mirror of
https://gitlab.com/mbunkus/mkvtoolnix.git
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245 lines
6.9 KiB
C++
245 lines
6.9 KiB
C++
/*
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mkvmerge -- utility for splicing together matroska files
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from component media subtypes
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p_aac.h
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Written by Moritz Bunkus <moritz@bunkus.org>
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Distributed under the GPL
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see the file COPYING for details
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or visit http://www.gnu.org/copyleft/gpl.html
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*/
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/*!
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\file
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\version \$Id: p_aac.cpp,v 1.8 2003/05/26 21:49:11 mosu Exp $
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\brief AAC output module
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\author Moritz Bunkus <moritz@bunkus.org>
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*/
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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include <errno.h>
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#include "pr_generic.h"
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#include "aac_common.h"
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#include "p_aac.h"
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#include "matroska.h"
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using namespace LIBMATROSKA_NAMESPACE;
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aac_packetizer_c::aac_packetizer_c(generic_reader_c *nreader, int nid,
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int nprofile,
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unsigned long nsamples_per_sec,
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int nchannels, track_info_t *nti,
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bool nheaderless)
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throw (error_c): generic_packetizer_c(nreader, nti) {
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packetno = 0;
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bytes_output = 0;
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packet_buffer = NULL;
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buffer_size = 0;
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samples_per_sec = nsamples_per_sec;
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channels = nchannels;
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id = nid;
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profile = nprofile;
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headerless = nheaderless;
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set_track_type(track_audio);
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duplicate_data_on_add(headerless);
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}
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aac_packetizer_c::~aac_packetizer_c() {
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if (packet_buffer != NULL)
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safefree(packet_buffer);
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}
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void aac_packetizer_c::add_to_buffer(unsigned char *buf, int size) {
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unsigned char *new_buffer;
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new_buffer = (unsigned char *)saferealloc(packet_buffer, buffer_size + size);
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memcpy(new_buffer + buffer_size, buf, size);
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packet_buffer = new_buffer;
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buffer_size += size;
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}
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int aac_packetizer_c::aac_packet_available() {
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int pos;
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aac_header_t aacheader;
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if (packet_buffer == NULL)
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return 0;
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pos = find_aac_header(packet_buffer, buffer_size, &aacheader);
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if (pos < 0)
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return 0;
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return 1;
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}
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void aac_packetizer_c::remove_aac_packet(int pos, int framesize) {
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int new_size;
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unsigned char *temp_buf;
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new_size = buffer_size - (pos + framesize);
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if (new_size != 0)
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temp_buf = (unsigned char *)safememdup(&packet_buffer[pos + framesize],
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new_size);
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else
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temp_buf = NULL;
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safefree(packet_buffer);
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packet_buffer = temp_buf;
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buffer_size = new_size;
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}
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unsigned char *aac_packetizer_c::get_aac_packet(unsigned long *header,
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aac_header_t *aacheader) {
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int pos, i, up_shift, down_shift;
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unsigned char *buf, *src;
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double pims;
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if (packet_buffer == NULL)
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return 0;
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pos = find_aac_header(packet_buffer, buffer_size, aacheader);
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if (pos < 0)
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return 0;
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if ((pos + aacheader->bytes) > buffer_size)
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return 0;
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pims = ((double)aacheader->bytes) * 1000.0 /
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((double)aacheader->bit_rate / 8.0);
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if (ti->async.displacement < 0) {
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/*
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* AAC audio synchronization. displacement < 0 means skipping an
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* appropriate number of packets at the beginning.
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*/
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ti->async.displacement += (int)pims;
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if (ti->async.displacement > -(pims / 2))
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ti->async.displacement = 0;
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remove_aac_packet(pos, aacheader->bytes);
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return 0;
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}
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if (verbose && (pos > 0))
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fprintf(stdout, "aac_packetizer: skipping %d bytes (no valid AAC header "
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"found). This might make audio/video go out of sync, but this "
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"stream is damaged.\n", pos);
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if ((aacheader->header_bit_size % 8) == 0)
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buf = (unsigned char *)safememdup(packet_buffer + pos +
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aacheader->header_byte_size,
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aacheader->data_byte_size);
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else {
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// Header is not byte aligned, i.e. MPEG-4 ADTS
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// This code is from mpeg4ip/server/mp4creator/aac.cpp
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up_shift = aacheader->header_bit_size % 8;
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down_shift = 8 - up_shift;
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src = packet_buffer + pos + aacheader->header_bit_size / 8;
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buf = (unsigned char *)safemalloc(aacheader->data_byte_size);
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buf[0] = src[0] << up_shift;
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for (i = 1; i < aacheader->data_byte_size; i++) {
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buf[i - 1] |= (src[i] >> down_shift);
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buf[i] = (src[i] << up_shift);
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}
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}
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if (ti->async.displacement > 0) {
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/*
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* AAC audio synchronization. displacement > 0 is solved by duplicating
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* the very first AAC packet as often as necessary. I cannot create
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* a packet with total silence because I don't know how, and simply
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* settings the packet's values to 0 does not work as the AAC header
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* contains a CRC of its data.
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*/
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ti->async.displacement -= (int)pims;
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if (ti->async.displacement < (pims / 2))
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ti->async.displacement = 0;
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return buf;
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}
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remove_aac_packet(pos, aacheader->bytes);
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return buf;
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}
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void aac_packetizer_c::set_headers() {
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if (id == AAC_ID_MPEG4) {
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if (profile == AAC_PROFILE_MAIN)
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set_codec_id(MKV_A_AAC_4MAIN);
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else if (profile == AAC_PROFILE_LC)
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set_codec_id(MKV_A_AAC_4LC);
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else if (profile == AAC_PROFILE_SSR)
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set_codec_id(MKV_A_AAC_4SSR);
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else if (profile == AAC_PROFILE_LTP)
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set_codec_id(MKV_A_AAC_4LTP);
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else
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die("aac_packetizer: Unknown AAC MPEG-4 object type %d.", profile);
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} else {
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if (profile == AAC_PROFILE_MAIN)
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set_codec_id(MKV_A_AAC_2MAIN);
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else if (profile == AAC_PROFILE_LC)
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set_codec_id(MKV_A_AAC_2LC);
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else if (profile == AAC_PROFILE_SSR)
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set_codec_id(MKV_A_AAC_2SSR);
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else
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die("aac_packetizer: Unknown AAC MPEG-2 profile %d.", profile);
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}
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set_audio_sampling_freq((float)samples_per_sec);
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set_audio_channels(channels);
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generic_packetizer_c::set_headers();
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}
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int aac_packetizer_c::process(unsigned char *buf, int size,
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int64_t timecode, int64_t, int64_t, int64_t) {
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unsigned char *packet;
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unsigned long header;
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aac_header_t aacheader;
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int64_t my_timecode;
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debug_enter("aac_packetizer_c::process");
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if (headerless) {
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if (timecode != -1)
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my_timecode = timecode;
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else
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my_timecode = (int64_t)(1000.0 * packetno * 1024 * ti->async.linear /
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samples_per_sec);
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add_packet(buf, size, my_timecode,
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(int64_t)(1000.0 * 1024 * ti->async.linear / samples_per_sec));
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debug_leave("aac_packetizer_c::process");
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return EMOREDATA;
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}
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if (timecode != -1)
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my_timecode = timecode;
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add_to_buffer(buf, size);
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while ((packet = get_aac_packet(&header, &aacheader)) != NULL) {
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if (timecode == -1)
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my_timecode = (int64_t)(1000.0 * packetno * 1024 * ti->async.linear /
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samples_per_sec);
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add_packet(packet, aacheader.data_byte_size, my_timecode,
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(int64_t)(1000.0 * 1024 * ti->async.linear / samples_per_sec));
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packetno++;
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}
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debug_leave("aac_packetizer_c::process");
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return EMOREDATA;
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}
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void aac_packetizer_c::dump_debug_info() {
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fprintf(stderr, "DBG> aac_packetizer_c: queue: %d; buffer size: %d\n",
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packet_queue.size(), buffer_size);
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}
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