Enable splice frames and partial appends for AAC; with test!

Fingers crossed, hopefully all known bugs are fixed.

BUG=356073,371633,369204
TEST=new unittest.

Review URL: https://codereview.chromium.org/278673003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@274457 0039d316-1c4b-4281-b951-d872f2087c98

media/filters/chunk_demuxer.cc

@@ -860,15 +860,13 @@ bool ChunkDemuxerStream::UpdateAudioConfig(const AudioDecoderConfig& config,
     DCHECK_EQ(state_, UNINITIALIZED);
 
     // On platforms which support splice frames, enable splice frames and
-    // partial append window support for a limited set of codecs.
-    // TODO(dalecurtis): Verify this works for codecs other than MP3 and Vorbis.
-    // Right now we want to be extremely conservative to ensure we don't break
-    // the world.
-    const bool mp3_or_vorbis =
-        config.codec() == kCodecMP3 || config.codec() == kCodecVorbis;
-    splice_frames_enabled_ = splice_frames_enabled_ && mp3_or_vorbis;
+    // partial append window support for most codecs (notably: not opus).
+    const bool codec_supported = config.codec() == kCodecMP3 ||
+                                 config.codec() == kCodecAAC ||
+                                 config.codec() == kCodecVorbis;
+    splice_frames_enabled_ = splice_frames_enabled_ && codec_supported;
     partial_append_window_trimming_enabled_ =
-        splice_frames_enabled_ && mp3_or_vorbis;
+        splice_frames_enabled_ && codec_supported;
 
     stream_.reset(
         new SourceBufferStream(config, log_cb, splice_frames_enabled_));

media/filters/pipeline_integration_test.cc

@@ -919,24 +919,35 @@ TEST_P(PipelineIntegrationTest, MediaSource_ADTS) {
 
 TEST_P(PipelineIntegrationTest, MediaSource_ADTS_TimestampOffset) {
   MockMediaSource source("sfx.adts", kADTS, kAppendWholeFile, GetParam());
-  StartPipelineWithMediaSource(&source);
+  StartHashedPipelineWithMediaSource(&source);
   EXPECT_EQ(325, source.last_timestamp_offset().InMilliseconds());
 
+  // Trim off multiple frames off the beginning of the segment which will cause
+  // the first decoded frame to be incorrect if preroll isn't implemented.
+  const base::TimeDelta adts_preroll_duration =
+      base::TimeDelta::FromSecondsD(2.5 * 1024 / 44100);
+  const base::TimeDelta append_time =
+      source.last_timestamp_offset() - adts_preroll_duration;
+
   scoped_refptr<DecoderBuffer> second_file = ReadTestDataFile("sfx.adts");
-  source.AppendAtTime(
-      source.last_timestamp_offset() - base::TimeDelta::FromMilliseconds(10),
-      second_file->data(),
-      second_file->data_size());
+  source.AppendAtTimeWithWindow(append_time,
+                                append_time + adts_preroll_duration,
+                                kInfiniteDuration(),
+                                second_file->data(),
+                                second_file->data_size());
   source.EndOfStream();
 
-  EXPECT_EQ(640, source.last_timestamp_offset().InMilliseconds());
+  EXPECT_EQ(592, source.last_timestamp_offset().InMilliseconds());
   EXPECT_EQ(1u, pipeline_->GetBufferedTimeRanges().size());
   EXPECT_EQ(0, pipeline_->GetBufferedTimeRanges().start(0).InMilliseconds());
-  EXPECT_EQ(640, pipeline_->GetBufferedTimeRanges().end(0).InMilliseconds());
+  EXPECT_EQ(592, pipeline_->GetBufferedTimeRanges().end(0).InMilliseconds());
 
   Play();
 
   EXPECT_TRUE(WaitUntilOnEnded());
+
+  // Verify preroll is stripped.
+  EXPECT_EQ("-0.06,0.97,-0.90,-0.70,-0.53,-0.34,", GetAudioHash());
 }
 
 TEST_F(PipelineIntegrationTest, BasicPlaybackHashed_MP3) {