Support MPEG1 audio in the MPEG2-TS stream parser.

media/formats/mp2t/es_parser_mpeg1audio.cc

+// Copyright 2014 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/formats/mp2t/es_parser_mpeg1audio.h"
+
+#include <list>
+
+#include "base/basictypes.h"
+#include "base/bind.h"
+#include "base/logging.h"
+#include "base/strings/string_number_conversions.h"
+#include "media/base/audio_timestamp_helper.h"
+#include "media/base/bit_reader.h"
+#include "media/base/buffers.h"
+#include "media/base/channel_layout.h"
+#include "media/base/media_log.h"
+#include "media/base/stream_parser_buffer.h"
+#include "media/formats/common/offset_byte_queue.h"
+#include "media/formats/mp2t/mp2t_common.h"
+#include "media/formats/mpeg/mpeg1_audio_stream_parser.h"
+
+namespace media {
+namespace mp2t {
+
+static void DummyMediaLog(const std::string& s) {
+}
+
+struct EsParserMpeg1Audio::Mpeg1AudioFrame {
+  // Pointer to the ES data.
+  const uint8* data;
+
+  // Frame size.
+  int size;
+
+  // Number of samples in the frame.
+  int sample_count;
+
+  // Frame offset in the ES queue.
+  int64 queue_offset;
+};
+
+bool EsParserMpeg1Audio::LookForMpeg1AudioFrame(

[acolwell GONE FROM CHROMIUM, 2014/09/04 22:20:53]

nit: Move this and the skip method to a later point in the file so they roughly
match the decl order.

[damienv1, 2014/09/05 20:06:13]

Done.

+    Mpeg1AudioFrame* mpeg1audio_frame) {
+  int es_size;
+  const uint8* es;
+  es_queue_->Peek(&es, &es_size);
+
+  int max_offset = es_size - MPEG1AudioStreamParser::kHeaderSize;
+  if (max_offset <= 0)
+    return false;
+
+  for (int offset = 0; offset < max_offset; offset++) {
+    const uint8* cur_buf = &es[offset];
+    if (cur_buf[0] != 0xff)
+      continue;
+
+    // TODO(damienv): properly supports media logs.
+    LogCB log_cb = base::Bind(&DummyMediaLog);
+    int remaining_size = es_size - offset;
+    DCHECK_GE(remaining_size, MPEG1AudioStreamParser::kHeaderSize);
+    MPEG1AudioStreamParser::Header header;
+    if (!MPEG1AudioStreamParser::ParseHeader(log_cb, cur_buf, &header))
+      continue;
+
+    if (remaining_size < header.frame_size) {
+      // Not a full frame: will resume when we have more data.
+      es_queue_->Pop(offset);

[acolwell GONE FROM CHROMIUM, 2014/09/04 22:20:53]

nit: You might want to add some text here indicating that the Pop() is dropping
garbage data that is appearing between frames.

[damienv1, 2014/09/05 20:06:13]

Done.

+      return false;
+    }
+
+    // Check whether there is another frame
+    // |frame_size| apart from the current one.
+    if (remaining_size >= header.frame_size + 2 &&
+        cur_buf[header.frame_size] != 0xff) {
+      continue;
+    }
+
+    es_queue_->Pop(offset);
+    es_queue_->Peek(&mpeg1audio_frame->data, &es_size);
+    mpeg1audio_frame->queue_offset = es_queue_->head();
+    mpeg1audio_frame->size = header.frame_size;
+    mpeg1audio_frame->sample_count = header.sample_count;
+    DVLOG(LOG_LEVEL_ES)
+        << "MPEG1 audio syncword @ pos=" << mpeg1audio_frame->queue_offset
+        << " frame_size=" << mpeg1audio_frame->size;
+    DVLOG(LOG_LEVEL_ES)
+        << "MPEG1 audio header: "
+        << base::HexEncode(mpeg1audio_frame->data,
+                           MPEG1AudioStreamParser::kHeaderSize);
+    return true;
+  }
+
+  es_queue_->Pop(max_offset);
+  return false;
+}
+
+void EsParserMpeg1Audio::SkipMpeg1AudioFrame(
+    const Mpeg1AudioFrame& mpeg1audio_frame) {
+  DCHECK_EQ(mpeg1audio_frame.queue_offset, es_queue_->head());
+  es_queue_->Pop(mpeg1audio_frame.size);
+}
+
+EsParserMpeg1Audio::EsParserMpeg1Audio(
+    const NewAudioConfigCB& new_audio_config_cb,
+    const EmitBufferCB& emit_buffer_cb)
+  : new_audio_config_cb_(new_audio_config_cb),
+    emit_buffer_cb_(emit_buffer_cb) {
+}
+
+EsParserMpeg1Audio::~EsParserMpeg1Audio() {
+}
+
+bool EsParserMpeg1Audio::ParseFromEsQueue() {
+  // Look for every MPEG1 audio frame in the ES buffer.
+  Mpeg1AudioFrame mpeg1audio_frame;
+  while (LookForMpeg1AudioFrame(&mpeg1audio_frame)) {
+    // Update the audio configuration if needed.
+    DCHECK_GE(mpeg1audio_frame.size, MPEG1AudioStreamParser::kHeaderSize);
+    if (!UpdateAudioConfiguration(mpeg1audio_frame.data))
+      return false;
+
+    // Get the PTS & the duration of this access unit.
+    TimingDesc current_timing_desc =
+        GetTimingDescriptor(mpeg1audio_frame.queue_offset);
+    if (current_timing_desc.pts != kNoTimestamp())
+      audio_timestamp_helper_->SetBaseTimestamp(current_timing_desc.pts);

[acolwell GONE FROM CHROMIUM, 2014/09/04 22:20:53]

It seems like this could introduce drift. Shouldn't this be in the block below
so that we only set the base on the first frame we know the timestamp for. It
seems like only if there is a large gap between pts and what the
audio_timestamps_helper says should we rebase the helper.

[damienv1, 2014/09/05 20:06:13]

I think the parser cannot do any assumption about the way data was muxed.
The parser does its best when timestamps are not available. However, when
timestamps
are available in the stream, they should be trusted.

+
+    if (audio_timestamp_helper_->base_timestamp() == kNoTimestamp()) {
+      DVLOG(1) << "Audio frame with unknown timestamp";
+      return false;
+    }
+    base::TimeDelta current_pts = audio_timestamp_helper_->GetTimestamp();
+    base::TimeDelta frame_duration =
+        audio_timestamp_helper_->GetFrameDuration(
+            mpeg1audio_frame.sample_count);
+
+    // Emit an audio frame.
+    bool is_key_frame = true;
+
+    // TODO(wolenetz/acolwell): Validate and use a common cross-parser TrackId
+    // type and allow multiple audio tracks. See https://crbug.com/341581.
+    scoped_refptr<StreamParserBuffer> stream_parser_buffer =
+        StreamParserBuffer::CopyFrom(
+            mpeg1audio_frame.data,
+            mpeg1audio_frame.size,
+            is_key_frame,
+            DemuxerStream::AUDIO, 0);
+    stream_parser_buffer->set_timestamp(current_pts);
+    stream_parser_buffer->set_duration(frame_duration);
+    emit_buffer_cb_.Run(stream_parser_buffer);
+
+    // Update the PTS of the next frame.
+    audio_timestamp_helper_->AddFrames(mpeg1audio_frame.sample_count);
+
+    // Skip the current frame.
+    SkipMpeg1AudioFrame(mpeg1audio_frame);
+  }
+
+  return true;
+}
+
+void EsParserMpeg1Audio::Flush() {
+}
+
+void EsParserMpeg1Audio::ResetInternal() {
+  last_audio_decoder_config_ = AudioDecoderConfig();
+}
+
+bool EsParserMpeg1Audio::UpdateAudioConfiguration(
+    const uint8* mpeg1audio_header) {
+  // TODO(damienv): properly supports media logs.
+  LogCB log_cb = base::Bind(&DummyMediaLog);

[acolwell GONE FROM CHROMIUM, 2014/09/04 22:20:53]

nit: You should probably just make this a member variable and only bind in the
constructor since that is likely what the code would need to look like if you
had the logging in place.

[damienv1, 2014/09/05 20:06:13]

Seems to be a good idea :)

+  MPEG1AudioStreamParser::Header header;
+  if (!MPEG1AudioStreamParser::ParseHeader(log_cb,
+                                           mpeg1audio_header,
+                                           &header)) {
+    return false;
+  }
+
+  // TODO(damienv): Verify whether Android playback requires the extra data
+  // field for Mpeg1 audio. If yes, we should generate this field.
+  AudioDecoderConfig audio_decoder_config(
+      kCodecMP3,
+      kSampleFormatS16,
+      header.channel_layout,
+      header.sample_rate,
+      NULL, 0,
+      false);
+
+  if (!audio_decoder_config.Matches(last_audio_decoder_config_)) {
+    DVLOG(1) << "Sampling frequency: " << header.sample_rate;
+    // Reset the timestamp helper to use a new time scale.
+    if (audio_timestamp_helper_ &&
+        audio_timestamp_helper_->base_timestamp() != kNoTimestamp()) {
+      base::TimeDelta base_timestamp = audio_timestamp_helper_->GetTimestamp();
+      audio_timestamp_helper_.reset(
+        new AudioTimestampHelper(header.sample_rate));
+      audio_timestamp_helper_->SetBaseTimestamp(base_timestamp);
+    } else {
+      audio_timestamp_helper_.reset(
+          new AudioTimestampHelper(header.sample_rate));
+    }
+    // Audio config notification.
+    last_audio_decoder_config_ = audio_decoder_config;
+    new_audio_config_cb_.Run(audio_decoder_config);
+  }
+
+  return true;
+}
+
+}  // namespace mp2t
+}  // namespace media