FFmpeg fixups for M33 roll. Now with more Opus!

media/filters/opus_audio_decoder.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/filters/opus_audio_decoder.h"
 
 #include <cmath>
 
 #include "base/bind.h"
 #include "base/callback_helpers.h"
@@ skipping 441 matching lines @@
     DVLOG(1) << "\tsample_rate : " << samples_per_second_
              << " -> " << config.samples_per_second();
     return false;
   }
 
   // Clean up existing decoder if necessary.
   CloseDecoder();
 
   // Allocate the output buffer if necessary.
   if (!output_buffer_)
-    output_buffer_.reset(new int16[kMaxOpusOutputPacketSizeSamples]);
+    output_buffer_.reset(new float[kMaxOpusOutputPacketSizeSamples]);
 
   // Parse the Opus Extra Data.
   OpusExtraData opus_extra_data;
   if (!ParseOpusExtraData(config.extra_data(), config.extra_data_size(),
                           config,
                           &opus_extra_data))
     return false;
 
   if (!config.codec_delay().InMicroseconds())
     return false;
@@ skipping 55 matching lines @@
 void OpusAudioDecoder::ResetTimestampState() {
   output_timestamp_helper_->SetBaseTimestamp(kNoTimestamp());
   last_input_timestamp_ = kNoTimestamp();
   frames_to_discard_ = TimeDeltaToAudioFrames(
       demuxer_stream_->audio_decoder_config().seek_preroll(),
       samples_per_second_);
 }
 
 bool OpusAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& input,
                               scoped_refptr<AudioBuffer>* output_buffer) {
-  int frames_decoded = opus_multistream_decode(opus_decoder_,
-                                               input->data(),
-                                               input->data_size(),
-                                               &output_buffer_[0],
-                                               kMaxOpusOutputPacketSizeSamples,
-                                               0);
+  int frames_decoded =
+      opus_multistream_decode_float(opus_decoder_,
+                                    input->data(),
+                                    input->data_size(),
+                                    &output_buffer_[0],
+                                    kMaxOpusOutputPacketSizeSamples,
+                                    0);
   if (frames_decoded < 0) {
     DVLOG(0) << "opus_multistream_decode failed for"
              << " timestamp: " << input->timestamp().InMicroseconds()
              << " us, duration: " << input->duration().InMicroseconds()
              << " us, packet size: " << input->data_size() << " bytes with"
              << " status: " << opus_strerror(frames_decoded);
     return false;
   }
 
   uint8* decoded_audio_data = reinterpret_cast<uint8*>(&output_buffer_[0]);
@@ skipping 11 matching lines @@
   // there is a seek to zero.
   // TODO(vigneshv): This should be checked for start of stream rather than
   // input timestamp of zero to accomodate streams that don't start at zero.
   if (input->timestamp() == base::TimeDelta())
     frames_to_discard_ = frame_delay_at_start_;
 
   if (bytes_decoded > 0 && frames_decoded > frames_to_discard_) {
     // Copy the audio samples into an output buffer.
     uint8* data[] = { decoded_audio_data };
     *output_buffer = AudioBuffer::CopyFrom(
-        kSampleFormatS16,
+        kSampleFormatF32,
         ChannelLayoutToChannelCount(channel_layout_),
         frames_decoded,
         data,
         output_timestamp_helper_->GetTimestamp() - timestamp_offset_,
         output_timestamp_helper_->GetFrameDuration(frames_decoded));
     output_timestamp_helper_->AddFrames(frames_decoded);
     if (frames_to_discard_ > 0) {
       output_buffer->get()->TrimStart(frames_to_discard_);
       frames_decoded -= frames_to_discard_;
       frames_to_discard_ = 0;
     }
     if (input->discard_padding().InMicroseconds() > 0) {
       int discard_padding = TimeDeltaToAudioFrames(input->discard_padding(),
                                                    samples_per_second_);
       if (discard_padding < 0 || discard_padding > frames_decoded) {
         DVLOG(1) << "Invalid file. Incorrect discard padding value.";
         return false;
       }
       output_buffer->get()->TrimEnd(discard_padding);
       frames_decoded -= discard_padding;
     }
   } else if (bytes_decoded > 0) {
     frames_to_discard_ -= frames_decoded;
     frames_decoded = 0;
   }
 
   // Decoding finished successfully, update statistics.
   PipelineStatistics statistics;
-  statistics.audio_bytes_decoded =
-      frames_decoded *
-      demuxer_stream_->audio_decoder_config().bytes_per_frame();
+  statistics.audio_bytes_decoded = bytes_decoded;

[vignesh, 2013/12/11 20:10:18]

not sure if this change is required. even though we decode bytes_decoded, we
don't give out that many bytes out of the decoder in some cases. so i think it
should be left as it is. acolwell@, thoughts?

[acolwell GONE FROM CHROMIUM, 2013/12/11 20:22:32]

This is actually still wrong. This stat is supposed to represent the number of
compressed bytes decoded. If I understand this code correctly, it should be the
following since it appears that opus_multistream_decode_float() consumes all of
the input data.

statistics.audio_bytes_decoded = input->data_size();

[DaleCurtis, 2013/12/11 22:19:36]

Done.

   statistics_cb_.Run(statistics);
 
+  // Discard the buffer to indicate we need more data.
+  if (!frames_decoded)
+    *output_buffer = NULL;
+
   return true;
 }
 
 }  // namespace media