media: Opus support for WebM in Media Source

media/filters/opus_audio_decoder.cc

Index: media/filters/opus_audio_decoder.cc
diff --git a/media/filters/opus_audio_decoder.cc b/media/filters/opus_audio_decoder.cc
index 115799ab71193fe829265268c3898a5cbbc6cf95..a79fb8847b7cd07eb14f8f9a3de2e037b9cf9252 100644
--- a/media/filters/opus_audio_decoder.cc
+++ b/media/filters/opus_audio_decoder.cc
@@ -250,7 +250,6 @@ OpusAudioDecoder::OpusAudioDecoder(
       channel_layout_(CHANNEL_LAYOUT_NONE),
       samples_per_second_(0),
       last_input_timestamp_(kNoTimestamp()),
-      output_bytes_to_drop_(0),
       skip_samples_(0) {
 }
 
@@ -457,10 +456,17 @@ bool OpusAudioDecoder::ConfigureDecoder() {
                   config,
                   &opus_header);
 
-  skip_samples_ = opus_header.skip_samples;
-
-  if (skip_samples_ > 0)
-    output_bytes_to_drop_ = skip_samples_ * config.bytes_per_frame();
+  if (!config.codec_delay()) {
+    skip_samples_ = opus_header.skip_samples;

[fgalligan1, 2013/08/26 21:10:06]

Is this the same value as below? I.e. skipped_bytes = opus_header.skip_samples *
bytesPerSample * numChannels?

[vignesh, 2013/08/26 21:44:23]

No it is not. If you look below (Line 542 in the red diff), i have modified the
Start Trimming code to make use of AudioBuffer.TrimStart() rather than having
our own implementation of it. The TrimStart function expects number of samples
and not number of bytes. So that conversion is not necessary anymore.

[fgalligan1, 2013/08/26 22:11:53]

Sorry I was just asking if opus_header.skip_samples included numChannels or not

[vignesh, 2013/08/26 23:55:48]

No it does not. Neither does the skip_samples_ variable in the else block. The
TrimStart and TrimEnd functions in AudioBuffer takes numChannels into account.
So we just call them with the number of samples.

+  } else {
+    // WebM container stores codec delay in nanoseconds. Convert it to samples.
+    skip_samples_ = config.codec_delay() * config.samples_per_second() /
+                    1000000000;
+    if (skip_samples_ != opus_header.skip_samples) {
+      DVLOG(1) << "Codec Delay in container does not match the value in Opus "
+               << "header. Using the value in container.";
+    }
+  }
 
   uint8 channel_mapping[kMaxVorbisChannels];
   memcpy(&channel_mapping,
@@ -508,7 +514,7 @@ void OpusAudioDecoder::CloseDecoder() {
 void OpusAudioDecoder::ResetTimestampState() {
   output_timestamp_helper_->SetBaseTimestamp(kNoTimestamp());
   last_input_timestamp_ = kNoTimestamp();
-  output_bytes_to_drop_ = 0;
+  skip_samples_ = 0;
 }
 
 bool OpusAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& input,
@@ -539,16 +545,6 @@ bool OpusAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& input,
     output_timestamp_helper_->SetBaseTimestamp(input->timestamp());
   }
 
-  if (decoded_audio_size > 0 && output_bytes_to_drop_ > 0) {
-    int dropped_size = std::min(decoded_audio_size, output_bytes_to_drop_);
-    DCHECK_EQ(dropped_size % kBytesPerChannel, 0);
-    decoded_audio_data += dropped_size;
-    decoded_audio_size -= dropped_size;
-    output_bytes_to_drop_ -= dropped_size;
-    samples_decoded = decoded_audio_size /
-                      demuxer_stream_->audio_decoder_config().bytes_per_frame();
-  }
-
   if (decoded_audio_size > 0) {
     // Copy the audio samples into an output buffer.
     uint8* data[] = { decoded_audio_data };
@@ -560,6 +556,17 @@ bool OpusAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& input,
         output_timestamp_helper_->GetTimestamp(),
         output_timestamp_helper_->GetFrameDuration(samples_decoded));
     output_timestamp_helper_->AddFrames(samples_decoded);
+    if (skip_samples_ > 0) {
+      int dropped_size = std::min(samples_decoded, skip_samples_);
+      output_buffer->get()->TrimStart(dropped_size);

[fgalligan1, 2013/08/26 21:10:06]

I'm assuming this works on samples? Does it take into account numChannels?

[vignesh, 2013/08/26 21:44:23]

Yes, it works on samples. I am unsure if it takes numChannels into account. Will
get back on this shortly.

[vignesh, 2013/08/26 23:55:48]

As mentioned in the other comment in this file, this function accoutns for
number of channels.

+      skip_samples_ -= dropped_size;
+      samples_decoded -= dropped_size;
+    }
+    if (input->discard_padding() > 0) {
+      int discard_padding = input->discard_padding() * samples_per_second_ /
+                            1000000000;
+      output_buffer->get()->TrimEnd(std::min(samples_decoded, discard_padding));
+    }
   }
 
   // Decoding finished successfully, update statistics.