// Copyright 2014 The Chromium Authors
// 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 <stddef.h>
#include <optional>
#include <vector>
#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/channel_layout.h"
#include "media/base/encryption_pattern.h"
#include "media/base/media_util.h"
#include "media/base/stream_parser_buffer.h"
#include "media/base/timestamp_constants.h"
#include "media/formats/common/offset_byte_queue.h"
#include "media/formats/mp2t/mp2t_common.h"
#include "media/formats/mpeg/adts_constants.h"
namespace media {
static int ExtractAdtsFrameSize(const uint8_t* adts_header) {
return ((static_cast<int>(adts_header[5]) >> 5) |
(static_cast<int>(adts_header[4]) << 3) |
((static_cast<int>(adts_header[3]) & 0x3) << 11));
}
static int AdtsHeaderSize(const uint8_t* adts_header) {
// protection absent bit: set to 1 if there is no CRC and 0 if there is CRC
return (adts_header[1] & 0x1) ? kADTSHeaderSizeNoCrc : kADTSHeaderSizeWithCrc;
}
// Return true if buf corresponds to an ADTS syncword.
// |buf| size must be at least 2.
static bool isAdtsSyncWord(const uint8_t* buf) {
// The first 12 bits must be 1.
// The layer field (2 bits) must be set to 0.
return (buf[0] == 0xff) && ((buf[1] & 0xf6) == 0xf0);
}
namespace mp2t {
struct EsParserAdts::AdtsFrame {
// Pointer to the ES data.
const uint8_t* data;
// Frame size;
int size;
int header_size;
// Frame offset in the ES queue.
int64_t queue_offset;
};
bool EsParserAdts::LookForAdtsFrame(AdtsFrame* adts_frame) {
int es_size;
const uint8_t* es;
es_queue_->Peek(&es, &es_size);
int max_offset = es_size - kADTSHeaderMinSize;
if (max_offset <= 0)
return false;
for (int offset = 0; offset < max_offset; offset++) {
const uint8_t* cur_buf = &es[offset];
if (!isAdtsSyncWord(cur_buf))
continue;
int frame_size = ExtractAdtsFrameSize(cur_buf);
if (frame_size < kADTSHeaderMinSize) {
// Too short to be an ADTS frame.
continue;
}
int header_size = AdtsHeaderSize(cur_buf);
int remaining_size = es_size - offset;
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
// |size| apart from the current one.
if (remaining_size >= frame_size + 2 &&
!isAdtsSyncWord(&cur_buf[frame_size])) {
continue;
}
es_queue_->Pop(offset);
es_queue_->Peek(&adts_frame->data, &es_size);
adts_frame->queue_offset = es_queue_->head();
adts_frame->size = frame_size;
adts_frame->header_size = header_size;
DVLOG(LOG_LEVEL_ES)
<< "ADTS syncword @ pos=" << adts_frame->queue_offset
<< " frame_size=" << adts_frame->size;
DVLOG(LOG_LEVEL_ES)
<< "ADTS header: "
<< base::HexEncode(adts_frame->data, kADTSHeaderMinSize);
return true;
}
es_queue_->Pop(max_offset);
return false;
}
void EsParserAdts::SkipAdtsFrame(const AdtsFrame& adts_frame) {
DCHECK_EQ(adts_frame.queue_offset, es_queue_->head());
es_queue_->Pop(adts_frame.size);
}
EsParserAdts::EsParserAdts(NewAudioConfigCB new_audio_config_cb,
EmitBufferCB emit_buffer_cb,
bool sbr_in_mimetype)
: new_audio_config_cb_(std::move(new_audio_config_cb)),
emit_buffer_cb_(std::move(emit_buffer_cb)),
get_decrypt_config_cb_(),
init_encryption_scheme_(EncryptionScheme::kUnencrypted),
sbr_in_mimetype_(sbr_in_mimetype) {}
EsParserAdts::EsParserAdts(NewAudioConfigCB new_audio_config_cb,
EmitBufferCB emit_buffer_cb,
GetDecryptConfigCB get_decrypt_config_cb,
EncryptionScheme init_encryption_scheme,
bool sbr_in_mimetype)
: new_audio_config_cb_(std::move(new_audio_config_cb)),
emit_buffer_cb_(std::move(emit_buffer_cb)),
get_decrypt_config_cb_(std::move(get_decrypt_config_cb)),
init_encryption_scheme_(init_encryption_scheme),
sbr_in_mimetype_(sbr_in_mimetype) {}
EsParserAdts::~EsParserAdts() = default;
void EsParserAdts::CalculateSubsamplesForAdtsFrame(
const AdtsFrame& adts_frame,
std::vector<SubsampleEntry>* subsamples) {
DCHECK(subsamples);
subsamples->clear();
int data_size = adts_frame.size - adts_frame.header_size;
int residue = data_size % 16;
int clear_bytes = adts_frame.header_size;
int encrypted_bytes = 0;
if (data_size <= 16) {
clear_bytes += data_size;
residue = 0;
} else {
clear_bytes += 16;
encrypted_bytes = adts_frame.size - clear_bytes - residue;
}
SubsampleEntry subsample(clear_bytes, encrypted_bytes);
subsamples->push_back(subsample);
if (residue) {
subsample.clear_bytes = residue;
subsample.cypher_bytes = 0;
subsamples->push_back(subsample);
}
}
bool EsParserAdts::ParseFromEsQueue() {
// Look for every ADTS frame in the ES buffer.
AdtsFrame adts_frame;
while (LookForAdtsFrame(&adts_frame)) {
// Update the audio configuration if needed.
DCHECK_GE(adts_frame.size, kADTSHeaderMinSize);
if (!UpdateAudioConfiguration(adts_frame.data, adts_frame.size))
return false;
// Get the PTS & the duration of this access unit.
TimingDesc current_timing_desc =
GetTimingDescriptor(adts_frame.queue_offset);
if (current_timing_desc.pts != kNoTimestamp)
audio_timestamp_helper_->SetBaseTimestamp(current_timing_desc.pts);
if (!audio_timestamp_helper_->base_timestamp()) {
DVLOG(1) << "Skipping audio frame with unknown timestamp";
SkipAdtsFrame(adts_frame);
continue;
}
base::TimeDelta current_pts = audio_timestamp_helper_->GetTimestamp();
base::TimeDelta frame_duration =
audio_timestamp_helper_->GetFrameDuration(kSamplesPerAACFrame);
// 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(adts_frame.data, adts_frame.size,
is_key_frame, DemuxerStream::AUDIO,
kMp2tAudioTrackId);
stream_parser_buffer->set_timestamp(current_pts);
stream_parser_buffer->SetDecodeTimestamp(
DecodeTimestamp::FromPresentationTime(current_pts));
stream_parser_buffer->set_duration(frame_duration);
if (get_decrypt_config_cb_) {
const DecryptConfig* base_decrypt_config = get_decrypt_config_cb_.Run();
if (base_decrypt_config) {
std::vector<SubsampleEntry> subsamples;
CalculateSubsamplesForAdtsFrame(adts_frame, &subsamples);
stream_parser_buffer->set_decrypt_config(
std::make_unique<DecryptConfig>(
base_decrypt_config->encryption_scheme(),
base_decrypt_config->key_id(), base_decrypt_config->iv(),
subsamples, EncryptionPattern()));
}
}
emit_buffer_cb_.Run(stream_parser_buffer);
// Update the PTS of the next frame.
audio_timestamp_helper_->AddFrames(kSamplesPerAACFrame);
// Skip the current frame.
SkipAdtsFrame(adts_frame);
}
return true;
}
void EsParserAdts::Flush() {
}
void EsParserAdts::ResetInternal() {
last_audio_decoder_config_ = AudioDecoderConfig();
}
bool EsParserAdts::UpdateAudioConfiguration(const uint8_t* adts_header,
int size) {
int orig_sample_rate;
ChannelLayout channel_layout;
std::vector<uint8_t> extra_data;
if (adts_parser_.ParseFrameHeader(adts_header, size, nullptr,
&orig_sample_rate, &channel_layout, nullptr,
nullptr, &extra_data) <= 0) {
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.)
// TODO(damienv) : Extend sample rate cap to 96kHz for Level 5 content.
const int extended_samples_per_second =
sbr_in_mimetype_ ? std::min(2 * orig_sample_rate, 48000)
: orig_sample_rate;
AudioDecoderConfig audio_decoder_config(
AudioCodec::kAAC, kSampleFormatS16, channel_layout,
extended_samples_per_second, extra_data, init_encryption_scheme_);
if (!audio_decoder_config.IsValidConfig()) {
DVLOG(1) << "Invalid config: "
<< audio_decoder_config.AsHumanReadableString();
return false;
}
if (!audio_decoder_config.Matches(last_audio_decoder_config_)) {
DVLOG(1) << "Sampling frequency: "
<< audio_decoder_config.samples_per_second()
<< " SBR=" << sbr_in_mimetype_;
DVLOG(1) << "Channel layout: "
<< ChannelLayoutToString(audio_decoder_config.channel_layout());
// For AAC audio with SBR (Spectral Band Replication) the sampling rate is
// doubled above, but AudioTimestampHelper should still use the original
// sample rate to compute audio timestamps and durations correctly.
// Reset the timestamp helper to use a new time scale.
if (audio_timestamp_helper_ && audio_timestamp_helper_->base_timestamp()) {
base::TimeDelta base_timestamp = audio_timestamp_helper_->GetTimestamp();
audio_timestamp_helper_.reset(new AudioTimestampHelper(orig_sample_rate));
audio_timestamp_helper_->SetBaseTimestamp(base_timestamp);
} else {
audio_timestamp_helper_.reset(new AudioTimestampHelper(orig_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
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