Improved audio buffer handling for iOS

webrtc/modules/audio_device/android/opensles_recorder.cc

 /*
  *  Copyright (c) 2016 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/android/opensles_recorder.h"
 
 #include <android/log.h>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
 
 #define TAG "OpenSLESRecorder"
 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
@@ skipping 303 matching lines @@
   // of samples (and not only multiple of 10ms) to match the native audio unit
   // buffer size.
   ALOGD("frames per native buffer: %" PRIuS,
         audio_parameters_.frames_per_buffer());
   ALOGD("frames per 10ms buffer: %" PRIuS,
         audio_parameters_.frames_per_10ms_buffer());
   ALOGD("bytes per native buffer: %" PRIuS,
         audio_parameters_.GetBytesPerBuffer());
   ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
   RTC_DCHECK(audio_device_buffer_);
-  fine_audio_buffer_.reset(new FineAudioBuffer(
-      audio_device_buffer_, audio_parameters_.GetBytesPerBuffer(),
-      audio_parameters_.sample_rate()));
+  fine_audio_buffer_.reset(
+      new FineAudioBuffer(audio_device_buffer_, audio_parameters_.sample_rate(),
+                          2 * audio_parameters_.GetBytesPerBuffer()));
   // Allocate queue of audio buffers that stores recorded audio samples.
   const int data_size_bytes = audio_parameters_.GetBytesPerBuffer();
   audio_buffers_.reset(new std::unique_ptr<SLint8[]>[kNumOfOpenSLESBuffers]);
   for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
     audio_buffers_[i].reset(new SLint8[data_size_bytes]);
   }
 }
 
 void OpenSLESRecorder::ReadBufferQueue() {
   RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
@@ skipping 13 matching lines @@
   last_rec_time_ = current_time;
   // Send recorded audio data to the WebRTC sink.
   // TODO(henrika): fix delay estimates. It is OK to use fixed values for now
   // since there is no support to turn off built-in EC in combination with
   // OpenSL ES anyhow. Hence, as is, the WebRTC based AEC (which would use
   // these estimates) will never be active.
   const size_t size_in_bytes =
       static_cast<size_t>(audio_parameters_.GetBytesPerBuffer());
   const int8_t* data =
       static_cast<const int8_t*>(audio_buffers_[buffer_index_].get());
-  fine_audio_buffer_->DeliverRecordedData(data, size_in_bytes, 25, 25);
+  fine_audio_buffer_->DeliverRecordedData(
+      rtc::ArrayView<const int8_t>(data, size_in_bytes), 25, 25);
   // Enqueue the utilized audio buffer and use if for recording again.
   EnqueueAudioBuffer();
 }
 
 bool OpenSLESRecorder::EnqueueAudioBuffer() {
   SLresult err =
       (*simple_buffer_queue_)
           ->Enqueue(simple_buffer_queue_, audio_buffers_[buffer_index_].get(),
                     audio_parameters_.GetBytesPerBuffer());
   if (SL_RESULT_SUCCESS != err) {
@@ skipping 32 matching lines @@
   SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
   ALOGD("state.count:%d state.index:%d", state.count, state.index);
 }
 
 SLuint32 OpenSLESRecorder::GetBufferCount() {
   SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
   return state.count;
 }
 
 }  // namespace webrtc