[Chromecast] Fix rendering delay tests

Fix some issues in MixerInputConnection which caused incorrect rendering
delay or failure to start the stream.

Bug: internal b/143135800
Test: cast_media_unittests
Change-Id: I2c1a39da00aa36d36f72741cb374a1b87e5b46bb
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1918180Reviewed-by: Yuchen Liu <yucliu@chromium.org>
Commit-Queue: Kenneth MacKay <kmackay@chromium.org>
Cr-Commit-Position: refs/heads/master@{#715517}

chromecast/media/cma/backend/audio_decoder_for_mixer.cc

@@ -46,13 +46,10 @@ namespace media {
 
 namespace {
 
-// Make fill size big enough that most buffers will fit into it. This is more
-// efficient in general since we don't need to allocate more buffers from the
-// heap (can use the buffer pool instead).
-const int kFillSizeFrames = 1536;
+const int kInitialFillSizeFrames = 512;
 const double kPlaybackRateEpsilon = 0.001;
 
-const int64_t kDefaultInputQueueMs = 90;
+const int64_t kDefaultInputQueueMs = 200;
 constexpr base::TimeDelta kFadeTime = base::TimeDelta::FromMilliseconds(5);
 const int kDefaultStartThresholdMs = 70;
 
@@ -108,6 +105,7 @@ AudioDecoderForMixer::AudioDecoderForMixer(
     MediaPipelineBackendForMixer* backend)
     : backend_(backend),
       task_runner_(backend->GetTaskRunner()),
+      buffer_pool_frames_(kInitialFillSizeFrames),
       pending_output_frames_(kNoPendingOutput),
       pool_(new ::media::AudioBufferMemoryPool()),
       weak_factory_(this) {
@@ -161,13 +159,21 @@ bool AudioDecoderForMixer::Start(int64_t playback_start_pts,
   return true;
 }
 
-void AudioDecoderForMixer::CreateMixerInput(const AudioConfig& config,
-                                            bool start_playback_asap) {
+void AudioDecoderForMixer::CreateBufferPool(const AudioConfig& config,
+                                            int frame_count) {
+  DCHECK_GT(frame_count, 0);
+  buffer_pool_frames_ = frame_count;
   buffer_pool_ = base::MakeRefCounted<IOBufferPool>(
-      kFillSizeFrames * sizeof(float) * config.channel_number +
+      frame_count * sizeof(float) * config.channel_number +
           kAudioMessageHeaderSize,
       std::numeric_limits<size_t>::max(), true /* threadsafe */);
   buffer_pool_->Preallocate(1);
+}
+
+void AudioDecoderForMixer::CreateMixerInput(const AudioConfig& config,
+                                            bool start_playback_asap) {
+  CreateBufferPool(config, buffer_pool_frames_);
+  DCHECK_GT(buffer_pool_frames_, 0);
 
   audio_resampler_ = std::make_unique<AudioResampler>(config.channel_number);
   audio_resampler_->SetMediaClockRate(av_sync_clock_rate_);
@@ -184,7 +190,7 @@ void AudioDecoderForMixer::CreateMixerInput(const AudioConfig& config,
   params.set_sample_rate(config.samples_per_second);
   params.set_num_channels(config.channel_number);
   params.set_channel_selection(ToPlayoutChannel(backend_->AudioChannel()));
-  params.set_fill_size_frames(kFillSizeFrames);
+  params.set_fill_size_frames(buffer_pool_frames_);
   params.set_start_threshold_frames(StartThreshold(config.samples_per_second));
   params.set_max_buffered_frames(MaxQueuedFrames(config.samples_per_second));
   params.set_use_fader(true);
@@ -542,14 +548,11 @@ void AudioDecoderForMixer::WritePcm(scoped_refptr<DecoderBufferBase> buffer) {
   // subtracts at most 1 frame.
   DCHECK_GT(frame_count, 0);
 
-  scoped_refptr<::net::IOBuffer> io_buffer;
-  if (resampled->data_size() >
-      buffer_pool_->buffer_size() - kAudioMessageHeaderSize) {
-    io_buffer = base::MakeRefCounted<::net::IOBuffer>(resampled->data_size() +
-                                                      kAudioMessageHeaderSize);
-  } else {
-    io_buffer = buffer_pool_->GetBuffer();
+  if (frame_count > buffer_pool_frames_) {
+    CreateBufferPool(config_, frame_count * 2);
   }
+
+  auto io_buffer = buffer_pool_->GetBuffer();
   memcpy(io_buffer->data() + kAudioMessageHeaderSize, resampled->data(),
          resampled->data_size());
 

chromecast/media/cma/backend/audio_decoder_for_mixer.h

@@ -77,6 +77,7 @@ class AudioDecoderForMixer
   void OnEosPlayed() override;
   void OnMixerError() override;
 
+  void CreateBufferPool(const AudioConfig& config, int frame_count);
   void CreateMixerInput(const AudioConfig& config, bool start_playback_asap);
   void CleanUpPcm();
   void ResetMixerInputForNewConfig(const AudioConfig& config);
@@ -99,6 +100,7 @@ class AudioDecoderForMixer
 
   Statistics stats_;
 
+  int buffer_pool_frames_ = 0;
   bool pending_buffer_complete_ = false;
   bool mixer_error_ = false;
   bool paused_ = false;

chromecast/media/cma/backend/mixer/mixer_input_connection.h

@@ -85,6 +85,7 @@ class MixerInputConnection : public mixer_service::MixerSocket::Delegate,
                          int64_t timestamp) override;
   void OnConnectionError() override;
 
+  void CreateBufferPool(int frame_count);
   void OnInactivityTimeout();
   void RestartPlaybackAt(int64_t timestamp, int64_t pts);
   void SetMediaPlaybackRate(double rate);
@@ -118,7 +119,7 @@ class MixerInputConnection : public mixer_service::MixerSocket::Delegate,
       EXCLUSIVE_LOCKS_REQUIRED(lock_);
 
   void WritePcm(scoped_refptr<net::IOBuffer> data);
-  RenderingDelay QueueData(scoped_refptr<net::IOBuffer> data)
+  int64_t QueueData(scoped_refptr<net::IOBuffer> data)
       EXCLUSIVE_LOCKS_REQUIRED(lock_);
 
   void PostPcmCompletion();
@@ -135,6 +136,7 @@ class MixerInputConnection : public mixer_service::MixerSocket::Delegate,
 
   const bool ignore_for_stream_count_;
   const int fill_size_;
+  const int algorithm_fill_size_;
   const int num_channels_;
   const int input_samples_per_second_;
   const mixer_service::SampleFormat sample_format_;
@@ -144,7 +146,7 @@ class MixerInputConnection : public mixer_service::MixerSocket::Delegate,
   const int playout_channel_;
 
   const scoped_refptr<base::SequencedTaskRunner> io_task_runner_;
-  const int max_queued_frames_;
+  int max_queued_frames_;
   // Minimum number of frames buffered before starting to fill data.
   int start_threshold_frames_;
 
@@ -154,6 +156,7 @@ class MixerInputConnection : public mixer_service::MixerSocket::Delegate,
   bool audio_ready_for_playback_fired_ = false;
   base::OneShotTimer inactivity_timer_;
   bool connection_error_ = false;
+  int buffer_pool_frames_ = 0;
 
   base::Lock lock_;
   State state_ GUARDED_BY(lock_) = State::kUninitialized;
@@ -163,7 +166,8 @@ class MixerInputConnection : public mixer_service::MixerSocket::Delegate,
   base::circular_deque<scoped_refptr<net::IOBuffer>> queue_ GUARDED_BY(lock_);
   int queued_frames_ GUARDED_BY(lock_) = 0;
   RenderingDelay mixer_rendering_delay_ GUARDED_BY(lock_);
-  RenderingDelay last_buffer_delay_ GUARDED_BY(lock_);
+  int64_t next_playback_timestamp_ GUARDED_BY(lock_) = INT64_MIN;
+  int mixer_read_size_ GUARDED_BY(lock_) = 0;
   int extra_delay_frames_ GUARDED_BY(lock_) = 0;
   int current_buffer_offset_ GUARDED_BY(lock_) = 0;
   AudioFader fader_ GUARDED_BY(lock_);

chromecast/media/cma/backend/multizone_backend_unittest.cc

@@ -368,10 +368,7 @@ void MultizoneBackendTest::OnEndOfStream() {
             kMaxRenderingDelayErrorUs);
 }
 
-// TODO(b/143135800): The new mixer backend didn't calculate the rendering delay
-// correctly when playback rate changes. Temporary disable the test to unblock
-// PoG.
-TEST_P(MultizoneBackendTest, DISABLED_RenderingDelay) {
+TEST_P(MultizoneBackendTest, RenderingDelay) {
   const TestParams& params = GetParam();
   int sample_rate = testing::get<0>(params);
   float playback_rate = testing::get<1>(params);
@@ -383,7 +380,7 @@ TEST_P(MultizoneBackendTest, DISABLED_RenderingDelay) {
   Start(playback_rate);
 }
 
-TEST_F(MultizoneBackendTest, DISABLED_RenderingDelayWithMultipleRateChanges) {
+TEST_F(MultizoneBackendTest, RenderingDelayWithMultipleRateChanges) {
   Initialize(48000 /* sample_rate */, 10 /* playback_rate_change_count */);
   AddEffectsStreams();
   Start(1.0f /* playback_rate */);