Roll FFMpeg for M26. Fix ffmpeg float audio decoding.

media/filters/audio_file_reader.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/audio_file_reader.h"
 
 #include "base/logging.h"
 #include "base/time.h"
 #include "media/base/audio_bus.h"
 #include "media/ffmpeg/ffmpeg_common.h"
@@ skipping 61 matching lines @@
 
   int result = avformat_find_stream_info(format_context, NULL);
   if (result < 0) {
     DLOG(WARNING)
         << "AudioFileReader::Open() : error in avformat_find_stream_info()";
     return false;
   }
 
   AVCodec* codec = avcodec_find_decoder(codec_context_->codec_id);
   if (codec) {
+    // MP3 decodes to S16P which we don't support, tell it to use S16 instead.

[scherkus (not reviewing), 2012/12/13 22:40:11]

should we also be checking the codec id?

[DaleCurtis, 2012/12/13 22:51:28]

We could but, but this is more generic and the check if() check on the sample
format will catch failures.

+    bool request_s16_format = codec_context_->sample_fmt == AV_SAMPLE_FMT_S16P;

[scherkus (not reviewing), 2012/12/13 22:40:11]

how about instead of a bool (which I found a bit tricky to follow), we use the
notion of an expected format? should we need to change/expand the code it'll fit
in nicely


AVSampleFormat expected = codec_context_->sample_fmt;

// MP3 blah blah blah
if (codec_context_->sample_fmt == AV_SAMPLE_FMT_S16P) {
  expected = AV_SAMPLE_FMT_S16;
} else if (some other requirement) {
   expected = // etc...
}

codec_context_->sample_fmt = expected;

// open

if (codec_context_->sample_fmt != expected) {
  // blah blah
}

[DaleCurtis, 2012/12/13 22:51:28]

Sounds good. Will do so in next patch set.

[DaleCurtis, 2012/12/14 20:48:32]

Done.

+    if (request_s16_format)
+      codec_context_->request_sample_fmt = AV_SAMPLE_FMT_S16;
+
     if ((result = avcodec_open2(codec_context_, codec, NULL)) < 0) {
       DLOG(WARNING) << "AudioFileReader::Open() : could not open codec -"
-          << " result: " << result;
+                    << " result: " << result;
+      return false;
+    }
+
+    // Ensure avcodec_open2() respected our format request.
+    if (request_s16_format && codec_context_->sample_fmt != AV_SAMPLE_FMT_S16) {
+      DLOG(ERROR) << "AudioFileReader::Open() : unable to configure a"
+                  << " supported sample format - "
+                  << codec_context_->sample_fmt;
       return false;
     }
   } else {
     DLOG(WARNING) << "AudioFileReader::Open() : could not find codec -"
-        << " result: " << result;
+                  << " result: " << result;
     return false;
   }
 
   return true;
 }
 
 void AudioFileReader::Close() {
   if (codec_context_) {
     avcodec_close(codec_context_);
     codec_context_ = NULL;
@@ skipping 56 matching lines @@
       int frames_read = av_frame->nb_samples;
       if (frames_read < 0) {
         continue_decoding = false;
         break;
       }
 
       // Truncate, if necessary, if the destination isn't big enough.
       if (current_frame + frames_read > audio_bus->frames())
         frames_read = audio_bus->frames() - current_frame;
 
-      // Deinterleave each channel and convert to 32bit floating-point
-      // with nominal range -1.0 -> +1.0.
-      audio_bus->FromInterleavedPartial(
-          av_frame->data[0], current_frame, frames_read, bytes_per_sample);
+      // Deinterleave each channel and convert to 32bit floating-point with
+      // nominal range -1.0 -> +1.0.  If the output is already in float planar
+      // format, just copy it into the AudioBus.
+      if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT) {
+        float* decoded_audio_data = reinterpret_cast<float*>(av_frame->data[0]);
+        int channels = audio_bus->channels();
+        for (int ch = 0; ch < channels; ++ch) {
+          float* bus_data = audio_bus->channel(ch) + current_frame;
+          for (int i = 0, offset = ch; i < frames_read;
+               ++i, offset += channels) {
+            bus_data[i] = decoded_audio_data[offset];
+          }
+        }
+      } else if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLTP) {
+        for (int ch = 0; ch < audio_bus->channels(); ++ch) {
+          memcpy(audio_bus->channel(ch) + current_frame,
+                 av_frame->extended_data[ch], sizeof(float) * frames_read);
+        }
+      } else {
+        audio_bus->FromInterleavedPartial(
+            av_frame->data[0], current_frame, frames_read, bytes_per_sample);
+      }
 
       current_frame += frames_read;
     } while (packet_temp.size > 0);
     av_free_packet(&packet);
   }
 
   // Zero any remaining frames.
   audio_bus->ZeroFramesPartial(
       current_frame, audio_bus->frames() - current_frame);
 
   // Returns the actual number of sample-frames decoded.
   // Ideally this represents the "true" exact length of the file.
   return current_frame;
 }
 
 }  // namespace media