FFmpeg/libavcodec/libvorbis.c
Carl Eugen Hoyos 7c8c55ff03 Warn if Vorbis does not support the provided channel layout.
The warning is printed with loglevel error because the output
stream is known to be broken.
Also warn if no channel layout was provided for multichannel input.
2011-04-15 11:48:03 +02:00

288 lines
10 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 Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Ogg Vorbis codec support via libvorbisenc.
* @author Mark Hills <mark@pogo.org.uk>
*/
#include <vorbis/vorbisenc.h>
#include "libavutil/opt.h"
#include "avcodec.h"
#include "bytestream.h"
#include "vorbis.h"
#undef NDEBUG
#include <assert.h>
#define OGGVORBIS_FRAME_SIZE 64
#define BUFFER_SIZE (1024*64)
typedef struct OggVorbisContext {
AVClass *av_class;
vorbis_info vi ;
vorbis_dsp_state vd ;
vorbis_block vb ;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
int eof;
/* decoder */
vorbis_comment vc ;
ogg_packet op;
double iblock;
} OggVorbisContext ;
static const AVOption options[]={
{"iblock", "Sets the impulse block bias", offsetof(OggVorbisContext, iblock), FF_OPT_TYPE_DOUBLE, 0, -15, 0, AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_ENCODING_PARAM},
{NULL}
};
static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
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 / 10.0))
return -1;
} else {
int minrate = avccontext->rc_min_rate > 0 ? avccontext->rc_min_rate : -1;
int maxrate = avccontext->rc_min_rate > 0 ? avccontext->rc_max_rate : -1;
/* constant bitrate */
if(vorbis_encode_setup_managed(vi, avccontext->channels,
avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate))
return -1;
/* variable bitrate by estimate, disable slow rate management */
if(minrate == -1 && maxrate == -1)
if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))
return -1;
}
/* cutoff frequency */
if(avccontext->cutoff > 0) {
cfreq = avccontext->cutoff / 1000.0;
if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
return -1;
}
if(context->iblock){
vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock);
}
if (avccontext->channels == 3 &&
avccontext->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
avccontext->channels == 4 &&
avccontext->channel_layout != AV_CH_LAYOUT_2_2 &&
avccontext->channel_layout != AV_CH_LAYOUT_QUAD ||
avccontext->channels == 5 &&
avccontext->channel_layout != AV_CH_LAYOUT_5POINT0 &&
avccontext->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||
avccontext->channels == 6 &&
avccontext->channel_layout != AV_CH_LAYOUT_5POINT1 &&
avccontext->channel_layout != AV_CH_LAYOUT_5POINT1_BACK ||
avccontext->channels == 7 &&
avccontext->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) ||
avccontext->channels == 8 &&
avccontext->channel_layout != AV_CH_LAYOUT_7POINT1) {
if (avccontext->channel_layout) {
char name[32];
av_get_channel_layout_string(name, sizeof(name), avccontext->channels,
avccontext->channel_layout);
av_log(avccontext, AV_LOG_ERROR, "%s not supported by Vorbis: "
"output stream will have incorrect "
"channel layout.\n", name);
} else {
av_log(avccontext, AV_LOG_WARNING, "No channel layout specified. The encoder "
"will use Vorbis channel layout for "
"%d channels.\n", avccontext->channels);
}
}
return vorbis_encode_setup_init(vi);
}
/* How many bytes are needed for a buffer of length 'l' */
static int xiph_len(int l) { return (1 + l / 255 + l); }
static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {
OggVorbisContext *context = avccontext->priv_data ;
ogg_packet header, header_comm, header_code;
uint8_t *p;
unsigned int offset;
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\n") ;
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);
avccontext->extradata_size=
1 + xiph_len(header.bytes) + xiph_len(header_comm.bytes) +
header_code.bytes;
p = avccontext->extradata =
av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_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;
assert(offset == 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 ;
ogg_packet op ;
signed short *audio = data ;
int l;
if(data) {
const int samples = avccontext->frame_size;
float **buffer ;
int c, channels = context->vi.channels;
buffer = vorbis_analysis_buffer(&context->vd, samples) ;
for (c = 0; c < channels; c++) {
int co = (channels > 8) ? c :
ff_vorbis_encoding_channel_layout_offsets[channels-1][c];
for(l = 0 ; l < samples ; l++)
buffer[c][l]=audio[l*channels+co]/32768.f;
}
vorbis_analysis_wrote(&context->vd, samples) ;
} else {
if(!context->eof)
vorbis_analysis_wrote(&context->vd, 0) ;
context->eof = 1;
}
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 && op.e_o_s)
continue;
if (context->buffer_index + sizeof(ogg_packet) + op.bytes > BUFFER_SIZE) {
av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
return -1;
}
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
if (l > buf_size) {
av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
return -1;
}
memcpy(packets, op2->packet, l);
context->buffer_index -= l + sizeof(ogg_packet);
memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
// av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);
}
return l;
}
static av_cold 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 ff_libvorbis_encoder = {
"libvorbis",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_VORBIS,
sizeof(OggVorbisContext),
oggvorbis_encode_init,
oggvorbis_encode_frame,
oggvorbis_encode_close,
.capabilities= CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
.priv_class= &class,
} ;