Support MPEG1 audio in the MPEG2-TS stream parser.

media/formats/mp2t/es_parser_mpeg1audio.cc

Index: media/formats/mp2t/es_parser_mpeg1audio.cc
diff --git a/media/formats/mp2t/es_parser_mpeg1audio.cc b/media/formats/mp2t/es_parser_mpeg1audio.cc
new file mode 100644
index 0000000000000000000000000000000000000000..f926cd98debca4758d955f509658654799eeee7a
--- /dev/null
+++ b/media/formats/mp2t/es_parser_mpeg1audio.cc
@@ -0,0 +1,497 @@
+// 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/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/stream_parser_buffer.h"
+#include "media/formats/common/offset_byte_queue.h"
+#include "media/formats/mp2t/mp2t_common.h"
+
+namespace media {
+namespace mp2t {
+
+// Map that determines which bitrate_index & channel_mode combinations
+// are allowed.

[wolenetz, 2014/08/26 20:26:31]

nit: describe columns and indices? (in case the referenced webpage becomes out
of sync with this)

+// Derived from: http://mpgedit.org/mpgedit/mpeg_format/MP3Format.html
+static const bool kIsAllowed[17][4] = {

[wolenetz, 2014/08/26 20:26:31]

nit: s/17/16/ ?

+  { true, true, true, true },      // free
+  { true, false, false, false },   // 32
+  { true, false, false, false },   // 48
+  { true, false, false, false },   // 56
+  { true, true, true, true },      // 64
+  { true, false, false, false },   // 80
+  { true, true, true, true },      // 96
+  { true, true, true, true },      // 112
+  { true, true, true, true },      // 128
+  { true, true, true, true },      // 160
+  { true, true, true, true },      // 192
+  { false, true, true, true },     // 224
+  { false, true, true, true },     // 256
+  { false, true, true, true },     // 320
+  { false, true, true, true },     // 384
+  { false, false, false, false }   // bad
+};
+
+// Maps version and layer information in the frame header
+// into an index for the |kBitrateMap|.
+// Derived from: http://mpgedit.org/mpgedit/mpeg_format/MP3Format.html
+static const int kVersionLayerMap[4][4] = {
+  // { reserved, L3, L2, L1 }
+  { 5, 4, 4, 3 },  // MPEG 2.5
+  { 5, 5, 5, 5 },  // reserved
+  { 5, 4, 4, 3 },  // MPEG 2
+  { 5, 2, 1, 0 }   // MPEG 1
+};
+
+// Maps the bitrate index field in the header and an index
+// from |kVersionLayerMap| to a frame bitrate.
+// Derived from: http://mpgedit.org/mpgedit/mpeg_format/MP3Format.html
+static const int kBitrateMap[16][6] = {
+  // { V1L1, V1L2, V1L3, V2L1, V2L2 & V2L3, reserved }
+  { 0, 0, 0, 0, 0, 0 },
+  { 32, 32, 32, 32, 8, 0 },
+  { 64, 48, 40, 48, 16, 0 },
+  { 96, 56, 48, 56, 24, 0 },
+  { 128, 64, 56, 64, 32, 0 },
+  { 160, 80, 64, 80, 40, 0 },
+  { 192, 96, 80, 96, 48, 0 },
+  { 224, 112, 96, 112, 56, 0 },
+  { 256, 128, 112, 128, 64, 0 },
+  { 288, 160, 128, 144, 80, 0 },
+  { 320, 192, 160, 160, 96, 0 },
+  { 352, 224, 192, 176, 112, 0 },
+  { 384, 256, 224, 192, 128, 0 },
+  { 416, 320, 256, 224, 144, 0 },
+  { 448, 384, 320, 256, 160, 0 },
+  { 0, 0, 0, 0, 0}
+};
+
+// Maps the sample rate index and version fields from the frame header
+// to a sample rate.
+// Derived from: http://mpgedit.org/mpgedit/mpeg_format/MP3Format.html
+static const int kSampleRateMap[4][4] = {
+  // { V2.5, reserved, V2, V1 }
+  { 11025, 0, 22050, 44100 },
+  { 12000, 0, 24000, 48000 },
+  { 8000, 0, 16000, 32000 },
+  { 0, 0, 0, 0 }
+};
+
+#if 0
+// Offset in bytes from the end of the MP3 header to "Xing" or "Info" tags which
+// indicate a frame is silent metadata frame.  Values taken from FFmpeg.
+static const int kXingHeaderMap[2][2] = {{32, 17}, {17, 9}};
+#endif
+
+// Frame header field constants.
+static const int kVersion2 = 2;
+static const int kVersionReserved = 1;
+static const int kVersion2_5 = 0;
+static const int kLayerReserved = 0;
+static const int kLayer1 = 3;
+static const int kLayer2 = 2;
+static const int kLayer3 = 1;
+static const int kBitrateFree = 0;
+static const int kBitrateBad = 0xf;
+static const int kSampleRateReserved = 3;
+
+int ParseMpegAudioFrameHeader(const uint8* data,
+                              int size,
+                              int* frame_size,
+                              int* sample_rate,
+                              ChannelLayout* channel_layout,
+                              int* sample_count,
+                              bool* metadata_frame) {
+  DCHECK(data);
+  DCHECK_GE(size, 0);
+  DCHECK(frame_size);
+
+  if (size < 4)
+    return 0;
+
+  BitReader reader(data, size);
+  int sync;
+  int version;
+  int layer;
+  int is_protected;
+  int bitrate_index;
+  int sample_rate_index;
+  int has_padding;
+  int is_private;
+  int channel_mode;
+  int other_flags;
+
+  if (!reader.ReadBits(11, &sync) ||
+      !reader.ReadBits(2, &version) ||
+      !reader.ReadBits(2, &layer) ||
+      !reader.ReadBits(1, &is_protected) ||
+      !reader.ReadBits(4, &bitrate_index) ||
+      !reader.ReadBits(2, &sample_rate_index) ||
+      !reader.ReadBits(1, &has_padding) ||
+      !reader.ReadBits(1, &is_private) ||
+      !reader.ReadBits(2, &channel_mode) ||
+      !reader.ReadBits(6, &other_flags)) {
+    return -1;
+  }
+
+  DVLOG(2) << "Header data :" << std::hex
+           << " sync 0x" << sync
+           << " version 0x" << version
+           << " layer 0x" << layer
+           << " bitrate_index 0x" << bitrate_index
+           << " sample_rate_index 0x" << sample_rate_index
+           << " channel_mode 0x" << channel_mode;
+
+  if (sync != 0x7ff ||
+      version == kVersionReserved ||
+      layer == kLayerReserved ||
+      bitrate_index == kBitrateFree || bitrate_index == kBitrateBad ||
+      sample_rate_index == kSampleRateReserved) {
+    return -1;
+  }
+
+  if (layer == kLayer2 && kIsAllowed[bitrate_index][channel_mode]) {
+    return -1;
+  }
+
+  int bitrate = kBitrateMap[bitrate_index][kVersionLayerMap[version][layer]];
+
+  if (bitrate == 0) {
+    return -1;
+  }
+
+  DVLOG(2) << " bitrate " << bitrate;
+
+  int frame_sample_rate = kSampleRateMap[sample_rate_index][version];
+  if (frame_sample_rate == 0) {
+    return -1;
+  }
+
+  if (sample_rate)
+    *sample_rate = frame_sample_rate;
+
+  // http://teslabs.com/openplayer/docs/docs/specs/mp3_structure2.pdf
+  // Table 2.1.5
+  int samples_per_frame;
+  switch (layer) {
+    case kLayer1:
+      samples_per_frame = 384;
+      break;
+
+    case kLayer2:
+      samples_per_frame = 1152;
+      break;
+
+    case kLayer3:
+      if (version == kVersion2 || version == kVersion2_5)
+        samples_per_frame = 576;
+      else
+        samples_per_frame = 1152;
+      break;
+
+    default:
+      return -1;
+  }
+
+  if (sample_count)
+    *sample_count = samples_per_frame;
+
+  // http://teslabs.com/openplayer/docs/docs/specs/mp3_structure2.pdf
+  // Text just below Table 2.1.5.
+  if (layer == kLayer1) {
+    // This formulation is a slight variation on the equation below,
+    // but has slightly different truncation characteristics to deal
+    // with the fact that Layer 1 has 4 byte "slots" instead of single
+    // byte ones.
+    *frame_size = 4 * (12 * bitrate * 1000 / frame_sample_rate);
+  } else {
+    *frame_size =
+        ((samples_per_frame / 8) * bitrate * 1000) / frame_sample_rate;
+  }
+
+  if (has_padding)
+    *frame_size += (layer == kLayer1) ? 4 : 1;
+
+  if (channel_layout) {
+    // Map Stereo(0), Joint Stereo(1), and Dual Channel (2) to
+    // CHANNEL_LAYOUT_STEREO and Single Channel (3) to CHANNEL_LAYOUT_MONO.
+    *channel_layout =
+        (channel_mode == 3) ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
+  }
+
+  if (metadata_frame)
+    *metadata_frame = false;
+
+  const int header_bytes_read = reader.bits_read() / 8;
+#if 1
+  return header_bytes_read;
+#else
+  if (layer != kLayer3)
+    return header_bytes_read;
+
+  // Check if this is a XING frame and tell the base parser to skip it if so.
+  const int xing_header_index =
+      kXingHeaderMap[version == kVersion2 ||
+                     version == kVersion2_5][channel_mode == 3];
+  uint32_t tag = 0;
+
+  // It's not a XING frame if the frame isn't big enough to be one.
+  if (*frame_size <
+      header_bytes_read + xing_header_index + static_cast<int>(sizeof(tag))) {
+    return header_bytes_read;
+  }
+
+  // If we don't have enough data available to check, return 0 so frame parsing
+  // will be retried once more data is available.
+  if (!reader.SkipBits(xing_header_index * 8) ||
+      !reader.ReadBits(sizeof(tag) * 8, &tag)) {
+    return 0;
+  }
+
+  // Check to see if the tag contains 'Xing' or 'Info'
+  if (tag == 0x496e666f || tag == 0x58696e67) {
+    if (metadata_frame)
+      *metadata_frame = true;
+    return reader.bits_read() / 8;
+  }
+
+  // If it wasn't a XING frame, just return the number consumed bytes.
+  return header_bytes_read;
+#endif
+}
+
+static const int kMpegAudioHeaderMinSize = 4;
+
+
+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(
+    Mpeg1AudioFrame* mpeg1audio_frame) {
+  int es_size;
+  const uint8* es;
+  es_queue_->Peek(&es, &es_size);
+
+  int max_offset = es_size - kMpegAudioHeaderMinSize;
+  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;
+
+    int frame_size;
+    int sample_rate;
+    ChannelLayout channel_layout;
+    int sample_count;
+    bool metadata_frame;
+
+    int remaining_size = es_size - offset;
+    int header_size =
+      ParseMpegAudioFrameHeader(cur_buf, remaining_size,
+                                &frame_size, &sample_rate, &channel_layout,
+                                &sample_count, &metadata_frame);
+
+    if (header_size < 0)
+      continue;
+
+    if (remaining_size < frame_size) {
+      // Not a full frame: will resume when we have more data.
+      es_queue_->Pop(offset);
+      return false;
+    }
+
+    // Check whether there is another frame
+    // |frame_size| apart from the current one.
+    if (remaining_size >= frame_size + 2 &&
+        cur_buf[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 = frame_size;
+    mpeg1audio_frame->sample_count = 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, kMpegAudioHeaderMinSize);
+    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),
+    es_queue_(new media::OffsetByteQueue()) {
+}
+
+EsParserMpeg1Audio::~EsParserMpeg1Audio() {
+}
+
+bool EsParserMpeg1Audio::Parse(
+    const uint8* buf, int size,
+    base::TimeDelta pts,
+    DecodeTimestamp dts) {
+  // The incoming PTS applies to the access unit that comes just after
+  // the beginning of |buf|.
+  if (pts != kNoTimestamp())
+    pts_list_.push_back(EsPts(es_queue_->tail(), pts));
+
+  // Copy the input data to the ES buffer.
+  es_queue_->Push(buf, size);
+
+  // 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, kMpegAudioHeaderMinSize);
+    if (!UpdateAudioConfiguration(mpeg1audio_frame.data))
+      return false;
+
+    // Get the PTS & the duration of this access unit.
+    while (!pts_list_.empty() &&
+           pts_list_.front().first <= mpeg1audio_frame.queue_offset) {
+      audio_timestamp_helper_->SetBaseTimestamp(pts_list_.front().second);
+      pts_list_.pop_front();
+    }
+
+    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::Reset() {
+  es_queue_.reset(new media::OffsetByteQueue());
+  pts_list_.clear();
+  last_audio_decoder_config_ = AudioDecoderConfig();
+}
+
+bool EsParserMpeg1Audio::UpdateAudioConfiguration(
+    const uint8* mpeg1audio_header) {
+  int frame_size;
+  int sample_rate;
+  ChannelLayout channel_layout;
+  int sample_count;
+  bool metadata_frame;
+  int header_size =
+    ParseMpegAudioFrameHeader(mpeg1audio_header, kMpegAudioHeaderMinSize,
+                              &frame_size, &sample_rate, &channel_layout,
+                              &sample_count, &metadata_frame);
+  if (header_size < 0)
+    return false;
+
+#if 0
+  // The following code is written according to ISO 14496 Part 3 Table 1.13 -
+  // Syntax of AudioSpecificConfig.
+  uint16 extra_data_int =
+      // Note: adts_profile is in the range [0,3], since the ADTS header only
+      // allows two bits for its value.
+      ((adts_profile + 1) << 11) +
+      (frequency_index << 7) +
+      (channel_configuration << 3);
+  uint8 extra_data[2] = {
+      static_cast<uint8>(extra_data_int >> 8),
+      static_cast<uint8>(extra_data_int & 0xff)
+  };
+#endif
+
+  AudioDecoderConfig audio_decoder_config(
+      kCodecMP3,
+      kSampleFormatS16,
+      channel_layout,
+      sample_rate,
+      NULL, 0,
+      false);
+
+  if (!audio_decoder_config.Matches(last_audio_decoder_config_)) {
+    DVLOG(1) << "Sampling frequency: " << 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(sample_rate));
+      audio_timestamp_helper_->SetBaseTimestamp(base_timestamp);
+    } else {
+      audio_timestamp_helper_.reset(
+          new AudioTimestampHelper(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