mkvtoolnix/p_vorbis.cpp

206 lines
6.1 KiB
C++

/*
mkvmerge -- utility for splicing together matroska files
from component media subtypes
p_vorbis.h
Written by Moritz Bunkus <moritz@bunkus.org>
Distributed under the GPL
see the file COPYING for details
or visit http://www.gnu.org/copyleft/gpl.html
*/
/*!
\file
\version $Id$
\brief Vorbis packetizer
\author Moritz Bunkus <moritz@bunkus.org>
*/
#include "config.h"
#ifdef HAVE_OGGVORBIS
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#if !defined(COMP_MSC)
#include <unistd.h>
#endif
#include <ogg/ogg.h>
#include <vorbis/codec.h>
#include "common.h"
#include "pr_generic.h"
#include "p_vorbis.h"
#include "matroska.h"
using namespace LIBMATROSKA_NAMESPACE;
vorbis_packetizer_c::vorbis_packetizer_c(generic_reader_c *nreader,
unsigned char *d_header, int l_header,
unsigned char *d_comments,
int l_comments,
unsigned char *d_codecsetup,
int l_codecsetup, track_info_t *nti)
throw (error_c): generic_packetizer_c(nreader, nti) {
int i;
packetno = 0;
last_bs = 0;
samples = 0;
memset(headers, 0, 3 * sizeof(ogg_packet));
headers[0].packet = (unsigned char *)safememdup(d_header, l_header);
headers[1].packet = (unsigned char *)safememdup(d_comments, l_comments);
headers[2].packet = (unsigned char *)safememdup(d_codecsetup, l_codecsetup);
headers[0].bytes = l_header;
headers[1].bytes = l_comments;
headers[2].bytes = l_codecsetup;
headers[0].b_o_s = 1;
headers[1].packetno = 1;
headers[2].packetno = 2;
vorbis_info_init(&vi);
vorbis_comment_init(&vc);
for (i = 0; i < 3; i++)
if (vorbis_synthesis_headerin(&vi, &vc, &headers[i]) < 0)
throw error_c("Error: vorbis_packetizer: Could not extract the "
"stream's parameters from the first packets.\n");
set_track_type(track_audio);
}
vorbis_packetizer_c::~vorbis_packetizer_c() {
int i;
for (i = 0; i < 3; i++)
if (headers[i].packet != NULL)
safefree(headers[i].packet);
}
void vorbis_packetizer_c::set_headers() {
unsigned char *buffer;
int n, offset, i, lsize;
set_codec_id(MKV_A_VORBIS);
// We use lacing for the blocks. The first bytes is the number of
// packets being laced which is one less than the number of blocks that
// are actually stored. For each packet in the lace there's the length
// coded like this:
// length = 0
// while (next byte == 255) { length += 255 }
// length += this byte which is < 255
// The last packet's length can be calculated by the length of
// the KaxCodecPrivate and all prior packets, so there's no length for it -
// and that's why the first byte is (num_packets - 1).
lsize = 1 + (headers[0].bytes / 255) + 1 + (headers[1].bytes / 255) + 1 +
headers[0].bytes + headers[1].bytes + headers[2].bytes;
buffer = (unsigned char *)safemalloc(lsize);
buffer[0] = 2; // The number of packets less one.
offset = 1;
for (i = 0; i < 2; i++) {
for (n = headers[i].bytes; n >= 255; n -= 255) {
buffer[offset] = 255;
offset++;
}
buffer[offset] = n;
offset++;
}
for (i = 0; i <= 2; i++) {
memcpy(&buffer[offset], headers[i].packet, headers[i].bytes);
offset += headers[i].bytes;
}
set_codec_private(buffer, lsize);
safefree(buffer);
set_audio_sampling_freq((float)vi.rate);
set_audio_channels(vi.channels);
generic_packetizer_c::set_headers();
}
/*
* Some notes - processing is straight-forward if no AV synchronization
* is needed - the packet is simply stored in the Matroska file.
* Unfortunately things are not that easy if AV sync is done. For a
* negative displacement packets are simply discarded if their timecode
* is set before the displacement. For positive displacements the packetizer
* has to generate silence packets and put them into the Matroska file first.
*/
int vorbis_packetizer_c::process(unsigned char *data, int size,
int64_t timecode, int64_t, int64_t, int64_t) {
unsigned char zero[2];
ogg_packet op;
int64_t this_bs, samples_here, samples_needed;
debug_enter("vorbis_packetizer_c::process");
// Recalculate the timecode if needed.
if (timecode == -1)
timecode = samples * 1000 / vi.rate;
// Positive displacement, first packet? Well then lets create silence.
if ((packetno == 0) && (ti->async.displacement > 0)) {
// Create a fake packet so we can use vorbis_packet_blocksize().
zero[0] = 0;
zero[1] = 0;
memset(&op, 0, sizeof(ogg_packet));
op.packet = zero;
op.bytes = 2;
// Calculate how many samples we have to create.
samples_needed = vi.rate * 1000 / ti->async.displacement;
this_bs = vorbis_packet_blocksize(&vi, &op);
samples_here = (this_bs + last_bs) / 4;
while ((samples + samples_here) < samples_needed) {
samples += samples_here;
last_bs = this_bs;
samples_here = (this_bs + last_bs) / 4;
add_packet(zero, 2, samples * 1000 / vi.rate, samples_here * 1000 /
vi.rate);
}
ti->async.displacement = 0;
}
// Update the number of samples we have processed so that we can
// calculate the timecode on the next call.
op.packet = data;
op.bytes = size;
this_bs = vorbis_packet_blocksize(&vi, &op);
samples_here = (this_bs + last_bs) / 4;
samples += samples_here;
last_bs = this_bs;
// Handle the displacement.
timecode += ti->async.displacement;
// Handle the linear sync - simply multiply with the given factor.
timecode = (int64_t)((double)timecode * ti->async.linear);
// If a negative sync value was used we may have to skip this packet.
if (timecode < 0) {
debug_leave("vorbis_packetizer_c::process");
return EMOREDATA;
}
add_packet(data, size, (int64_t)timecode, samples_here * 1000 / vi.rate);
debug_leave("vorbis_packetizer_c::process");
return EMOREDATA;
}
void vorbis_packetizer_c::dump_debug_info() {
fprintf(stderr, "DBG> vorbis_packetizer_c: queue: %d\n",
packet_queue.size());
}
#endif // HAVE_OGGVORBIS