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-rw-r--r--media/filters/audio_file_reader.cc34
-rw-r--r--media/filters/audio_file_reader.h9
-rw-r--r--media/filters/audio_file_reader_unittest.cc18
-rw-r--r--media/filters/chunk_demuxer.cc2
-rw-r--r--media/filters/ffmpeg_audio_decoder.cc21
-rw-r--r--media/filters/ffmpeg_audio_decoder.h4
-rw-r--r--media/filters/pipeline_integration_test.cc9
-rw-r--r--webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.cc20
-rw-r--r--webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.h4
9 files changed, 98 insertions, 23 deletions
diff --git a/media/filters/audio_file_reader.cc b/media/filters/audio_file_reader.cc
index 58c4501..32627b0 100644
--- a/media/filters/audio_file_reader.cc
+++ b/media/filters/audio_file_reader.cc
@@ -15,21 +15,16 @@ namespace media {
AudioFileReader::AudioFileReader(FFmpegURLProtocol* protocol)
: codec_context_(NULL),
stream_index_(0),
- protocol_(protocol) {
+ protocol_(protocol),
+ channels_(0),
+ sample_rate_(0),
+ av_sample_format_(0) {
}
AudioFileReader::~AudioFileReader() {
Close();
}
-int AudioFileReader::channels() const {
- return codec_context_->channels;
-}
-
-int AudioFileReader::sample_rate() const {
- return codec_context_->sample_rate;
-}
-
base::TimeDelta AudioFileReader::duration() const {
const AVRational av_time_base = {1, AV_TIME_BASE};
@@ -110,6 +105,11 @@ bool AudioFileReader::Open() {
return false;
}
+ // Store initial values to guard against midstream configuration changes.
+ channels_ = codec_context_->channels;
+ sample_rate_ = codec_context_->sample_rate;
+ av_sample_format_ = codec_context_->sample_fmt;
+
return true;
}
@@ -179,6 +179,22 @@ int AudioFileReader::Read(AudioBus* audio_bus) {
break;
}
+ if (av_frame->sample_rate != sample_rate_ ||
+ av_frame->channels != channels_ ||
+ av_frame->format != av_sample_format_) {
+ DLOG(ERROR) << "Unsupported midstream configuration change!"
+ << " Sample Rate: " << av_frame->sample_rate << " vs "
+ << sample_rate_
+ << ", Channels: " << av_frame->channels << " vs "
+ << channels_
+ << ", Sample Format: " << av_frame->format << " vs "
+ << av_sample_format_;
+
+ // This is an unrecoverable error, so bail out.
+ 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;
diff --git a/media/filters/audio_file_reader.h b/media/filters/audio_file_reader.h
index c9996f39..e345dc0 100644
--- a/media/filters/audio_file_reader.h
+++ b/media/filters/audio_file_reader.h
@@ -43,8 +43,8 @@ class MEDIA_EXPORT AudioFileReader {
int Read(AudioBus* audio_bus);
// These methods can be called once Open() has been called.
- int channels() const;
- int sample_rate() const;
+ int channels() const { return channels_; }
+ int sample_rate() const { return sample_rate_; }
// Please note that duration() and number_of_frames() attempt to be accurate,
// but are only estimates. For some encoded formats, the actual duration
@@ -58,6 +58,11 @@ class MEDIA_EXPORT AudioFileReader {
AVCodecContext* codec_context_;
int stream_index_;
FFmpegURLProtocol* protocol_;
+ int channels_;
+ int sample_rate_;
+
+ // AVSampleFormat initially requested; not Chrome's SampleFormat.
+ int av_sample_format_;
DISALLOW_COPY_AND_ASSIGN(AudioFileReader);
};
diff --git a/media/filters/audio_file_reader_unittest.cc b/media/filters/audio_file_reader_unittest.cc
index 72c2603..64c58e0 100644
--- a/media/filters/audio_file_reader_unittest.cc
+++ b/media/filters/audio_file_reader_unittest.cc
@@ -73,11 +73,19 @@ class AudioFileReaderTest : public testing::Test {
ReadAndVerify(hash, trimmed_frames);
}
- void RunFailingTest(const char* fn) {
+ void RunTestFailingDemux(const char* fn) {
Initialize(fn);
EXPECT_FALSE(reader_->Open());
}
+ void RunTestFailingDecode(const char* fn) {
+ Initialize(fn);
+ EXPECT_TRUE(reader_->Open());
+ scoped_ptr<AudioBus> decoded_audio_data = AudioBus::Create(
+ reader_->channels(), reader_->number_of_frames());
+ EXPECT_EQ(reader_->Read(decoded_audio_data.get()), 0);
+ }
+
protected:
scoped_refptr<DecoderBuffer> data_;
scoped_ptr<InMemoryUrlProtocol> protocol_;
@@ -91,7 +99,7 @@ TEST_F(AudioFileReaderTest, WithoutOpen) {
}
TEST_F(AudioFileReaderTest, InvalidFile) {
- RunFailingTest("ten_byte_file");
+ RunTestFailingDemux("ten_byte_file");
}
TEST_F(AudioFileReaderTest, WithVideo) {
@@ -134,10 +142,14 @@ TEST_F(AudioFileReaderTest, AAC) {
RunTest("sfx.m4a", NULL, 1, 44100,
base::TimeDelta::FromMicroseconds(312001), 13759, 13312);
}
+
+TEST_F(AudioFileReaderTest, MidStreamConfigChangesFail) {
+ RunTestFailingDecode("midstream_config_change.mp3");
+}
#endif
TEST_F(AudioFileReaderTest, VorbisInvalidChannelLayout) {
- RunFailingTest("9ch.ogg");
+ RunTestFailingDemux("9ch.ogg");
}
TEST_F(AudioFileReaderTest, WaveValidFourChannelLayout) {
diff --git a/media/filters/chunk_demuxer.cc b/media/filters/chunk_demuxer.cc
index 1f091c3..8162bbf 100644
--- a/media/filters/chunk_demuxer.cc
+++ b/media/filters/chunk_demuxer.cc
@@ -218,7 +218,7 @@ class ChunkDemuxerStream : public DemuxerStream {
// Append() belong to a media segment that starts at |start_timestamp|.
void OnNewMediaSegment(TimeDelta start_timestamp);
- // Called when mid-stream config updates occur.
+ // Called when midstream config updates occur.
// Returns true if the new config is accepted.
// Returns false if the new config should trigger an error.
bool UpdateAudioConfig(const AudioDecoderConfig& config);
diff --git a/media/filters/ffmpeg_audio_decoder.cc b/media/filters/ffmpeg_audio_decoder.cc
index be142d5..6180063 100644
--- a/media/filters/ffmpeg_audio_decoder.cc
+++ b/media/filters/ffmpeg_audio_decoder.cc
@@ -43,7 +43,9 @@ FFmpegAudioDecoder::FFmpegAudioDecoder(
codec_context_(NULL),
bits_per_channel_(0),
channel_layout_(CHANNEL_LAYOUT_NONE),
+ channels_(0),
samples_per_second_(0),
+ av_sample_format_(0),
bytes_per_frame_(0),
last_input_timestamp_(kNoTimestamp()),
output_bytes_to_drop_(0),
@@ -303,6 +305,11 @@ bool FFmpegAudioDecoder::ConfigureDecoder() {
output_timestamp_helper_.reset(new AudioTimestampHelper(
config.bytes_per_frame(), config.samples_per_second()));
bytes_per_frame_ = config.bytes_per_frame();
+
+ // Store initial values to guard against midstream configuration changes.
+ channels_ = codec_context_->channels;
+ av_sample_format_ = codec_context_->sample_fmt;
+
return true;
}
@@ -387,10 +394,16 @@ void FFmpegAudioDecoder::RunDecodeLoop(
int decoded_audio_size = 0;
if (frame_decoded) {
- int output_sample_rate = av_frame_->sample_rate;
- if (output_sample_rate != samples_per_second_) {
- DLOG(ERROR) << "Output sample rate (" << output_sample_rate
- << ") doesn't match expected rate " << samples_per_second_;
+ if (av_frame_->sample_rate != samples_per_second_ ||
+ av_frame_->channels != channels_ ||
+ av_frame_->format != av_sample_format_) {
+ DLOG(ERROR) << "Unsupported midstream configuration change!"
+ << " Sample Rate: " << av_frame_->sample_rate << " vs "
+ << samples_per_second_
+ << ", Channels: " << av_frame_->channels << " vs "
+ << channels_
+ << ", Sample Format: " << av_frame_->format << " vs "
+ << av_sample_format_;
// This is an unrecoverable error, so bail out.
QueuedAudioBuffer queue_entry = { kDecodeError, NULL };
diff --git a/media/filters/ffmpeg_audio_decoder.h b/media/filters/ffmpeg_audio_decoder.h
index d2ba8c5..99fef1b 100644
--- a/media/filters/ffmpeg_audio_decoder.h
+++ b/media/filters/ffmpeg_audio_decoder.h
@@ -65,8 +65,12 @@ class MEDIA_EXPORT FFmpegAudioDecoder : public AudioDecoder {
// Decoded audio format.
int bits_per_channel_;
ChannelLayout channel_layout_;
+ int channels_;
int samples_per_second_;
+ // AVSampleFormat initially requested; not Chrome's SampleFormat.
+ int av_sample_format_;
+
// Used for computing output timestamps.
scoped_ptr<AudioTimestampHelper> output_timestamp_helper_;
int bytes_per_frame_;
diff --git a/media/filters/pipeline_integration_test.cc b/media/filters/pipeline_integration_test.cc
index 3e2eb19..e96da57 100644
--- a/media/filters/pipeline_integration_test.cc
+++ b/media/filters/pipeline_integration_test.cc
@@ -660,6 +660,15 @@ TEST_F(PipelineIntegrationTest,
EXPECT_EQ(PIPELINE_ERROR_DECODE, WaitUntilEndedOrError());
source.Abort();
}
+
+// Verify files which change configuration midstream fail gracefully.
+TEST_F(PipelineIntegrationTest, MidStreamConfigChangesFail) {
+ ASSERT_TRUE(Start(
+ GetTestDataFilePath("midstream_config_change.mp3"), PIPELINE_OK));
+ Play();
+ ASSERT_EQ(WaitUntilEndedOrError(), PIPELINE_ERROR_DECODE);
+}
+
#endif
TEST_F(PipelineIntegrationTest, BasicPlayback_16x9AspectRatio) {
diff --git a/webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.cc b/webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.cc
index 8edbb01..d7f3f27 100644
--- a/webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.cc
+++ b/webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.cc
@@ -87,6 +87,8 @@ FFmpegCdmAudioDecoder::FFmpegCdmAudioDecoder(cdm::Host* host)
av_frame_(NULL),
bits_per_channel_(0),
samples_per_second_(0),
+ channels_(0),
+ av_sample_format_(0),
bytes_per_frame_(0),
last_input_timestamp_(media::kNoTimestamp()),
output_bytes_to_drop_(0) {
@@ -154,6 +156,10 @@ bool FFmpegCdmAudioDecoder::Initialize(const cdm::AudioDecoderConfig& config) {
serialized_audio_frames_.reserve(bytes_per_frame_ * samples_per_second_);
is_initialized_ = true;
+ // Store initial values to guard against midstream configuration changes.
+ channels_ = codec_context_->channels;
+ av_sample_format_ = codec_context_->sample_fmt;
+
return true;
}
@@ -269,10 +275,16 @@ cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
int decoded_audio_size = 0;
if (frame_decoded) {
- int output_sample_rate = av_frame_->sample_rate;
- if (output_sample_rate != samples_per_second_) {
- DLOG(ERROR) << "Output sample rate (" << output_sample_rate
- << ") doesn't match expected rate " << samples_per_second_;
+ if (av_frame_->sample_rate != samples_per_second_ ||
+ av_frame_->channels != channels_ ||
+ av_frame_->format != av_sample_format_) {
+ DLOG(ERROR) << "Unsupported midstream configuration change!"
+ << " Sample Rate: " << av_frame_->sample_rate << " vs "
+ << samples_per_second_
+ << ", Channels: " << av_frame_->channels << " vs "
+ << channels_
+ << ", Sample Format: " << av_frame_->format << " vs "
+ << av_sample_format_;
return cdm::kDecodeError;
}
diff --git a/webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.h b/webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.h
index 266b4ad..1c2c819 100644
--- a/webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.h
+++ b/webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.h
@@ -69,6 +69,10 @@ class FFmpegCdmAudioDecoder {
// Audio format.
int bits_per_channel_;
int samples_per_second_;
+ int channels_;
+
+ // AVSampleFormat initially requested; not Chrome's SampleFormat.
+ int av_sample_format_;
// Used for computing output timestamps.
scoped_ptr<media::AudioTimestampHelper> output_timestamp_helper_;
generated by cgit v1.1 at 2024-10-30 12:04:09 +0000