
     48   // It is possible that we store the desired frame size - 1 samples. Since new
     49   // audio frames are pulled in chunks of 10ms we will need a buffer that can
     50   // hold desired_frame_size - 1 + 10ms of data. We omit the - 1.
     51   return desired_frame_size_bytes_ + bytes_per_10_ms_;
     52 }
     53 
     54 void FineAudioBuffer::ResetPlayout() {
     55   playout_cached_buffer_start_ = 0;
     56   playout_cached_bytes_ = 0;
     57   memset(playout_cache_buffer_.get(), 0, bytes_per_10_ms_);
     58 }
     59 
     60 void FineAudioBuffer::ResetRecord() {
     61   record_cached_bytes_ = 0;
     62   record_read_pos_ = 0;
     63   record_write_pos_ = 0;
     64   memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
     65 }
     66 
     67 void FineAudioBuffer::GetPlayoutData(int8_t* buffer) {
     68   if (desired_frame_size_bytes_ <= playout_cached_bytes_) {
     69     memcpy(buffer, &playout_cache_buffer_.get()[playout_cached_buffer_start_],
     70            desired_frame_size_bytes_);
     71     playout_cached_buffer_start_ += desired_frame_size_bytes_;
     72     playout_cached_bytes_ -= desired_frame_size_bytes_;
     73     RTC_CHECK_LT(playout_cached_buffer_start_ + playout_cached_bytes_,
     74                  bytes_per_10_ms_);
     75     return;
     76   }
     77   memcpy(buffer, &playout_cache_buffer_.get()[playout_cached_buffer_start_],
     78          playout_cached_bytes_);
     79   // Push another n*10ms of audio to |buffer|. n > 1 if
     80   // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we
     81   // write the audio after the cached bytes copied earlier.
     82   int8_t* unwritten_buffer = &buffer[playout_cached_bytes_];
     83   int bytes_left =
     84       static_cast<int>(desired_frame_size_bytes_ - playout_cached_bytes_);
     85   // Ceiling of integer division: 1 + ((x - 1) / y)
     86   size_t number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
     87   for (size_t i = 0; i < number_of_requests; ++i) {
     88     device_buffer_->RequestPlayoutData(samples_per_10_ms_);
     89     int num_out = device_buffer_->GetPlayoutData(unwritten_buffer);
     90     if (static_cast<size_t>(num_out) != samples_per_10_ms_) {
     91       RTC_CHECK_EQ(num_out, 0);
     92       playout_cached_bytes_ = 0;
     93       return;
     94     }
     95     unwritten_buffer += bytes_per_10_ms_;
     96     RTC_CHECK_GE(bytes_left, 0);
     97     bytes_left -= static_cast<int>(bytes_per_10_ms_);
     98   }
     99   RTC_CHECK_LE(bytes_left, 0);
    100   // Put the samples that were written to |buffer| but are not used in the
    101   // cache.
    102   size_t cache_location = desired_frame_size_bytes_;
    103   int8_t* cache_ptr = &buffer[cache_location];
    104   playout_cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
    105                           (desired_frame_size_bytes_ - playout_cached_bytes_);
    106   // If playout_cached_bytes_ is larger than the cache buffer, uninitialized
    107   // memory will be read.
    108   RTC_CHECK_LE(playout_cached_bytes_, bytes_per_10_ms_);
    109   RTC_CHECK_EQ(static_cast<size_t>(-bytes_left), playout_cached_bytes_);
    110   playout_cached_buffer_start_ = 0;
    111   memcpy(playout_cache_buffer_.get(), cache_ptr, playout_cached_bytes_);
    112 }
    113 
    114 void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer,
    115                                           size_t size_in_bytes,
    116                                           int playout_delay_ms,
    117                                           int record_delay_ms) {
    118   // Check if the temporary buffer can store the incoming buffer. If not,
    119   // move the remaining (old) bytes to the beginning of the temporary buffer
    120   // and start adding new samples after the old samples.
    121   if (record_write_pos_ + size_in_bytes > required_record_buffer_size_bytes_) {
    122     if (record_cached_bytes_ > 0) {
    123       memmove(record_cache_buffer_.get(),
    124               record_cache_buffer_.get() + record_read_pos_,
    125               record_cached_bytes_);
    126     }
    127     record_write_pos_ = record_cached_bytes_;
    128     record_read_pos_ = 0;
    129   }
    130   // Add recorded samples to a temporary buffer.
    131   memcpy(record_cache_buffer_.get() + record_write_pos_, buffer, size_in_bytes);
    132   record_write_pos_ += size_in_bytes;
    133   record_cached_bytes_ += size_in_bytes;
    134   // Consume samples in temporary buffer in chunks of 10ms until there is not
    135   // enough data left. The number of remaining bytes in the cache is given by
    136   // |record_cached_bytes_| after this while loop is done.
    137   while (record_cached_bytes_ >= bytes_per_10_ms_) {
    138     device_buffer_->SetRecordedBuffer(
    139         record_cache_buffer_.get() + record_read_pos_, samples_per_10_ms_);
    140     device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
    141     device_buffer_->DeliverRecordedData();
    142     // Read next chunk of 10ms data.
    143     record_read_pos_ += bytes_per_10_ms_;
    144     // Reduce number of cached bytes with the consumed amount.
    145     record_cached_bytes_ -= bytes_per_10_ms_;
    146   }
    147 }
    148 
    149 }  // namespace webrtc
    150