Index: media/filters/ffmpeg_audio_decoder.cc |
diff --git a/media/filters/ffmpeg_audio_decoder.cc b/media/filters/ffmpeg_audio_decoder.cc |
index bbac958043379519f3c6642a691e20dbd9d47c60..f3869fef205bb13fc2f6cfed670e06c8e1ae2c56 100644 |
--- a/media/filters/ffmpeg_audio_decoder.cc |
+++ b/media/filters/ffmpeg_audio_decoder.cc |
@@ -56,7 +56,15 @@ FFmpegAudioDecoder::FFmpegAudioDecoder(MessageLoop* message_loop) |
pending_reads_(0) { |
} |
-FFmpegAudioDecoder::~FFmpegAudioDecoder() {} |
+FFmpegAudioDecoder::~FFmpegAudioDecoder() { |
+ // XXX: should we require Stop() to be called? this might end up getting |
+ // called on a random thread due to refcounting. |
+ if (codec_context_) { |
+ av_free(codec_context_->extradata); |
+ avcodec_close(codec_context_); |
+ av_free(codec_context_); |
+ } |
+} |
void FFmpegAudioDecoder::Flush(FilterCallback* callback) { |
message_loop_->PostTask( |
@@ -102,32 +110,32 @@ void FFmpegAudioDecoder::DoInitialize( |
scoped_ptr<FilterCallback> c(callback); |
demuxer_stream_ = stream; |
- AVStream* av_stream = demuxer_stream_->GetAVStream(); |
- CHECK(av_stream); |
- |
+ AudioDecoderConfig* config = stream->audio_decoder_config(); |
Ami GONE FROM CHROMIUM
2011/09/12 20:54:21
Hopefully this can be a const& or at least a const
scherkus (not reviewing)
2011/09/19 21:19:45
Done.
|
stats_callback_.reset(stats_callback); |
- // Grab the AVStream's codec context and make sure we have sensible values. |
- codec_context_ = av_stream->codec; |
- int bps = av_get_bits_per_sample_fmt(codec_context_->sample_fmt); |
- if (codec_context_->channels <= 0 || |
- codec_context_->channels > Limits::kMaxChannels || |
- (codec_context_->channel_layout == 0 && codec_context_->channels > 2) || |
- bps <= 0 || bps > Limits::kMaxBitsPerSample || |
- codec_context_->sample_rate <= 0 || |
- codec_context_->sample_rate > Limits::kMaxSampleRate) { |
+ // Sanity checking. |
Ami GONE FROM CHROMIUM
2011/09/12 20:54:21
Shouldn't this be happening earlier (ideally durin
scherkus (not reviewing)
2011/09/19 21:19:45
I'm torn.. on one hand you could argue that said l
|
+ if (config->codec() == kUnknownAudioCodec || |
+ config->channel_layout() == CHANNEL_LAYOUT_UNSUPPORTED || |
+ config->bits_per_channel() <= 0 || |
+ config->bits_per_channel() > Limits::kMaxBitsPerSample || |
+ config->sample_rate() <= 0 || |
+ config->sample_rate() > Limits::kMaxSampleRate) { |
DLOG(ERROR) << "Invalid audio stream -" |
- << " channels: " << codec_context_->channels |
- << " channel layout:" << codec_context_->channel_layout |
- << " bps: " << bps |
- << " sample rate: " << codec_context_->sample_rate; |
+ << " codec: " << config->codec() |
+ << " channel layout: " << config->channel_layout() |
+ << " bits per channel: " << config->bits_per_channel() |
+ << " sample rate: " << config->sample_rate(); |
+ // XXX: format error? |
host()->SetError(PIPELINE_ERROR_DECODE); |
callback->Run(); |
return; |
} |
- // Serialize calls to avcodec_open(). |
+ // Initialize AVCodecContext structure. |
+ codec_context_ = avcodec_alloc_context(); |
+ AudioDecoderConfigToAVCodecContext(config, codec_context_); |
+ |
AVCodec* codec = avcodec_find_decoder(codec_context_->codec_id); |
if (!codec || avcodec_open(codec_context_, codec) < 0) { |
DLOG(ERROR) << "Could not initialize audio decoder: " |
@@ -139,11 +147,9 @@ void FFmpegAudioDecoder::DoInitialize( |
} |
// Success! |
- bits_per_channel_= av_get_bits_per_sample_fmt(codec_context_->sample_fmt); |
- channel_layout_ = |
- ChannelLayoutToChromeChannelLayout(codec_context_->channel_layout, |
- codec_context_->channels); |
- sample_rate_ = codec_context_->sample_rate; |
+ bits_per_channel_ = config->bits_per_channel(); |
+ channel_layout_ = config->channel_layout(); |
+ sample_rate_ = config->sample_rate(); |
callback->Run(); |
} |