Upgrade AudioRendererAlgorithm to use WSOLA,

media/filters/audio_renderer_algorithm.cc

Index: media/filters/audio_renderer_algorithm.cc
diff --git a/media/filters/audio_renderer_algorithm.cc b/media/filters/audio_renderer_algorithm.cc
index 97f0811384159ff35868bb45f10eabdc7a94d86b..52c39cb8d0196e34500aa8e269962a5466dce408 100644
--- a/media/filters/audio_renderer_algorithm.cc
+++ b/media/filters/audio_renderer_algorithm.cc
@@ -12,41 +12,47 @@
 #include "media/audio/audio_util.h"
 #include "media/base/audio_buffer.h"
 #include "media/base/audio_bus.h"
+#include "media/filters/wsola_internals.h"
 
 namespace media {
 
-// The starting size in frames for |audio_buffer_|. Previous usage maintained a
-// queue of 16 AudioBuffers, each of 512 frames. This worked well, so we
-// maintain this number of frames.
-static const int kStartingBufferSizeInFrames = 16 * 512;
-
 // The maximum size in frames for the |audio_buffer_|. Arbitrarily determined.
 // This number represents 3 seconds of 96kHz/16 bit 7.1 surround sound.
 static const int kMaxBufferSizeInFrames = 3 * 96000;
 
-// Duration of audio segments used for crossfading (in seconds).
-static const double kWindowDuration = 0.08;
-
-// Duration of crossfade between audio segments (in seconds).
-static const double kCrossfadeDuration = 0.008;
-
 // Max/min supported playback rates for fast/slow audio. Audio outside of these
 // ranges are muted.
 // Audio at these speeds would sound better under a frequency domain algorithm.
 static const float kMinPlaybackRate = 0.5f;
 static const float kMaxPlaybackRate = 4.0f;
 
+// Overlap-and-add window size in milliseconds.
+static const int kOlaWindowSizeMs = 20;
+
+// Size of search interval in milliseconds. The search interval is
+// [-delta delta] around |output_index_| * |playback_rate_|. So the search
+// interval is 2 * delta.
+static const int kWsolaSearchIntervalMs = 30;
+
+// The starting size in frames for |audio_buffer_|. Previous usage maintained a
+// queue of 16 AudioBuffers, each of 512 frames. This worked well, so we
+// maintain this number of frames.
+static const int kStartingBufferSizeInFrames = 16 * 512;
+
 AudioRendererAlgorithm::AudioRendererAlgorithm()
     : channels_(0),
       samples_per_second_(0),
       playback_rate_(0),
-      frames_in_crossfade_(0),
-      index_into_window_(0),
-      crossfade_frame_number_(0),
       muted_(false),
       muted_partial_frame_(0),
-      window_size_(0),
-      capacity_(kStartingBufferSizeInFrames) {
+      capacity_(kStartingBufferSizeInFrames),
+      output_index_(0),
+      search_block_center_offset_(0),
+      num_candidate_frames_(0),
+      target_block_index_(0),
+      ola_window_size_(0),
+      ola_hop_size_(0),
+      num_complete_frames_(0) {
 }
 
 AudioRendererAlgorithm::~AudioRendererAlgorithm() {}
@@ -59,9 +65,31 @@ void AudioRendererAlgorithm::Initialize(float initial_playback_rate,
   samples_per_second_ = params.sample_rate();
   SetPlaybackRate(initial_playback_rate);
 
-  window_size_ = samples_per_second_ * kWindowDuration;
-  frames_in_crossfade_ = samples_per_second_ * kCrossfadeDuration;
-  crossfade_buffer_ = AudioBus::Create(channels_, frames_in_crossfade_);
+  num_candidate_frames_ =
+      (kWsolaSearchIntervalMs * samples_per_second_) / 1000 + 1;
+
+  // Make sure window size in an even number.
+  ola_window_size_ = (kOlaWindowSizeMs * samples_per_second_ / 1000 / 2) * 2;
+
+  ola_hop_size_ = ola_window_size_ / 2;
+
+  search_block_center_offset_ = (num_candidate_frames_ - 1) / 2 + (
+      ola_window_size_ / 2 - 1);

[marpan, 2013/08/02 17:56:54]

The second term is because of offset to first frame?

[turaj, 2013/08/02 23:45:59]

If we have L+1 candidates, it means we have to check L/2 candidate blocks to the
left of the search-block-center. The first sample of the left most candidate
block is L/2 + (|ola_window_size|/2 - 1) to the left of the search-block-center.

In another word, we are computing cross correlation of target-block (of size
|ola_window_size|) with search-block for lags -L/2 to L/2 around the
search-block-center. Therefore the left most sample needed for this computation
is -(L/2 + (|ola_window_size|/2 - 1)). 

On 2013/08/02 17:56:54, marpan wrote:

+
+  ola_window_.reset(new float[ola_window_size_]);
+  internal::GetSymmetricHanningWindow(ola_window_size_, ola_window_.get());
+
+  transition_window_.reset(new float[ola_window_size_ * 2]);
+  internal::GetSymmetricHanningWindow(2 * ola_window_size_,
+                                      transition_window_.get());
+
+  wsola_output_ = AudioBus::Create(channels_, ola_window_size_ + ola_hop_size_);
+
+  // Auxiliary containers.
+  optimal_block_ = AudioBus::Create(channels_, ola_window_size_);
+  search_block_ = AudioBus::Create(
+      channels_, num_candidate_frames_ + (ola_window_size_ - 1));
+  target_block_ = AudioBus::Create(channels_, ola_window_size_);
 }
 
 int AudioRendererAlgorithm::FillBuffer(AudioBus* dest, int requested_frames) {
@@ -93,12 +121,12 @@ int AudioRendererAlgorithm::FillBuffer(AudioBus* dest, int requested_frames) {
     return frames_to_render;
   }
 
-  int slower_step = ceil(window_size_ * playback_rate_);
-  int faster_step = ceil(window_size_ / playback_rate_);
+  int slower_step = ceil(ola_window_size_ * playback_rate_);
+  int faster_step = ceil(ola_window_size_ / playback_rate_);
 
   // Optimize the most common |playback_rate_| ~= 1 case to use a single copy
   // instead of copying frame by frame.
-  if (window_size_ <= faster_step && slower_step >= window_size_) {
+  if (ola_window_size_ <= faster_step && slower_step >= ola_window_size_) {
     const int frames_to_copy =
         std::min(audio_buffer_.frames(), requested_frames);
     const int frames_read = audio_buffer_.ReadFrames(frames_to_copy, 0, dest);
@@ -106,277 +134,228 @@ int AudioRendererAlgorithm::FillBuffer(AudioBus* dest, int requested_frames) {
     return frames_read;
   }
 
-  int total_frames_rendered = 0;
-  while (total_frames_rendered < requested_frames) {
-    if (index_into_window_ >= window_size_)
-      ResetWindow();
-
-    int rendered_frames = 0;
-    if (window_size_ > faster_step) {
-      rendered_frames =
-          OutputFasterPlayback(dest,
-                               total_frames_rendered,
-                               requested_frames - total_frames_rendered,
-                               window_size_,
-                               faster_step);
-    } else if (slower_step < window_size_) {
-      rendered_frames =
-          OutputSlowerPlayback(dest,
-                               total_frames_rendered,
-                               requested_frames - total_frames_rendered,
-                               slower_step,
-                               window_size_);
-    } else {
-      NOTREACHED();
-    }
+  int total_frames_rendered = WsolaOutput(requested_frames, dest);
+  return total_frames_rendered;
+}
 
-    if (rendered_frames == 0)
-      break;
+void AudioRendererAlgorithm::SetPlaybackRate(float new_rate) {
+  DCHECK_GE(new_rate, 0);
+  // Round it to two decimal digits.
+  playback_rate_ = floor(new_rate * 100.f + 0.5f) / 100;
+  muted_ =
+      playback_rate_ < kMinPlaybackRate || playback_rate_ > kMaxPlaybackRate;
+}
 
-    total_frames_rendered += rendered_frames;
-  }
-  return total_frames_rendered;
+void AudioRendererAlgorithm::FlushBuffers() {
+  // Clear the queue of decoded packets (releasing the buffers).
+  audio_buffer_.Clear();
+  output_index_ = 0;
+  target_block_index_ = 0;
+  wsola_output_->Zero();
+  num_complete_frames_ = 0;
 }
 
-void AudioRendererAlgorithm::ResetWindow() {
-  DCHECK_LE(index_into_window_, window_size_);
-  index_into_window_ = 0;
-  crossfade_frame_number_ = 0;
+base::TimeDelta AudioRendererAlgorithm::GetTime() {
+  return audio_buffer_.current_time();
 }
 
-int AudioRendererAlgorithm::OutputFasterPlayback(AudioBus* dest,
-                                                 int dest_offset,
-                                                 int requested_frames,
-                                                 int input_step,
-                                                 int output_step) {
-  // Ensure we don't run into OOB read/write situation.
-  CHECK_GT(input_step, output_step);
-  DCHECK_LT(index_into_window_, window_size_);
-  DCHECK_GT(playback_rate_, 1.0);
-  DCHECK(!muted_);
-
-  if (audio_buffer_.frames() < 1)
-    return 0;
+void AudioRendererAlgorithm::EnqueueBuffer(
+    const scoped_refptr<AudioBuffer>& buffer_in) {
+  DCHECK(!buffer_in->end_of_stream());
+  audio_buffer_.Append(buffer_in);
+}
 
-  // The audio data is output in a series of windows. For sped-up playback,
-  // the window is comprised of the following phases:
-  //
-  //  a) Output raw data.
-  //  b) Save bytes for crossfade in |crossfade_buffer_|.
-  //  c) Drop data.
-  //  d) Output crossfaded audio leading up to the next window.
-  //
-  // The duration of each phase is computed below based on the |window_size_|
-  // and |playback_rate_|.
-  DCHECK_LE(frames_in_crossfade_, output_step);
-
-  // This is the index of the end of phase a, beginning of phase b.
-  int outtro_crossfade_begin = output_step - frames_in_crossfade_;
-
-  // This is the index of the end of phase b, beginning of phase c.
-  int outtro_crossfade_end = output_step;
-
-  // This is the index of the end of phase c, beginning of phase d.
-  // This phase continues until |index_into_window_| reaches |window_size_|, at
-  // which point the window restarts.
-  int intro_crossfade_begin = input_step - frames_in_crossfade_;
-
-  // a) Output raw frames if we haven't reached the crossfade section.
-  if (index_into_window_ < outtro_crossfade_begin) {
-    // Read as many frames as we can and return the count. If it's not enough,
-    // we will get called again.
-    const int frames_to_copy =
-        std::min(requested_frames, outtro_crossfade_begin - index_into_window_);
-    int copied = audio_buffer_.ReadFrames(frames_to_copy, dest_offset, dest);
-    index_into_window_ += copied;
-    return copied;
-  }
+bool AudioRendererAlgorithm::IsQueueFull() {
+  return audio_buffer_.frames() >= capacity_;
+}
+
+void AudioRendererAlgorithm::IncreaseQueueCapacity() {
+  capacity_ = std::min(2 * capacity_, kMaxBufferSizeInFrames);
+}
 
-  // b) Save outtro crossfade frames into intermediate buffer, but do not output
-  //    anything to |dest|.
-  if (index_into_window_ < outtro_crossfade_end) {
-    // This phase only applies if there are bytes to crossfade.
-    DCHECK_GT(frames_in_crossfade_, 0);
-    int crossfade_start = index_into_window_ - outtro_crossfade_begin;
-    int crossfade_count = outtro_crossfade_end - index_into_window_;
-    int copied = audio_buffer_.ReadFrames(
-        crossfade_count, crossfade_start, crossfade_buffer_.get());
-    index_into_window_ += copied;
-
-    // Did we get all the frames we need? If not, return and let subsequent
-    // calls try to get the rest.
-    if (copied != crossfade_count)
-      return 0;
+bool AudioRendererAlgorithm::CanPerformWsola() const {
+  const int search_block_size = num_candidate_frames_ + (ola_window_size_ - 1);
+  const int frames = audio_buffer_.frames();
+  if (target_block_index_ + ola_window_size_ <= frames &&
+      GetSearchRegionIndex() + search_block_size <= frames) {
+    return true;
   }
+  return false;
+}
 
-  // c) Drop frames until we reach the intro crossfade section.
-  if (index_into_window_ < intro_crossfade_begin) {
-    // Check if there is enough data to skip all the frames needed. If not,
-    // return 0 and let subsequent calls try to skip it all.
-    int seek_frames = intro_crossfade_begin - index_into_window_;
-    if (audio_buffer_.frames() < seek_frames)
-      return 0;
-    audio_buffer_.SeekFrames(seek_frames);
+int AudioRendererAlgorithm::WsolaOutput(int requested_frames, AudioBus* dest) {
+  DCHECK_EQ(channels_, dest->channels());
 
-    // We've dropped all the frames that need to be dropped.
-    index_into_window_ += seek_frames;
+  // First read the frames which are ready.
+  int rendered_frames = ReadWsolaOutput(requested_frames, 0, dest);
+  while (rendered_frames < requested_frames && CanPerformWsola()) {
+    rendered_frames += Wsola(requested_frames - rendered_frames,
+                             rendered_frames, dest);
   }
-
-  // d) Crossfade and output a frame, as long as we have data.
-  if (audio_buffer_.frames() < 1)
-    return 0;
-  DCHECK_GT(frames_in_crossfade_, 0);
-  DCHECK_LT(index_into_window_, window_size_);
-
-  int offset_into_buffer = index_into_window_ - intro_crossfade_begin;
-  int copied = audio_buffer_.ReadFrames(1, dest_offset, dest);
-  DCHECK_EQ(copied, 1);
-  CrossfadeFrame(crossfade_buffer_.get(),
-                 offset_into_buffer,
-                 dest,
-                 dest_offset,
-                 offset_into_buffer);
-  index_into_window_ += copied;
-  return copied;
+  return rendered_frames;
 }
 
-int AudioRendererAlgorithm::OutputSlowerPlayback(AudioBus* dest,
-                                                 int dest_offset,
-                                                 int requested_frames,
-                                                 int input_step,
-                                                 int output_step) {
-  // Ensure we don't run into OOB read/write situation.
-  CHECK_LT(input_step, output_step);
-  DCHECK_LT(index_into_window_, window_size_);
-  DCHECK_LT(playback_rate_, 1.0);
-  DCHECK_NE(playback_rate_, 0);
-  DCHECK(!muted_);
-
-  if (audio_buffer_.frames() < 1)
-    return 0;
+int AudioRendererAlgorithm::Wsola(
+    int requested_frames, int dest_offset, AudioBus* dest) {
+  if (!GetOptimalBlock())
+    return 0;  // We cannot continue as |optimal_block| is not found.
+               // There was not enough data.
+
+  // Overlap-and-add.
+  for (int k = 0; k < channels_; ++k) {
+    float* ch_opt_frame = optimal_block_->channel(k);
+    float* ch_output = wsola_output_->channel(k) + num_complete_frames_;
+    for (int n = 0; n < ola_hop_size_; ++n) {
+      ch_output[n] = ch_output[n] * ola_window_[ola_hop_size_ + n] +
+          ch_opt_frame[n] * ola_window_[n];
+    }
 
-  // The audio data is output in a series of windows. For slowed down playback,
-  // the window is comprised of the following phases:
-  //
-  //  a) Output raw data.
-  //  b) Output and save bytes for crossfade in |crossfade_buffer_|.
-  //  c) Output* raw data.
-  //  d) Output* crossfaded audio leading up to the next window.
-  //
-  // * Phases c) and d) do not progress |audio_buffer_|'s cursor so that the
-  // |audio_buffer_|'s cursor is in the correct place for the next window.
-  //
-  // The duration of each phase is computed below based on the |window_size_|
-  // and |playback_rate_|.
-  DCHECK_LE(frames_in_crossfade_, input_step);
-
-  // This is the index of the end of phase a, beginning of phase b.
-  int intro_crossfade_begin = input_step - frames_in_crossfade_;
-
-  // This is the index of the end of phase b, beginning of phase c.
-  int intro_crossfade_end = input_step;
-
-  // This is the index of the end of phase c,  beginning of phase d.
-  // This phase continues until |index_into_window_| reaches |window_size_|, at
-  // which point the window restarts.
-  int outtro_crossfade_begin = output_step - frames_in_crossfade_;
-
-  // a) Output raw frames.
-  if (index_into_window_ < intro_crossfade_begin) {
-    // Read as many frames as we can and return the count. If it's not enough,
-    // we will get called again.
-    const int frames_to_copy =
-        std::min(requested_frames, intro_crossfade_begin - index_into_window_);
-    int copied = audio_buffer_.ReadFrames(frames_to_copy, dest_offset, dest);
-    index_into_window_ += copied;
-    return copied;
+    // Copy the second half to the output.
+    memcpy(&ch_output[ola_hop_size_], &ch_opt_frame[ola_hop_size_],
+           sizeof(*ch_opt_frame) * ola_hop_size_);
   }
 
-  // b) Save the raw frames for the intro crossfade section, then copy the
-  //    same frames to |dest|.
-  if (index_into_window_ < intro_crossfade_end) {
-    const int frames_to_copy =
-        std::min(requested_frames, intro_crossfade_end - index_into_window_);
-    int offset = index_into_window_ - intro_crossfade_begin;
-    int copied = audio_buffer_.ReadFrames(
-        frames_to_copy, offset, crossfade_buffer_.get());
-    crossfade_buffer_->CopyPartialFramesTo(offset, copied, dest_offset, dest);
-    index_into_window_ += copied;
-    return copied;
-  }
+  num_complete_frames_ += ola_hop_size_;
+  output_index_ += ola_hop_size_;
 
-  // c) Output a raw frame into |dest| without advancing the |audio_buffer_|
-  //    cursor.
-  int audio_buffer_offset = index_into_window_ - intro_crossfade_end;
-  DCHECK_GE(audio_buffer_offset, 0);
-  if (audio_buffer_.frames() <= audio_buffer_offset)
-    return 0;
-  int copied =
-      audio_buffer_.PeekFrames(1, audio_buffer_offset, dest_offset, dest);
-  DCHECK_EQ(1, copied);
-
-  // d) Crossfade the next frame of |crossfade_buffer_| into |dest| if we've
-  //    reached the outtro crossfade section of the window.
-  if (index_into_window_ >= outtro_crossfade_begin) {
-    int offset_into_crossfade_buffer =
-        index_into_window_ - outtro_crossfade_begin;
-    CrossfadeFrame(dest,
-                   dest_offset,
-                   crossfade_buffer_.get(),
-                   offset_into_crossfade_buffer,
-                   offset_into_crossfade_buffer);
-  }
+  RemoveOldInputFrames();
+  return ReadWsolaOutput(requested_frames, dest_offset, dest);
+}
 
-  index_into_window_ += copied;
-  return copied;
+int AudioRendererAlgorithm::GetSearchRegionIndex() const {
+  // Center of the search region, in frames.
+  const int search_block_center_index = static_cast<int>(floor(
+      output_index_ * playback_rate_ + 0.5));
+
+  // Index of the beginning of the search region, in frames.
+  return search_block_center_index - search_block_center_offset_;
 }
 
-void AudioRendererAlgorithm::CrossfadeFrame(AudioBus* intro,
-                                            int intro_offset,
-                                            AudioBus* outtro,
-                                            int outtro_offset,
-                                            int fade_offset) {
-  float crossfade_ratio =
-      static_cast<float>(fade_offset) / frames_in_crossfade_;
-  for (int channel = 0; channel < channels_; ++channel) {
-    outtro->channel(channel)[outtro_offset] =
-        (1.0f - crossfade_ratio) * intro->channel(channel)[intro_offset] +
-        (crossfade_ratio) * outtro->channel(channel)[outtro_offset];
-  }
+void AudioRendererAlgorithm::RemoveOldInputFrames() {
+  const int earliest_used_index = std::min(target_block_index_,
+                                           GetSearchRegionIndex());
+
+  if (earliest_used_index < 0)
+    return;  // Nothing to remove
+
+  // Assuming |playback_rate_| * 100 == floor(|playback_rate_| * 100)
+  // that is |playback_rate_| is represented by 2 decimal digits, only.
+  // We eliminate blocks of size 100 * |playback_rate_| from input.
+  const int kOutputFramesPerBlock = 100;
+  const int input_frames_per_block =
+      static_cast<int>(floor(playback_rate_ * kOutputFramesPerBlock + 0.5f));
+  const int blocks_to_remove = earliest_used_index / input_frames_per_block;
+  const int input_frames_to_remove = input_frames_per_block * blocks_to_remove;
+
+  // Remove frames from input and adjust indices accordingly.
+  audio_buffer_.SeekFrames(input_frames_to_remove);
+  target_block_index_ -= input_frames_to_remove;
+
+  // Adjust output index.
+  output_index_ -= kOutputFramesPerBlock * blocks_to_remove;
+  DCHECK_GE(output_index_, 0);
 }
 
-void AudioRendererAlgorithm::SetPlaybackRate(float new_rate) {
-  DCHECK_GE(new_rate, 0);
-  playback_rate_ = new_rate;
-  muted_ =
-      playback_rate_ < kMinPlaybackRate || playback_rate_ > kMaxPlaybackRate;
+int AudioRendererAlgorithm::ReadWsolaOutput(
+    int requested_frames, int dest_offset, AudioBus* dest) {
+  int rendered_frames = std::min(num_complete_frames_, requested_frames);
 
-  ResetWindow();
-}
+  if (rendered_frames == 0)
+    return 0;  // There is nothing to read from |wsola_output_|, return.
 
-void AudioRendererAlgorithm::FlushBuffers() {
-  ResetWindow();
+  wsola_output_->CopyPartialFramesTo(0, rendered_frames, dest_offset, dest);
 
-  // Clear the queue of decoded packets (releasing the buffers).
-  audio_buffer_.Clear();
+  // Remove the frames which are read.
+  int frames_to_move = wsola_output_->frames() - rendered_frames;
+  for (int k = 0; k < channels_; ++k) {
+    float* ch = wsola_output_->channel(k);
+    memmove(ch, &ch[rendered_frames], sizeof(*ch) * frames_to_move);
+  }
+  num_complete_frames_ -= rendered_frames;
+  return rendered_frames;
 }
 
-base::TimeDelta AudioRendererAlgorithm::GetTime() {
-  return audio_buffer_.current_time();
-}
+bool AudioRendererAlgorithm::TargetIsWithinSearchRegion() const {
+  const int search_block_index = GetSearchRegionIndex();
+  const int search_block_size = num_candidate_frames_ + (ola_window_size_ - 1);
 
-void AudioRendererAlgorithm::EnqueueBuffer(
-    const scoped_refptr<AudioBuffer>& buffer_in) {
-  DCHECK(!buffer_in->end_of_stream());
-  audio_buffer_.Append(buffer_in);
+  if (target_block_index_ >= search_block_index &&
+      target_block_index_ + ola_window_size_ <=
+      search_block_index + search_block_size) {
+    return true;
+  }
+  return false;
 }
 
-bool AudioRendererAlgorithm::IsQueueFull() {
-  return audio_buffer_.frames() >= capacity_;
+bool AudioRendererAlgorithm::GetOptimalBlock() {
+  int optimal_index = 0;
+  if (TargetIsWithinSearchRegion()) {
+    optimal_index = target_block_index_;
+    // Get the optimal window.
+    if (!PeekAudioWithZeroAppend(optimal_index, optimal_block_.get()))
+      return false;
+  } else {
+    if (!PeekAudioWithZeroAppend(target_block_index_, target_block_.get()))
+      return false;
+    const int search_region_index = GetSearchRegionIndex();

[marpan, 2013/08/02 17:56:54]

You use "search_block_index" for this above, better to use same name.

[turaj, 2013/08/02 23:45:59]

Absolutely.

On 2013/08/02 17:56:54, marpan wrote:

+
+    if (!PeekAudioWithZeroAppend(search_region_index, search_block_.get()))
+      return false;
+
+    int last_optimal = target_block_index_ - ola_hop_size_ -
+        search_region_index;
+    internal::Interval exclude_iterval = std::make_pair(last_optimal - 80,
+                                                        last_optimal + 80);
+    // |optimal_index| is in frames and it is relative to the beginning
+    // of the |search_block_|.
+    optimal_index = internal::OptimalIndex(
+        search_block_.get(), target_block_.get(), exclude_iterval);
+
+    // Translate |index| w.r.t. the beginning of |audio_buffer_|.
+    optimal_index += search_region_index;
+
+    // Get the optimal window.
+    PeekAudioWithZeroAppend(optimal_index, optimal_block_.get());
+
+    // Make a transition from target window to the optimal window if different.

[marpan, 2013/08/02 17:56:54]

you mean...from target block to the optimal block..?

[turaj, 2013/08/02 23:45:59]

You right, my mistake.

On 2013/08/02 17:56:54, marpan wrote:

+    // Target window has the best continuation to the current current output.

[marpan, 2013/08/02 17:56:54]

Target block instead of "window"

[marpan, 2013/08/02 17:56:54]

Remove one of the  "current"

[turaj, 2013/08/02 23:45:59]

Done.

[turaj, 2013/08/02 23:45:59]

Done.

+    // Optimal block is the most similar block to the target, however, it might
+    // introduce some discontinuity when over-lap-added. Therefore, we combine
+    // them for a smoother transition.
+    for (int k = 0; k < channels_; ++k) {
+      float* ch_opt = optimal_block_->channel(k);
+      float* ch_target = target_block_->channel(k);
+      for (int n = 0; n < ola_window_size_; ++n) {
+        ch_opt[n] = ch_opt[n] * transition_window_[n] + ch_target[n] *

[marpan, 2013/08/02 17:56:54]

May want to comment about transition_window. Is it of size 2L to get varying
weight such that weighting term favors target near n=0 and optimal near L?

[turaj, 2013/08/02 23:45:59]

Done.

+            transition_window_[ola_window_size_ + n];

[marpan, 2013/08/02 17:56:54]

No change needed for this comment. Just wondering if some measure can be used in
OptimalIndex() selection that incorporates both similarity and
discontinuity/smoothing constraint, so to avoid this extra smoothing step (with
preset weighting term) after OptimalIndex().

[turaj, 2013/08/02 23:45:59]

I guess one can do something along the lines you suggest. It you accept my
notation the optimal_block at the end of this function is

optimal_block = RampOutWindow * target_block + RampInWindow * most_similar;

So this suggest that instead of maximizing DotProduct(target_block,
candidate_block) to find most_similar, one might maximize
DotProduct(target_block, RampInWindow * candidate_block). To save complexity one
can evaluate DotProduct(target_block * candidate_block, RampInWindow).

I can test this, if you think it makes sense.




On 2013/08/02 17:56:54, marpan wrote:

[marpan, 2013/08/06 17:14:10]

No need to make any change for this comment.

+      }
+    }
+  }
+
+  // Next target is one hop ahead of the current optimal.
+  target_block_index_ = optimal_index + ola_hop_size_;
+  return true;
 }
 
-void AudioRendererAlgorithm::IncreaseQueueCapacity() {
-  capacity_ = std::min(2 * capacity_, kMaxBufferSizeInFrames);
+bool AudioRendererAlgorithm::PeekAudioWithZeroAppend(
+    int read_offset_frames, AudioBus* dest) {
+  int num_frames = dest->frames();
+  if (read_offset_frames + num_frames > audio_buffer_.frames())
+    return false;
+
+  int write_offset = 0;
+  int num_frames_to_read = dest->frames();
+  if (read_offset_frames < 0) {
+    int num_zero_frames_appended = std::min(-read_offset_frames,
+                                            num_frames_to_read);
+    read_offset_frames = 0;
+    num_frames_to_read -= num_zero_frames_appended;
+    write_offset = num_zero_frames_appended;
+    dest->ZeroFrames(num_zero_frames_appended);
+  }
+  audio_buffer_.PeekFrames(num_frames_to_read, read_offset_frames,
+                           write_offset, dest);
+  return true;
 }
 
 }  // namespace media