shaka-packager/packager/media/formats/mp2t/es_parser_adts.cc

277 lines
8.7 KiB
C++

// 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_adts.h"
#include <stdint.h>
#include <list>
#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/media_sample.h"
#include "media/base/timestamp.h"
#include "media/formats/mp2t/adts_header.h"
#include "media/formats/mp2t/mp2t_common.h"
#include "media/formats/mpeg/adts_constants.h"
namespace edash_packager {
namespace media {
// Return true if buf corresponds to an ADTS syncword.
// |buf| size must be at least 2.
static bool isAdtsSyncWord(const uint8_t* buf) {
return (buf[0] == 0xff) && ((buf[1] & 0xf6) == 0xf0);
}
// Look for an ADTS syncword.
// |new_pos| returns
// - either the byte position of the ADTS frame (if found)
// - or the byte position of 1st byte that was not processed (if not found).
// In every case, the returned value in |new_pos| is such that new_pos >= pos
// |frame_sz| returns the size of the ADTS frame (if found).
// Return whether a syncword was found.
static bool LookForSyncWord(const uint8_t* raw_es,
int raw_es_size,
int pos,
int* new_pos,
int* frame_sz) {
DCHECK_GE(pos, 0);
DCHECK_LE(pos, raw_es_size);
int max_offset = raw_es_size - kAdtsHeaderMinSize;
if (pos >= max_offset) {
// Do not change the position if:
// - max_offset < 0: not enough bytes to get a full header
// Since pos >= 0, this is a subcase of the next condition.
// - pos >= max_offset: might be the case after reading one full frame,
// |pos| is then incremented by the frame size and might then point
// to the end of the buffer.
*new_pos = pos;
return false;
}
for (int offset = pos; offset < max_offset; offset++) {
const uint8_t* cur_buf = &raw_es[offset];
if (!isAdtsSyncWord(cur_buf))
// The first 12 bits must be 1.
// The layer field (2 bits) must be set to 0.
continue;
int frame_size =
mp2t::AdtsHeader::GetAdtsFrameSize(cur_buf, kAdtsHeaderMinSize);
if (frame_size < kAdtsHeaderMinSize) {
// Too short to be an ADTS frame.
continue;
}
// Check whether there is another frame
// |size| apart from the current one.
int remaining_size = raw_es_size - offset;
if (remaining_size >= frame_size + 2 &&
!isAdtsSyncWord(&cur_buf[frame_size])) {
continue;
}
*new_pos = offset;
*frame_sz = frame_size;
return true;
}
*new_pos = max_offset;
return false;
}
namespace mp2t {
EsParserAdts::EsParserAdts(uint32_t pid,
const NewStreamInfoCB& new_stream_info_cb,
const EmitSampleCB& emit_sample_cb,
bool sbr_in_mimetype)
: EsParser(pid),
new_stream_info_cb_(new_stream_info_cb),
emit_sample_cb_(emit_sample_cb),
sbr_in_mimetype_(sbr_in_mimetype) {
}
EsParserAdts::~EsParserAdts() {
}
bool EsParserAdts::Parse(const uint8_t* buf,
int size,
int64_t pts,
int64_t dts) {
int raw_es_size;
const uint8_t* raw_es;
// The incoming PTS applies to the access unit that comes just after
// the beginning of |buf|.
if (pts != kNoTimestamp) {
es_byte_queue_.Peek(&raw_es, &raw_es_size);
pts_list_.push_back(EsPts(raw_es_size, pts));
}
// Copy the input data to the ES buffer.
es_byte_queue_.Push(buf, size);
es_byte_queue_.Peek(&raw_es, &raw_es_size);
// Look for every ADTS frame in the ES buffer starting at offset = 0
int es_position = 0;
int frame_size;
while (LookForSyncWord(raw_es, raw_es_size, es_position,
&es_position, &frame_size)) {
const uint8_t* frame_ptr = raw_es + es_position;
DVLOG(LOG_LEVEL_ES)
<< "ADTS syncword @ pos=" << es_position
<< " frame_size=" << frame_size;
DVLOG(LOG_LEVEL_ES)
<< "ADTS header: "
<< base::HexEncode(frame_ptr, kAdtsHeaderMinSize);
// Do not process the frame if this one is a partial frame.
int remaining_size = raw_es_size - es_position;
if (frame_size > remaining_size)
break;
size_t header_size = AdtsHeader::GetAdtsHeaderSize(frame_ptr, frame_size);
// Update the audio configuration if needed.
DCHECK_GE(frame_size, kAdtsHeaderMinSize);
if (!UpdateAudioConfiguration(frame_ptr, frame_size))
return false;
// Get the PTS & the duration of this access unit.
while (!pts_list_.empty() &&
pts_list_.front().first <= es_position) {
audio_timestamp_helper_->SetBaseTimestamp(pts_list_.front().second);
pts_list_.pop_front();
}
int64_t current_pts = audio_timestamp_helper_->GetTimestamp();
int64_t frame_duration =
audio_timestamp_helper_->GetFrameDuration(kSamplesPerAACFrame);
// Emit an audio frame.
bool is_key_frame = true;
scoped_refptr<MediaSample> sample =
MediaSample::CopyFrom(
frame_ptr + header_size,
frame_size - header_size,
is_key_frame);
sample->set_pts(current_pts);
sample->set_dts(current_pts);
sample->set_duration(frame_duration);
emit_sample_cb_.Run(pid(), sample);
// Update the PTS of the next frame.
audio_timestamp_helper_->AddFrames(kSamplesPerAACFrame);
// Skip the current frame.
es_position += frame_size;
}
// Discard all the bytes that have been processed.
DiscardEs(es_position);
return true;
}
void EsParserAdts::Flush() {
}
void EsParserAdts::Reset() {
es_byte_queue_.Reset();
pts_list_.clear();
last_audio_decoder_config_ = scoped_refptr<AudioStreamInfo>();
}
bool EsParserAdts::UpdateAudioConfiguration(const uint8_t* adts_frame,
size_t adts_frame_size) {
const uint8_t kAacSampleSizeBits(16);
AdtsHeader adts_header;
if (!adts_header.Parse(adts_frame, adts_frame_size)) {
LOG(ERROR) << "Error parsing ADTS frame header.";
return false;
}
std::vector<uint8_t> audio_specific_config;
if (!adts_header.GetAudioSpecificConfig(&audio_specific_config))
return false;
if (last_audio_decoder_config_) {
// Verify that the audio decoder config has not changed.
if (last_audio_decoder_config_->extra_data() == audio_specific_config) {
// Audio configuration has not changed.
return true;
}
NOTIMPLEMENTED() << "Varying audio configurations are not supported.";
return false;
}
// The following code is written according to ISO 14496 Part 3 Table 1.11 and
// Table 1.22. (Table 1.11 refers to the capping to 48000, Table 1.22 refers
// to SBR doubling the AAC sample rate.)
int samples_per_second = adts_header.GetSamplingFrequency();
int extended_samples_per_second = sbr_in_mimetype_
? std::min(2 * samples_per_second, 48000)
: samples_per_second;
last_audio_decoder_config_ = scoped_refptr<StreamInfo>(
new AudioStreamInfo(
pid(),
kMpeg2Timescale,
kInfiniteDuration,
kCodecAAC,
AudioStreamInfo::GetCodecString(kCodecAAC,
adts_header.GetObjectType()),
std::string(),
kAacSampleSizeBits,
adts_header.GetNumChannels(),
extended_samples_per_second,
audio_specific_config.data(),
audio_specific_config.size(),
false));
DVLOG(1) << "Sampling frequency: " << samples_per_second;
DVLOG(1) << "Extended sampling frequency: " << extended_samples_per_second;
DVLOG(1) << "Channel config: " << adts_header.GetNumChannels();
DVLOG(1) << "Object type: " << adts_header.GetObjectType();
// Reset the timestamp helper to use a new sampling frequency.
if (audio_timestamp_helper_) {
int64_t base_timestamp = audio_timestamp_helper_->GetTimestamp();
audio_timestamp_helper_.reset(
new AudioTimestampHelper(kMpeg2Timescale, samples_per_second));
audio_timestamp_helper_->SetBaseTimestamp(base_timestamp);
} else {
audio_timestamp_helper_.reset(
new AudioTimestampHelper(kMpeg2Timescale, extended_samples_per_second));
}
// Audio config notification.
new_stream_info_cb_.Run(last_audio_decoder_config_);
return true;
}
void EsParserAdts::DiscardEs(int nbytes) {
DCHECK_GE(nbytes, 0);
if (nbytes <= 0)
return;
// Adjust the ES position of each PTS.
for (EsPtsList::iterator it = pts_list_.begin(); it != pts_list_.end(); ++it)
it->first -= nbytes;
// Discard |nbytes| of ES.
es_byte_queue_.Pop(nbytes);
}
} // namespace mp2t
} // namespace media
} // namespace edash_packager