Refactor the AudioDevice for iOS and improve the performance and stability

webrtc/modules/audio_device/fine_audio_buffer.cc

Index: webrtc/modules/audio_device/fine_audio_buffer.cc
diff --git a/webrtc/modules/audio_device/fine_audio_buffer.cc b/webrtc/modules/audio_device/fine_audio_buffer.cc
new file mode 100644
index 0000000000000000000000000000000000000000..374d8ed3b6e2e3b4c672dc5136c47ab868e052c2
--- /dev/null
+++ b/webrtc/modules/audio_device/fine_audio_buffer.cc
@@ -0,0 +1,150 @@
+/*
+ *  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)
+    : 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)),
+      playout_cached_buffer_start_(0),
+      playout_cached_bytes_(0),
+      // Allocate extra space on the recording side to reduce the number of
+      // memmove() calls.
+      required_record_buffer_size_bytes_(
+          5 * (desired_frame_size_bytes + bytes_per_10_ms_)),
+      record_cached_bytes_(0),
+      record_read_pos_(0),
+      record_write_pos_(0) {
+  playout_cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
+  record_cache_buffer_.reset(new int8_t[required_record_buffer_size_bytes_]);
+  memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
+}
+
+FineAudioBuffer::~FineAudioBuffer() {}
+
+size_t FineAudioBuffer::RequiredPlayoutBufferSizeBytes() {
+  // It is possible that we store the desired frame size - 1 samples. Since new
+  // audio frames are pulled in chunks of 10ms we will need a buffer that can
+  // hold desired_frame_size - 1 + 10ms of data. We omit the - 1.
+  return desired_frame_size_bytes_ + bytes_per_10_ms_;
+}
+
+void FineAudioBuffer::ResetPlayout() {
+  playout_cached_buffer_start_ = 0;
+  playout_cached_bytes_ = 0;
+  memset(playout_cache_buffer_.get(), 0, bytes_per_10_ms_);
+}
+
+void FineAudioBuffer::ResetRecord() {
+  record_cached_bytes_ = 0;
+  record_read_pos_ = 0;
+  record_write_pos_ = 0;
+  memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
+}
+
+void FineAudioBuffer::GetPlayoutData(int8_t* buffer) {
+  if (desired_frame_size_bytes_ <= playout_cached_bytes_) {
+    memcpy(buffer, &playout_cache_buffer_.get()[playout_cached_buffer_start_],
+           desired_frame_size_bytes_);
+    playout_cached_buffer_start_ += desired_frame_size_bytes_;
+    playout_cached_bytes_ -= desired_frame_size_bytes_;
+    CHECK_LT(playout_cached_buffer_start_ + playout_cached_bytes_,
+             bytes_per_10_ms_);
+    return;
+  }
+  memcpy(buffer, &playout_cache_buffer_.get()[playout_cached_buffer_start_],
+         playout_cached_bytes_);
+  // Push another n*10ms of audio to |buffer|. n > 1 if
+  // |desired_frame_size_bytes_| is greater than 10ms of audio. Note that we
+  // write the audio after the cached bytes copied earlier.
+  int8_t* unwritten_buffer = &buffer[playout_cached_bytes_];
+  int bytes_left =
+      static_cast<int>(desired_frame_size_bytes_ - playout_cached_bytes_);
+  // Ceiling of integer division: 1 + ((x - 1) / y)
+  size_t number_of_requests = 1 + (bytes_left - 1) / (bytes_per_10_ms_);
+  for (size_t i = 0; i < number_of_requests; ++i) {
+    device_buffer_->RequestPlayoutData(samples_per_10_ms_);
+    int num_out = device_buffer_->GetPlayoutData(unwritten_buffer);
+    if (static_cast<size_t>(num_out) != samples_per_10_ms_) {
+      CHECK_EQ(num_out, 0);
+      playout_cached_bytes_ = 0;
+      return;
+    }
+    unwritten_buffer += bytes_per_10_ms_;
+    CHECK_GE(bytes_left, 0);
+    bytes_left -= static_cast<int>(bytes_per_10_ms_);
+  }
+  CHECK_LE(bytes_left, 0);
+  // Put the samples that were written to |buffer| but are not used in the
+  // cache.
+  size_t cache_location = desired_frame_size_bytes_;
+  int8_t* cache_ptr = &buffer[cache_location];
+  playout_cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
+                          (desired_frame_size_bytes_ - playout_cached_bytes_);
+  // If playout_cached_bytes_ is larger than the cache buffer, uninitialized
+  // memory will be read.
+  CHECK_LE(playout_cached_bytes_, bytes_per_10_ms_);
+  CHECK_EQ(static_cast<size_t>(-bytes_left), playout_cached_bytes_);
+  playout_cached_buffer_start_ = 0;
+  memcpy(playout_cache_buffer_.get(), cache_ptr, playout_cached_bytes_);
+}
+
+void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer,
+                                          size_t size_in_bytes,
+                                          int playout_delay_ms,
+                                          int record_delay_ms) {
+  CHECK_EQ(size_in_bytes, desired_frame_size_bytes_);
+  // Check if the temporary buffer can store the incoming buffer. If not,
+  // move the remaining (old) bytes to the beginning of the temporary buffer
+  // and start adding new samples after the old samples.
+  if (record_write_pos_ + size_in_bytes > required_record_buffer_size_bytes_) {
+    if (record_cached_bytes_ > 0) {
+      memmove(record_cache_buffer_.get(),
+              record_cache_buffer_.get() + record_read_pos_,
+              record_cached_bytes_);
+    }
+    record_write_pos_ = record_cached_bytes_;
+    record_read_pos_ = 0;
+  }
+  // Add recorded samples to a temporary buffer.
+  memcpy(record_cache_buffer_.get() + record_write_pos_, buffer, size_in_bytes);
+  record_write_pos_ += size_in_bytes;
+  record_cached_bytes_ += size_in_bytes;
+  // Consume samples in temporary buffer in chunks of 10ms until there is not
+  // enough data left. The number of remaining bytes in the cache is given by
+  // |record_cached_bytes_| after this while loop is done.
+  while (record_cached_bytes_ >= bytes_per_10_ms_) {
+    device_buffer_->SetRecordedBuffer(
+        record_cache_buffer_.get() + record_read_pos_, samples_per_10_ms_);
+    device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
+    device_buffer_->DeliverRecordedData();
+    // Read next chunk of 10ms data.
+    record_read_pos_ += bytes_per_10_ms_;
+    // Reduce number of cached bytes with the consumed amount.
+    record_cached_bytes_ -= bytes_per_10_ms_;
+  }
+}
+
+}  // namespace webrtc