diff options
author | scherkus@chromium.org <scherkus@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2014-06-04 21:05:50 +0000 |
---|---|---|
committer | scherkus@chromium.org <scherkus@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2014-06-04 21:05:50 +0000 |
commit | 7922a4ba476cd64ba63770070f35ca2381bc3cc7 (patch) | |
tree | e6776e874a3d2f1233827fbd0a8fd689e77c9b39 | |
parent | 3c375f6c833e08602ac5eb8ce38866cbd46190fe (diff) | |
download | chromium_src-7922a4ba476cd64ba63770070f35ca2381bc3cc7.zip chromium_src-7922a4ba476cd64ba63770070f35ca2381bc3cc7.tar.gz chromium_src-7922a4ba476cd64ba63770070f35ca2381bc3cc7.tar.bz2 |
Revert 258215 "Remove muting for extreme playbackRates."
It exposed a bug in the WSOLA algorithm that causes it to incorrectly
trigger underflows.
> Remove muting for extreme playbackRates.
>
> Audio was muted below 0.5x and above 4x as the quality degraded
> significantly under the crossfade algorithm. The quality is now much
> better under the WSLOA algorithm (r220343).
>
> BUG=289354
> R=scherkus@chromium.org
> R=dalecurtis@chromium.org
>
> Review URL: https://codereview.chromium.org/205093002
BUG=368083
TBR=sandersd@chromium.org
Review URL: https://codereview.chromium.org/313213002
git-svn-id: svn://svn.chromium.org/chrome/branches/1985/src@274903 0039d316-1c4b-4281-b951-d872f2087c98
-rw-r--r-- | media/filters/audio_renderer_algorithm.cc | 35 | ||||
-rw-r--r-- | media/filters/audio_renderer_algorithm.h | 11 | ||||
-rw-r--r-- | media/filters/audio_renderer_algorithm_unittest.cc | 2 |
3 files changed, 47 insertions, 1 deletions
diff --git a/media/filters/audio_renderer_algorithm.cc b/media/filters/audio_renderer_algorithm.cc index db65fe9..e73ce65 100644 --- a/media/filters/audio_renderer_algorithm.cc +++ b/media/filters/audio_renderer_algorithm.cc @@ -46,6 +46,12 @@ namespace media { // |search_block_index_| = |search_block_center_offset_| - // |search_block_center_offset_|. +// 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; @@ -70,6 +76,8 @@ AudioRendererAlgorithm::AudioRendererAlgorithm() : channels_(0), samples_per_second_(0), playback_rate_(0), + muted_(false), + muted_partial_frame_(0), capacity_(kStartingBufferSizeInFrames), output_time_(0.0), search_block_center_offset_(0), @@ -143,6 +151,31 @@ int AudioRendererAlgorithm::FillBuffer(AudioBus* dest, int requested_frames) { DCHECK_EQ(channels_, dest->channels()); + // Optimize the |muted_| case to issue a single clear instead of performing + // the full crossfade and clearing each crossfaded frame. + if (muted_) { + int frames_to_render = + std::min(static_cast<int>(audio_buffer_.frames() / playback_rate_), + requested_frames); + + // Compute accurate number of frames to actually skip in the source data. + // Includes the leftover partial frame from last request. However, we can + // only skip over complete frames, so a partial frame may remain for next + // time. + muted_partial_frame_ += frames_to_render * playback_rate_; + int seek_frames = static_cast<int>(muted_partial_frame_); + dest->ZeroFrames(frames_to_render); + audio_buffer_.SeekFrames(seek_frames); + + // Determine the partial frame that remains to be skipped for next call. If + // the user switches back to playing, it may be off time by this partial + // frame, which would be undetectable. If they subsequently switch to + // another playback rate that mutes, the code will attempt to line up the + // frames again. + muted_partial_frame_ -= seek_frames; + return frames_to_render; + } + int slower_step = ceil(ola_window_size_ * playback_rate_); int faster_step = ceil(ola_window_size_ / playback_rate_); @@ -167,6 +200,8 @@ int AudioRendererAlgorithm::FillBuffer(AudioBus* dest, int requested_frames) { void AudioRendererAlgorithm::SetPlaybackRate(float new_rate) { DCHECK_GE(new_rate, 0); playback_rate_ = new_rate; + muted_ = + playback_rate_ < kMinPlaybackRate || playback_rate_ > kMaxPlaybackRate; } void AudioRendererAlgorithm::FlushBuffers() { diff --git a/media/filters/audio_renderer_algorithm.h b/media/filters/audio_renderer_algorithm.h index f251ff72..39e4db6 100644 --- a/media/filters/audio_renderer_algorithm.h +++ b/media/filters/audio_renderer_algorithm.h @@ -19,6 +19,8 @@ // are preserved. See audio_renderer_algorith.cc for a more elaborate // description of the algorithm. // +// Audio at very low or very high playback rates are muted to preserve quality. +// #ifndef MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_ #define MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_ @@ -82,6 +84,9 @@ class MEDIA_EXPORT AudioRendererAlgorithm { // Returns the samples per second for this audio stream. int samples_per_second() { return samples_per_second_; } + // Is the sound currently muted? + bool is_muted() { return muted_; } + private: // Within |search_block_|, find the block of data that is most similar to // |target_block_|, and write it in |optimal_block_|. This method assumes that @@ -135,6 +140,12 @@ class MEDIA_EXPORT AudioRendererAlgorithm { // Buffered audio data. AudioBufferQueue audio_buffer_; + // True if the audio should be muted. + bool muted_; + + // If muted, keep track of partial frames that should have been skipped over. + double muted_partial_frame_; + // How many frames to have in the queue before we report the queue is full. int capacity_; diff --git a/media/filters/audio_renderer_algorithm_unittest.cc b/media/filters/audio_renderer_algorithm_unittest.cc index 596c8cc..0f63922 100644 --- a/media/filters/audio_renderer_algorithm_unittest.cc +++ b/media/filters/audio_renderer_algorithm_unittest.cc @@ -151,7 +151,7 @@ class AudioRendererAlgorithmTest : public testing::Test { bool all_zero = true; for (int i = 0; i < frames_written && all_zero; ++i) all_zero = audio_data->channel(ch)[i] == 0.0f; - ASSERT_FALSE(all_zero) << " for channel " << ch; + ASSERT_EQ(algorithm_.is_muted(), all_zero) << " for channel " << ch; } } |