Improved audio buffer handling for iOS

webrtc/modules/audio_device/fine_audio_buffer.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
 
 #include <memory.h>
 #include <stdio.h>
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 
 namespace webrtc {
 
 FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
-                                 size_t desired_frame_size_bytes,
-                                 int sample_rate)
+                                 int sample_rate,
+                                 size_t capacity)
     : device_buffer_(device_buffer),
-      desired_frame_size_bytes_(desired_frame_size_bytes),
       sample_rate_(sample_rate),
       samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
-      bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)) {
-  LOG(INFO) << "desired_frame_size_bytes:" << desired_frame_size_bytes;
+      bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
+      playout_buffer_(0, capacity) {

[kwiberg-webrtc, 2017/05/29 04:09:02]

You don't specify an initial capacity for record_buffer_. Should you?

[henrika_webrtc, 2017/05/29 10:33:51]

This CL is focusing on playout side since that is where we have experienced
issues lately.
I will take it one more round and check if we can do similar improvements for
recording side.
But I would like to do that in a separate CL.

[kwiberg-webrtc, 2017/05/29 11:07:00]

Acknowledged.

It might make sense to call the argument playout_buffer_capacity or
something---"capacity" is probably too broad.

[henrika_webrtc, 2017/05/29 14:30:05]

Actually. I kept "capacity" and ensured that the recording side is affected in
the same way. If not, construction of an object which only does recording would
be confusing. I also modified the signature for the recording side to
rtc::ArrayView.

+  LOG(INFO) << "samples_per_10_ms_:" << samples_per_10_ms_;
 }
 
 FineAudioBuffer::~FineAudioBuffer() {}
 
 void FineAudioBuffer::ResetPlayout() {
   playout_buffer_.Clear();
 }
 
 void FineAudioBuffer::ResetRecord() {
   record_buffer_.Clear();
 }
 
-void FineAudioBuffer::GetPlayoutData(int8_t* buffer) {
-  const size_t num_bytes = desired_frame_size_bytes_;
+void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int8_t> audio_buffer) {
   // Ask WebRTC for new data in chunks of 10ms until we have enough to
   // fulfill the request. It is possible that the buffer already contains
   // enough samples from the last round.
+  const size_t num_bytes = audio_buffer.size();
   while (playout_buffer_.size() < num_bytes) {
     // Get 10ms decoded audio from WebRTC.
     device_buffer_->RequestPlayoutData(samples_per_10_ms_);
     // Append |bytes_per_10_ms_| elements to the end of the buffer.
     const size_t bytes_written = playout_buffer_.AppendData(
         bytes_per_10_ms_, [&](rtc::ArrayView<int8_t> buf) {
           const size_t samples_per_channel =
               device_buffer_->GetPlayoutData(buf.data());
           // TODO(henrika): this class is only used on mobile devices and is
           // currently limited to mono. Modifications are needed for stereo.
           return sizeof(int16_t) * samples_per_channel;
         });
     RTC_DCHECK_EQ(bytes_per_10_ms_, bytes_written);
   }
   // Provide the requested number of bytes to the consumer.
-  memcpy(buffer, playout_buffer_.data(), num_bytes);
+  memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes);
   // Move remaining samples to start of buffer to prepare for next round.
   memmove(playout_buffer_.data(), playout_buffer_.data() + num_bytes,
           playout_buffer_.size() - num_bytes);
   playout_buffer_.SetSize(playout_buffer_.size() - num_bytes);
 }
 
 void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer,
                                           size_t size_in_bytes,
                                           int playout_delay_ms,
                                           int record_delay_ms) {
   // Always append new data and grow the buffer if needed.
   record_buffer_.AppendData(buffer, size_in_bytes);
   // Consume samples from buffer in chunks of 10ms until there is not
   // enough data left. The number of remaining bytes in the cache is given by
   // the new size of the buffer.
   while (record_buffer_.size() >= bytes_per_10_ms_) {
     device_buffer_->SetRecordedBuffer(record_buffer_.data(),
                                       samples_per_10_ms_);
     device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
     device_buffer_->DeliverRecordedData();
     memmove(record_buffer_.data(), record_buffer_.data() + bytes_per_10_ms_,
             record_buffer_.size() - bytes_per_10_ms_);
     record_buffer_.SetSize(record_buffer_.size() - bytes_per_10_ms_);
   }
 }
 
 }  // namespace webrtc