// 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 "packager/media/formats/mp2t/es_parser_adts.h" #include #include #include #include "packager/base/logging.h" #include "packager/base/strings/string_number_conversions.h" #include "packager/media/base/audio_timestamp_helper.h" #include "packager/media/base/bit_reader.h" #include "packager/media/base/media_sample.h" #include "packager/media/base/timestamp.h" #include "packager/media/formats/mp2t/adts_header.h" #include "packager/media/formats/mp2t/mp2t_common.h" #include "packager/media/formats/mpeg/adts_constants.h" namespace shaka { 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 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(); } 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 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_->codec_config() == 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(new AudioStreamInfo( pid(), kMpeg2Timescale, kInfiniteDuration, kCodecAAC, AudioStreamInfo::GetCodecString(kCodecAAC, adts_header.GetObjectType()), audio_specific_config.data(), audio_specific_config.size(), kAacSampleSizeBits, adts_header.GetNumChannels(), extended_samples_per_second, 0 /* seek preroll */, 0 /* codec delay */, 0 /* max bitrate */, 0 /* avg bitrate */, std::string(), 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 shaka