ffmpeg/libavcodec/oggvorbis.c

384 lines
12 KiB
C

/*
* copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file oggvorbis.c
* Ogg Vorbis codec support via libvorbisenc.
* @author Mark Hills <mark@pogo.org.uk>
*/
#include <vorbis/vorbisenc.h>
#include "avcodec.h"
#undef NDEBUG
#include <assert.h>
#define OGGVORBIS_FRAME_SIZE 64
#define BUFFER_SIZE (1024*64)
typedef struct OggVorbisContext {
vorbis_info vi ;
vorbis_dsp_state vd ;
vorbis_block vb ;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
/* decoder */
vorbis_comment vc ;
ogg_packet op;
} OggVorbisContext ;
static int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
double cfreq;
if(avccontext->flags & CODEC_FLAG_QSCALE) {
/* variable bitrate */
if(vorbis_encode_setup_vbr(vi, avccontext->channels,
avccontext->sample_rate,
avccontext->global_quality / (float)FF_QP2LAMBDA))
return -1;
} else {
/* constant bitrate */
if(vorbis_encode_setup_managed(vi, avccontext->channels,
avccontext->sample_rate, -1, avccontext->bit_rate, -1))
return -1;
#ifdef OGGVORBIS_VBR_BY_ESTIMATE
/* variable bitrate by estimate */
if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE_AVG, NULL))
return -1;
#endif
}
/* cutoff frequency */
if(avccontext->cutoff > 0) {
cfreq = avccontext->cutoff / 1000.0;
if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
return -1;
}
return vorbis_encode_setup_init(vi);
}
static int oggvorbis_encode_init(AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
ogg_packet header, header_comm, header_code;
uint8_t *p;
unsigned int offset, len;
vorbis_info_init(&context->vi) ;
if(oggvorbis_init_encoder(&context->vi, avccontext) < 0) {
av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed") ;
return -1 ;
}
vorbis_analysis_init(&context->vd, &context->vi) ;
vorbis_block_init(&context->vd, &context->vb) ;
vorbis_comment_init(&context->vc);
vorbis_comment_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT) ;
vorbis_analysis_headerout(&context->vd, &context->vc, &header,
&header_comm, &header_code);
len = header.bytes + header_comm.bytes + header_code.bytes;
avccontext->extradata_size= 64 + len + len/255;
p = avccontext->extradata= av_mallocz(avccontext->extradata_size);
p[0] = 2;
offset = 1;
offset += av_xiphlacing(&p[offset], header.bytes);
offset += av_xiphlacing(&p[offset], header_comm.bytes);
memcpy(&p[offset], header.packet, header.bytes);
offset += header.bytes;
memcpy(&p[offset], header_comm.packet, header_comm.bytes);
offset += header_comm.bytes;
memcpy(&p[offset], header_code.packet, header_code.bytes);
offset += header_code.bytes;
avccontext->extradata_size = offset;
avccontext->extradata= av_realloc(avccontext->extradata, avccontext->extradata_size);
/* vorbis_block_clear(&context->vb);
vorbis_dsp_clear(&context->vd);
vorbis_info_clear(&context->vi);*/
vorbis_comment_clear(&context->vc);
avccontext->frame_size = OGGVORBIS_FRAME_SIZE ;
avccontext->coded_frame= avcodec_alloc_frame();
avccontext->coded_frame->key_frame= 1;
return 0 ;
}
static int oggvorbis_encode_frame(AVCodecContext *avccontext,
unsigned char *packets,
int buf_size, void *data)
{
OggVorbisContext *context = avccontext->priv_data ;
float **buffer ;
ogg_packet op ;
signed short *audio = data ;
int l, samples = data ? OGGVORBIS_FRAME_SIZE : 0;
buffer = vorbis_analysis_buffer(&context->vd, samples) ;
if(context->vi.channels == 1) {
for(l = 0 ; l < samples ; l++)
buffer[0][l]=audio[l]/32768.f;
} else {
for(l = 0 ; l < samples ; l++){
buffer[0][l]=audio[l*2]/32768.f;
buffer[1][l]=audio[l*2+1]/32768.f;
}
}
vorbis_analysis_wrote(&context->vd, samples) ;
while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
vorbis_analysis(&context->vb, NULL);
vorbis_bitrate_addblock(&context->vb) ;
while(vorbis_bitrate_flushpacket(&context->vd, &op)) {
/* i'd love to say the following line is a hack, but sadly it's
* not, apparently the end of stream decision is in libogg. */
if(op.bytes==1)
continue;
memcpy(context->buffer + context->buffer_index, &op, sizeof(ogg_packet));
context->buffer_index += sizeof(ogg_packet);
memcpy(context->buffer + context->buffer_index, op.packet, op.bytes);
context->buffer_index += op.bytes;
// av_log(avccontext, AV_LOG_DEBUG, "e%d / %d\n", context->buffer_index, op.bytes);
}
}
l=0;
if(context->buffer_index){
ogg_packet *op2= (ogg_packet*)context->buffer;
op2->packet = context->buffer + sizeof(ogg_packet);
l= op2->bytes;
avccontext->coded_frame->pts= av_rescale_q(op2->granulepos, (AVRational){1, avccontext->sample_rate}, avccontext->time_base);
//FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
memcpy(packets, op2->packet, l);
context->buffer_index -= l + sizeof(ogg_packet);
memcpy(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
// av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);
}
return l;
}
static int oggvorbis_encode_close(AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
/* ogg_packet op ; */
vorbis_analysis_wrote(&context->vd, 0) ; /* notify vorbisenc this is EOF */
vorbis_block_clear(&context->vb);
vorbis_dsp_clear(&context->vd);
vorbis_info_clear(&context->vi);
av_freep(&avccontext->coded_frame);
av_freep(&avccontext->extradata);
return 0 ;
}
AVCodec oggvorbis_encoder = {
"vorbis",
CODEC_TYPE_AUDIO,
CODEC_ID_VORBIS,
sizeof(OggVorbisContext),
oggvorbis_encode_init,
oggvorbis_encode_frame,
oggvorbis_encode_close,
.capabilities= CODEC_CAP_DELAY,
} ;
static int oggvorbis_decode_init(AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
uint8_t *p= avccontext->extradata;
int i, hsizes[3];
unsigned char *headers[3], *extradata = avccontext->extradata;
vorbis_info_init(&context->vi) ;
vorbis_comment_init(&context->vc) ;
if(! avccontext->extradata_size || ! p) {
av_log(avccontext, AV_LOG_ERROR, "vorbis extradata absent\n");
return -1;
}
if(p[0] == 0 && p[1] == 30) {
for(i = 0; i < 3; i++){
hsizes[i] = *p++ << 8;
hsizes[i] += *p++;
headers[i] = p;
p += hsizes[i];
}
} else if(*p == 2) {
unsigned int offset = 1;
p++;
for(i=0; i<2; i++) {
hsizes[i] = 0;
while((*p == 0xFF) && (offset < avccontext->extradata_size)) {
hsizes[i] += 0xFF;
offset++;
p++;
}
if(offset >= avccontext->extradata_size - 1) {
av_log(avccontext, AV_LOG_ERROR,
"vorbis header sizes damaged\n");
return -1;
}
hsizes[i] += *p;
offset++;
p++;
}
hsizes[2] = avccontext->extradata_size - hsizes[0]-hsizes[1]-offset;
#if 0
av_log(avccontext, AV_LOG_DEBUG,
"vorbis header sizes: %d, %d, %d, / extradata_len is %d \n",
hsizes[0], hsizes[1], hsizes[2], avccontext->extradata_size);
#endif
headers[0] = extradata + offset;
headers[1] = extradata + offset + hsizes[0];
headers[2] = extradata + offset + hsizes[0] + hsizes[1];
} else {
av_log(avccontext, AV_LOG_ERROR,
"vorbis initial header len is wrong: %d\n", *p);
return -1;
}
for(i=0; i<3; i++){
context->op.b_o_s= i==0;
context->op.bytes = hsizes[i];
context->op.packet = headers[i];
if(vorbis_synthesis_headerin(&context->vi, &context->vc, &context->op)<0){
av_log(avccontext, AV_LOG_ERROR, "%d. vorbis header damaged\n", i+1);
return -1;
}
}
avccontext->channels = context->vi.channels;
avccontext->sample_rate = context->vi.rate;
avccontext->time_base= (AVRational){1, avccontext->sample_rate};
vorbis_synthesis_init(&context->vd, &context->vi);
vorbis_block_init(&context->vd, &context->vb);
return 0 ;
}
static inline int conv(int samples, float **pcm, char *buf, int channels) {
int i, j, val ;
ogg_int16_t *ptr, *data = (ogg_int16_t*)buf ;
float *mono ;
for(i = 0 ; i < channels ; i++){
ptr = &data[i];
mono = pcm[i] ;
for(j = 0 ; j < samples ; j++) {
val = mono[j] * 32767.f;
if(val > 32767) val = 32767 ;
if(val < -32768) val = -32768 ;
*ptr = val ;
ptr += channels;
}
}
return 0 ;
}
static int oggvorbis_decode_frame(AVCodecContext *avccontext,
void *data, int *data_size,
uint8_t *buf, int buf_size)
{
OggVorbisContext *context = avccontext->priv_data ;
float **pcm ;
ogg_packet *op= &context->op;
int samples, total_samples, total_bytes;
if(!buf_size){
//FIXME flush
return 0;
}
op->packet = buf;
op->bytes = buf_size;
// av_log(avccontext, AV_LOG_DEBUG, "%d %d %d %"PRId64" %"PRId64" %d %d\n", op->bytes, op->b_o_s, op->e_o_s, op->granulepos, op->packetno, buf_size, context->vi.rate);
/* for(i=0; i<op->bytes; i++)
av_log(avccontext, AV_LOG_DEBUG, "%02X ", op->packet[i]);
av_log(avccontext, AV_LOG_DEBUG, "\n");*/
if(vorbis_synthesis(&context->vb, op) == 0)
vorbis_synthesis_blockin(&context->vd, &context->vb) ;
total_samples = 0 ;
total_bytes = 0 ;
while((samples = vorbis_synthesis_pcmout(&context->vd, &pcm)) > 0) {
conv(samples, pcm, (char*)data + total_bytes, context->vi.channels) ;
total_bytes += samples * 2 * context->vi.channels ;
total_samples += samples ;
vorbis_synthesis_read(&context->vd, samples) ;
}
*data_size = total_bytes ;
return buf_size ;
}
static int oggvorbis_decode_close(AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
vorbis_info_clear(&context->vi) ;
vorbis_comment_clear(&context->vc) ;
return 0 ;
}
AVCodec libvorbis_decoder = {
"libvorbis",
CODEC_TYPE_AUDIO,
CODEC_ID_VORBIS,
sizeof(OggVorbisContext),
oggvorbis_decode_init,
NULL,
oggvorbis_decode_close,
oggvorbis_decode_frame,
.capabilities= CODEC_CAP_DELAY,
} ;