#include "media/base/audio_buffer.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_decoder_config.h"
-#include "media/base/audio_timestamp_helper.h"
+#include "media/base/audio_discard_helper.h"
#include "media/base/bind_to_current_loop.h"
#include "media/base/decoder_buffer.h"
#include "media/base/limits.h"
}
FFmpegAudioDecoder::FFmpegAudioDecoder(
- const scoped_refptr<base::SingleThreadTaskRunner>& task_runner)
+ const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
+ const LogCB& log_cb)
: task_runner_(task_runner),
state_(kUninitialized),
av_sample_format_(0),
- last_input_timestamp_(kNoTimestamp()),
- output_frames_to_drop_(0) {}
+ log_cb_(log_cb) {
+}
FFmpegAudioDecoder::~FFmpegAudioDecoder() {
DCHECK_EQ(state_, kUninitialized);
task_runner_->PostTask(FROM_HERE, closure);
}
-void FFmpegAudioDecoder::Stop(const base::Closure& closure) {
+void FFmpegAudioDecoder::Stop() {
DCHECK(task_runner_->BelongsToCurrentThread());
- base::ScopedClosureRunner runner(BindToCurrentLoop(closure));
if (state_ == kUninitialized)
return;
// Make sure we are notified if http://crbug.com/49709 returns. Issue also
// occurs with some damaged files.
- if (!buffer->end_of_stream() && buffer->timestamp() == kNoTimestamp() &&
- output_timestamp_helper_->base_timestamp() == kNoTimestamp()) {
+ if (!buffer->end_of_stream() && buffer->timestamp() == kNoTimestamp()) {
DVLOG(1) << "Received a buffer without timestamps!";
decode_cb.Run(kDecodeError, NULL);
return;
}
- if (!buffer->end_of_stream()) {
- if (last_input_timestamp_ == kNoTimestamp() &&
- codec_context_->codec_id == AV_CODEC_ID_VORBIS &&
- buffer->timestamp() < base::TimeDelta()) {
- // Dropping frames for negative timestamps as outlined in section A.2
- // in the Vorbis spec. http://xiph.org/vorbis/doc/Vorbis_I_spec.html
- output_frames_to_drop_ = floor(0.5 + -buffer->timestamp().InSecondsF() *
- config_.samples_per_second());
- } else {
- if (last_input_timestamp_ != kNoTimestamp() &&
- buffer->timestamp() < last_input_timestamp_) {
- const base::TimeDelta diff =
- buffer->timestamp() - last_input_timestamp_;
- DLOG(WARNING)
- << "Input timestamps are not monotonically increasing! "
- << " ts " << buffer->timestamp().InMicroseconds() << " us"
- << " diff " << diff.InMicroseconds() << " us";
- }
-
- last_input_timestamp_ = buffer->timestamp();
- }
+ if (!buffer->end_of_stream() && !discard_helper_->initialized() &&
+ codec_context_->codec_id == AV_CODEC_ID_VORBIS &&
+ buffer->timestamp() < base::TimeDelta()) {
+ // Dropping frames for negative timestamps as outlined in section A.2
+ // in the Vorbis spec. http://xiph.org/vorbis/doc/Vorbis_I_spec.html
+ const int discard_frames =
+ discard_helper_->TimeDeltaToFrames(-buffer->timestamp());
+ discard_helper_->Reset(discard_frames);
}
// Transition to kFlushCodec on the first end of stream buffer.
bool FFmpegAudioDecoder::FFmpegDecode(
const scoped_refptr<DecoderBuffer>& buffer) {
-
DCHECK(queued_audio_.empty());
AVPacket packet;
// skipping end of stream packets since they have a size of zero.
do {
int frame_decoded = 0;
- int result = avcodec_decode_audio4(
+ const int result = avcodec_decode_audio4(
codec_context_.get(), av_frame_.get(), &frame_decoded, &packet);
if (result < 0) {
packet.size -= result;
packet.data += result;
- if (output_timestamp_helper_->base_timestamp() == kNoTimestamp() &&
- !buffer->end_of_stream()) {
- DCHECK(buffer->timestamp() != kNoTimestamp());
- if (output_frames_to_drop_ > 0) {
- // Currently Vorbis is the only codec that causes us to drop samples.
- // If we have to drop samples it always means the timeline starts at 0.
- DCHECK_EQ(codec_context_->codec_id, AV_CODEC_ID_VORBIS);
- output_timestamp_helper_->SetBaseTimestamp(base::TimeDelta());
- } else {
- output_timestamp_helper_->SetBaseTimestamp(buffer->timestamp());
- }
- }
-
scoped_refptr<AudioBuffer> output;
- int decoded_frames = 0;
- int original_frames = 0;
- int channels = DetermineChannels(av_frame_.get());
+ const int channels = DetermineChannels(av_frame_.get());
if (frame_decoded) {
if (av_frame_->sample_rate != config_.samples_per_second() ||
channels != ChannelLayoutToChannelCount(config_.channel_layout()) ||
<< ", Sample Format: " << av_frame_->format << " vs "
<< av_sample_format_;
+ if (config_.codec() == kCodecAAC &&
+ av_frame_->sample_rate == 2 * config_.samples_per_second()) {
+ MEDIA_LOG(log_cb_) << "Implicit HE-AAC signalling is being used."
+ << " Please use mp4a.40.5 instead of mp4a.40.2 in"
+ << " the mimetype.";
+ }
// This is an unrecoverable error, so bail out.
queued_audio_.clear();
av_frame_unref(av_frame_.get());
DCHECK_EQ(ChannelLayoutToChannelCount(config_.channel_layout()),
output->channel_count());
- original_frames = av_frame_->nb_samples;
- int unread_frames = output->frame_count() - original_frames;
+ const int unread_frames = output->frame_count() - av_frame_->nb_samples;
DCHECK_GE(unread_frames, 0);
if (unread_frames > 0)
output->TrimEnd(unread_frames);
- // If there are frames to drop, get rid of as many as we can.
- if (output_frames_to_drop_ > 0) {
- int drop = std::min(output->frame_count(), output_frames_to_drop_);
- output->TrimStart(drop);
- output_frames_to_drop_ -= drop;
- }
-
- decoded_frames = output->frame_count();
av_frame_unref(av_frame_.get());
}
// WARNING: |av_frame_| no longer has valid data at this point.
-
- if (decoded_frames > 0) {
- // Set the timestamp/duration once all the extra frames have been
- // discarded.
- output->set_timestamp(output_timestamp_helper_->GetTimestamp());
- output->set_duration(
- output_timestamp_helper_->GetFrameDuration(decoded_frames));
- output_timestamp_helper_->AddFrames(decoded_frames);
- } else if (IsEndOfStream(result, original_frames, buffer)) {
+ const int decoded_frames = frame_decoded ? output->frame_count() : 0;
+ if (IsEndOfStream(result, decoded_frames, buffer)) {
DCHECK_EQ(packet.size, 0);
- output = AudioBuffer::CreateEOSBuffer();
- } else {
- // In case all the frames in the buffer were dropped.
- output = NULL;
- }
-
- if (output.get())
+ queued_audio_.push_back(AudioBuffer::CreateEOSBuffer());
+ } else if (discard_helper_->ProcessBuffers(buffer, output)) {
queued_audio_.push_back(output);
-
+ }
} while (packet.size > 0);
return true;
// Success!
av_frame_.reset(av_frame_alloc());
- output_timestamp_helper_.reset(
- new AudioTimestampHelper(config_.samples_per_second()));
-
+ discard_helper_.reset(new AudioDiscardHelper(config_.samples_per_second(),
+ config_.codec_delay()));
av_sample_format_ = codec_context_->sample_fmt;
if (codec_context_->channels !=
state_ = kUninitialized;
return false;
}
+
+ ResetTimestampState();
return true;
}
void FFmpegAudioDecoder::ResetTimestampState() {
- output_timestamp_helper_->SetBaseTimestamp(kNoTimestamp());
- last_input_timestamp_ = kNoTimestamp();
- output_frames_to_drop_ = 0;
+ discard_helper_->Reset(config_.codec_delay());
}
} // namespace media
projects / platform / framework / web / crosswalk.git / blobdiff