Revert "Use resampler for playback speeds close to 1.0"

This reverts commit ab98b39d.

Reason for revert: Consistently failing on MSan https://crbug.com/1018617

Original change's description:
> Use resampler for playback speeds close to 1.0
> 
> The WSOLA algorithm introduces noticeable audio artifacts when doing
> small adjustments to playback rate (e.g. 1.03 playback speed). The
> distortions can be described as warbling or transient stuttering.
> 
> This CL introduces fixes this issue by using resampling instead, for
> speeds close to 1.00. The resampled audio doesn't have any distortions,
> but its pitch is shifted proportionally to the playback changes. The
> maximal slowdown/speed up we use resampling for is 5-6%, which
> corresponds to shifting by 1 note tone. Beyond that, the WSOLA
> artifacts become tolerable, and the pitch shifting doesn't.
> 
> Bug: 920019
> Change-Id: Ia1f93dc6a5de8be14e054b9e165fb9508af491f1
> Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1875654
> Commit-Queue: Thomas Guilbert <tguilbert@chromium.org>
> Reviewed-by: Dale Curtis <dalecurtis@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#709679}

TBR=dalecurtis@chromium.org,tguilbert@chromium.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: 920019
Change-Id: I5c7f531c446ebb2af24d26698994a0747dd68aba
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1882255Reviewed-by: Meredith Lane <meredithl@chromium.org>
Commit-Queue: Meredith Lane <meredithl@chromium.org>
Cr-Commit-Position: refs/heads/master@{#709788}

media/filters/audio_renderer_algorithm.cc

@@ -7,13 +7,11 @@
 #include <algorithm>
 #include <cmath>
 
-#include "base/bind.h"
 #include "base/logging.h"
 #include "cc/base/math_util.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_timestamp_helper.h"
 #include "media/base/limits.h"
-#include "media/base/multi_channel_resampler.h"
 #include "media/filters/wsola_internals.h"
 
 namespace media {
@@ -159,73 +157,6 @@ void AudioRendererAlgorithm::SetChannelMask(std::vector<bool> channel_mask) {
     CreateSearchWrappers();
 }
 
-void AudioRendererAlgorithm::OnResamplerRead(int frame_delay,
-                                             AudioBus* audio_bus) {
-  input_frames_read_or_buffered_ +=
-      audio_buffer_.ReadFrames(audio_bus->frames(), 0, audio_bus);
-}
-
-int AudioRendererAlgorithm::CalculateOutputFramesResampled(
-    double playback_rate,
-    int requested_frames) {
-  double input_frames_consumed =
-      input_frames_read_or_buffered_ - resampler_->BufferedFrames();
-
-  int output_frames =
-      static_cast<int>(input_frames_consumed / playback_rate + 0.5);
-
-  // The first or second call to resample appears to consume more input frames
-  // than it actually does. This is due to the internals of |resampler_|
-  // treating the first data read differently, to prime internal buffers. We
-  // therefore appear to have read up to SincResampler::kKernelSize more frames
-  // when using the difference between calls to |resampler_->BufferedFrames()|.
-  // |resampler_| never actually writes more frames than we request out of it,
-  // so we can safely cap this value here.
-  return std::min(output_frames, requested_frames);
-}
-
-int AudioRendererAlgorithm::ResampleAndFill(AudioBus* dest,
-                                            int dest_offset,
-                                            int requested_frames,
-                                            double playback_rate) {
-  if (!resampler_) {
-    resampler_ = std::make_unique<MultiChannelResampler>(
-        channels_, playback_rate, SincResampler::kDefaultRequestSize,
-        base::BindRepeating(&AudioRendererAlgorithm::OnResamplerRead,
-                            base::Unretained(this)));
-  }
-
-  resampler_->SetRatio(playback_rate);
-
-  // Reset with leftover frames from previous resampling iteration.
-  input_frames_read_or_buffered_ = resampler_->BufferedFrames();
-
-  // Directly use |dest| for the most common case of having 0 offset.
-  if (!dest_offset) {
-    resampler_->Resample(requested_frames, dest);
-
-    return CalculateOutputFramesResampled(playback_rate, requested_frames);
-  }
-
-  // This is only really used once, at the beginning of a stream, which means
-  // we can use a temporary variable, rather than saving it as a member.
-  // NOTE: We don't wrap |dest|'s channel data in an AudioBus wrapper, because
-  // |dest_offset| isn't aligned always with AudioBus::kChannelAlignment.
-  std::unique_ptr<AudioBus> resampler_output =
-      AudioBus::Create(channels_, requested_frames);
-
-  resampler_->Resample(requested_frames, resampler_output.get());
-
-  int output_frames =
-      CalculateOutputFramesResampled(playback_rate, requested_frames);
-
-  DCHECK_LE(output_frames, resampler_output->frames());
-
-  resampler_output->CopyPartialFramesTo(0, output_frames, dest_offset, dest);
-
-  return output_frames;
-}
-
 int AudioRendererAlgorithm::FillBuffer(AudioBus* dest,
                                        int dest_offset,
                                        int requested_frames,
@@ -254,18 +185,6 @@ int AudioRendererAlgorithm::FillBuffer(AudioBus* dest,
     return frames_read;
   }
 
-  // WSOLA at playback rates that are close to 1.0 produces noticeable
-  // warbling and stuttering. We prefer resampling the audio at these speeds.
-  // This does results in a noticeable pitch shift.
-  // NOTE: The cutoff values are arbitrary, and picked based off of a tradeoff
-  // between "resample pitch shift" vs "WSOLA distortions".
-  constexpr double kLowerResampleThreshold = 0.95;
-  constexpr double kUpperResampleThreshold = 1.06;
-  if (kLowerResampleThreshold <= playback_rate &&
-      playback_rate <= kUpperResampleThreshold) {
-    return ResampleAndFill(dest, dest_offset, requested_frames, playback_rate);
-  }
-
   // Allocate structures on first non-1.0 playback rate; these can eat a fair
   // chunk of memory. ~56kB for stereo 48kHz, up to ~765kB for 7.1 192kHz.
   if (!ola_window_) {
@@ -316,8 +235,6 @@ void AudioRendererAlgorithm::FlushBuffers() {
     wsola_output_->Zero();
   num_complete_frames_ = 0;
 
-  resampler_.reset();
-
   // Reset |capacity_| so growth triggered by underflows doesn't penalize seek
   // time.
   capacity_ = initial_capacity_;

media/filters/audio_renderer_algorithm.h

@@ -35,7 +35,6 @@
 namespace media {
 
 class AudioBus;
-class MultiChannelResampler;
 
 class MEDIA_EXPORT AudioRendererAlgorithm {
  public:
@@ -146,24 +145,6 @@ class MEDIA_EXPORT AudioRendererAlgorithm {
   // mask has been specified.
   void CreateSearchWrappers();
 
-  // Uses |resampler_| to speed up or slowdown audio, by using a resampling
-  // ratio of |playback_rate|.
-  int ResampleAndFill(AudioBus* dest,
-                      int dest_offset,
-                      int requested_frames,
-                      double playback_rate);
-
-  // Called by |resampler_| to get more audio data.
-  void OnResamplerRead(int frame_delay, AudioBus* audio_bus);
-
-  // Calculate how many frames |resampler_| wrote to output, based off of
-  // |input_frames_read_or_buffered_| and |resampler_->BufferedFrames()|.
-  //
-  // NOTE: The return value is always <= |request_frames|. See comment in the
-  //       implementation file.
-  int CalculateOutputFramesResampled(double playback_rate,
-                                     int requested_frames);
-
   // Parameters.
   AudioRendererAlgorithmParameters audio_renderer_algorithm_params_;
 
@@ -217,14 +198,6 @@ class MEDIA_EXPORT AudioRendererAlgorithm {
   // specifies the index where the next WSOLA window has to overlap-and-add.
   int num_complete_frames_;
 
-  // Used to replace WSOLA algorithm at playback speeds close to 1.0. This is to
-  // prevent noticeable audio artifacts introduced by WSOLA, at the expense of
-  // changing the pitch of the audio.
-  std::unique_ptr<MultiChannelResampler> resampler_;
-
-  // Number of input frames read or buffered by |resampler_|.
-  double input_frames_read_or_buffered_ = 0;
-
   // This stores a part of the output that is created but couldn't be rendered.
   // Output is generated frame-by-frame which at some point might exceed the
   // number of requested samples. Furthermore, due to overlap-and-add,

media/filters/audio_renderer_algorithm_unittest.cc

@@ -189,14 +189,13 @@ class AudioRendererAlgorithmTest : public testing::Test {
     const int kDefaultFramesRequested = kOutputDurationInSec *
         algorithm_.samples_per_second();
 
-    TestPlaybackRate(playback_rate, kDefaultBufferSize, kDefaultFramesRequested,
-                     0);
+    TestPlaybackRate(
+        playback_rate, kDefaultBufferSize, kDefaultFramesRequested);
   }
 
   void TestPlaybackRate(double playback_rate,
                         int buffer_size_in_frames,
-                        int total_frames_requested,
-                        int dest_offset) {
+                        int total_frames_requested) {
     int initial_frames_enqueued = frames_enqueued_;
     int initial_frames_buffered = algorithm_.frames_buffered();
 
@@ -212,10 +211,9 @@ class AudioRendererAlgorithmTest : public testing::Test {
     int frames_remaining = total_frames_requested;
     bool first_fill_buffer = true;
     while (frames_remaining > 0) {
-      int frames_requested =
-          std::min(buffer_size_in_frames - dest_offset, frames_remaining);
-      int frames_written = algorithm_.FillBuffer(
-          bus.get(), dest_offset, frames_requested, playback_rate);
+      int frames_requested = std::min(buffer_size_in_frames, frames_remaining);
+      int frames_written =
+          algorithm_.FillBuffer(bus.get(), 0, frames_requested, playback_rate);
       ASSERT_GT(frames_written, 0) << "Requested: " << frames_requested
                                    << ", playing at " << playback_rate;
 
@@ -358,7 +356,7 @@ TEST_F(AudioRendererAlgorithmTest, InitializeWithLargeParameters) {
 TEST_F(AudioRendererAlgorithmTest, FillBuffer_Bitstream) {
   Initialize(CHANNEL_LAYOUT_STEREO, kSampleFormatEac3, kSamplesPerSecond,
              kSamplesPerSecond / 100);
-  TestPlaybackRate(1.0, kFrameSize, 16 * kFrameSize, /* dest_offset */ 0);
+  TestPlaybackRate(1.0, kFrameSize, 16 * kFrameSize);
 }
 
 TEST_F(AudioRendererAlgorithmTest, FillBuffer_NormalRate) {
@@ -376,36 +374,6 @@ TEST_F(AudioRendererAlgorithmTest, FillBuffer_NearlyNormalSlowerRate) {
   TestPlaybackRate(0.9999);
 }
 
-// This test verifies that the resampling based time stretch algorithms works.
-// The range of playback rates in which we use resampling is [0.95, 1.06].
-TEST_F(AudioRendererAlgorithmTest, FillBuffer_ResamplingRates) {
-  Initialize();
-  TestPlaybackRate(0.94);  // WSOLA.
-  TestPlaybackRate(0.95);  // Lower limit of resampling.
-  TestPlaybackRate(0.97);
-  TestPlaybackRate(1.00);
-  TestPlaybackRate(1.04);
-  TestPlaybackRate(1.06);  // Upper limit of resampling.
-  TestPlaybackRate(1.07);  // WSOLA.
-}
-
-TEST_F(AudioRendererAlgorithmTest, FillBuffer_WithOffset) {
-  Initialize();
-  const int kBufferSize = algorithm_.samples_per_second() / 10;
-  const int kOffset = kBufferSize / 10;
-  const int kFramesRequested =
-      kOutputDurationInSec * algorithm_.samples_per_second();
-
-  // No time-strech.
-  TestPlaybackRate(1.00, kBufferSize, kFramesRequested, kOffset);
-
-  // Resampling based time-strech.
-  TestPlaybackRate(1.05, kBufferSize, kFramesRequested, kOffset);
-
-  // WSOLA based time-strech.
-  TestPlaybackRate(1.25, kBufferSize, kFramesRequested, kOffset);
-}
-
 TEST_F(AudioRendererAlgorithmTest, FillBuffer_OneAndAQuarterRate) {
   Initialize();
   TestPlaybackRate(1.25);
@@ -479,9 +447,9 @@ TEST_F(AudioRendererAlgorithmTest, FillBuffer_SmallBufferSize) {
   Initialize();
   static const int kBufferSizeInFrames = 1;
   static const int kFramesRequested = kOutputDurationInSec * kSamplesPerSecond;
-  TestPlaybackRate(1.0, kBufferSizeInFrames, kFramesRequested, 0);
-  TestPlaybackRate(0.5, kBufferSizeInFrames, kFramesRequested, 0);
-  TestPlaybackRate(1.5, kBufferSizeInFrames, kFramesRequested, 0);
+  TestPlaybackRate(1.0, kBufferSizeInFrames, kFramesRequested);
+  TestPlaybackRate(0.5, kBufferSizeInFrames, kFramesRequested);
+  TestPlaybackRate(1.5, kBufferSizeInFrames, kFramesRequested);
 }
 
 TEST_F(AudioRendererAlgorithmTest, FillBuffer_LargeBufferSize) {