*/
/**
- * @file libavcodec/libvorbis.c
+ * @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 BUFFER_SIZE (1024*64)
typedef struct OggVorbisContext {
+ AVClass *av_class;
vorbis_info vi ;
vorbis_dsp_state vd ;
vorbis_block vb ;
/* 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, {.dbl = 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 int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
+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))
+ 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, -1, avccontext->bit_rate, -1))
+ avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate))
return -1;
-#ifdef OGGVORBIS_VBR_BY_ESTIMATE
- /* variable bitrate by estimate */
- if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE_AVG, NULL))
- return -1;
-#endif
+ /* 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 */
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, len;
+ 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") ;
+ av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n") ;
return -1 ;
}
vorbis_analysis_init(&context->vd, &context->vi) ;
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);
+ 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 += 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);
+ assert(offset == avccontext->extradata_size);
/* vorbis_block_clear(&context->vb);
vorbis_dsp_clear(&context->vd);
int l;
if(data) {
- int samples = OGGVORBIS_FRAME_SIZE;
+ const int samples = avccontext->frame_size;
float **buffer ;
+ int c, channels = context->vi.channels;
buffer = vorbis_analysis_buffer(&context->vd, samples) ;
- if(context->vi.channels == 1) {
+ 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[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;
- }
+ buffer[c][l]=audio[l*channels+co]/32768.f;
}
vorbis_analysis_wrote(&context->vd, samples) ;
} else {
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)
+ 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);
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);
- memcpy(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
+ memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
// av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);
}
}
-AVCodec libvorbis_encoder = {
+AVCodec ff_libvorbis_encoder = {
"libvorbis",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_VORBIS,
sizeof(OggVorbisContext),
oggvorbis_encode_init,
oggvorbis_encode_frame,
oggvorbis_encode_close,
.capabilities= CODEC_CAP_DELAY,
- .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
+ .priv_class= &class,
} ;
projects / ffmpeg / blobdiff