2 * copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
4 * This file is part of FFmpeg.
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 * Ogg Vorbis codec support via libvorbisenc.
24 * @author Mark Hills <mark@pogo.org.uk>
27 #include <vorbis/vorbisenc.h>
29 #include "libavutil/opt.h"
31 #include "bytestream.h"
37 #define OGGVORBIS_FRAME_SIZE 64
39 #define BUFFER_SIZE (1024*64)
41 typedef struct OggVorbisContext {
46 uint8_t buffer[BUFFER_SIZE];
57 static const AVOption options[]={
58 {"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},
61 static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
63 static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext) {
64 OggVorbisContext *context = avccontext->priv_data ;
67 if(avccontext->flags & CODEC_FLAG_QSCALE) {
68 /* variable bitrate */
69 if(vorbis_encode_setup_vbr(vi, avccontext->channels,
70 avccontext->sample_rate,
71 avccontext->global_quality / (float)FF_QP2LAMBDA / 10.0))
74 int minrate = avccontext->rc_min_rate > 0 ? avccontext->rc_min_rate : -1;
75 int maxrate = avccontext->rc_min_rate > 0 ? avccontext->rc_max_rate : -1;
77 /* constant bitrate */
78 if(vorbis_encode_setup_managed(vi, avccontext->channels,
79 avccontext->sample_rate, minrate, avccontext->bit_rate, maxrate))
82 /* variable bitrate by estimate, disable slow rate management */
83 if(minrate == -1 && maxrate == -1)
84 if(vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))
88 /* cutoff frequency */
89 if(avccontext->cutoff > 0) {
90 cfreq = avccontext->cutoff / 1000.0;
91 if(vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))
96 vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock);
99 return vorbis_encode_setup_init(vi);
102 /* How many bytes are needed for a buffer of length 'l' */
103 static int xiph_len(int l) { return (1 + l / 255 + l); }
105 static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext) {
106 OggVorbisContext *context = avccontext->priv_data ;
107 ogg_packet header, header_comm, header_code;
111 vorbis_info_init(&context->vi) ;
112 if(oggvorbis_init_encoder(&context->vi, avccontext) < 0) {
113 av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n") ;
116 vorbis_analysis_init(&context->vd, &context->vi) ;
117 vorbis_block_init(&context->vd, &context->vb) ;
119 vorbis_comment_init(&context->vc);
120 vorbis_comment_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT) ;
122 vorbis_analysis_headerout(&context->vd, &context->vc, &header,
123 &header_comm, &header_code);
125 avccontext->extradata_size=
126 1 + xiph_len(header.bytes) + xiph_len(header_comm.bytes) +
128 p = avccontext->extradata =
129 av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
132 offset += av_xiphlacing(&p[offset], header.bytes);
133 offset += av_xiphlacing(&p[offset], header_comm.bytes);
134 memcpy(&p[offset], header.packet, header.bytes);
135 offset += header.bytes;
136 memcpy(&p[offset], header_comm.packet, header_comm.bytes);
137 offset += header_comm.bytes;
138 memcpy(&p[offset], header_code.packet, header_code.bytes);
139 offset += header_code.bytes;
140 assert(offset == avccontext->extradata_size);
142 /* vorbis_block_clear(&context->vb);
143 vorbis_dsp_clear(&context->vd);
144 vorbis_info_clear(&context->vi);*/
145 vorbis_comment_clear(&context->vc);
147 avccontext->frame_size = OGGVORBIS_FRAME_SIZE ;
149 avccontext->coded_frame= avcodec_alloc_frame();
150 avccontext->coded_frame->key_frame= 1;
156 static int oggvorbis_encode_frame(AVCodecContext *avccontext,
157 unsigned char *packets,
158 int buf_size, void *data)
160 OggVorbisContext *context = avccontext->priv_data ;
162 signed short *audio = data ;
166 const int samples = avccontext->frame_size;
168 int c, channels = context->vi.channels;
170 buffer = vorbis_analysis_buffer(&context->vd, samples) ;
171 for (c = 0; c < channels; c++) {
172 int co = (channels > 8) ? c :
173 ff_vorbis_encoding_channel_layout_offsets[channels-1][c];
174 for(l = 0 ; l < samples ; l++)
175 buffer[c][l]=audio[l*channels+co]/32768.f;
177 vorbis_analysis_wrote(&context->vd, samples) ;
180 vorbis_analysis_wrote(&context->vd, 0) ;
184 while(vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
185 vorbis_analysis(&context->vb, NULL);
186 vorbis_bitrate_addblock(&context->vb) ;
188 while(vorbis_bitrate_flushpacket(&context->vd, &op)) {
189 /* i'd love to say the following line is a hack, but sadly it's
190 * not, apparently the end of stream decision is in libogg. */
191 if(op.bytes==1 && op.e_o_s)
193 if (context->buffer_index + sizeof(ogg_packet) + op.bytes > BUFFER_SIZE) {
194 av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
197 memcpy(context->buffer + context->buffer_index, &op, sizeof(ogg_packet));
198 context->buffer_index += sizeof(ogg_packet);
199 memcpy(context->buffer + context->buffer_index, op.packet, op.bytes);
200 context->buffer_index += op.bytes;
201 // av_log(avccontext, AV_LOG_DEBUG, "e%d / %d\n", context->buffer_index, op.bytes);
206 if(context->buffer_index){
207 ogg_packet *op2= (ogg_packet*)context->buffer;
208 op2->packet = context->buffer + sizeof(ogg_packet);
211 avccontext->coded_frame->pts= av_rescale_q(op2->granulepos, (AVRational){1, avccontext->sample_rate}, avccontext->time_base);
212 //FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
215 av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.");
219 memcpy(packets, op2->packet, l);
220 context->buffer_index -= l + sizeof(ogg_packet);
221 memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
222 // av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);
229 static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext) {
230 OggVorbisContext *context = avccontext->priv_data ;
231 /* ogg_packet op ; */
233 vorbis_analysis_wrote(&context->vd, 0) ; /* notify vorbisenc this is EOF */
235 vorbis_block_clear(&context->vb);
236 vorbis_dsp_clear(&context->vd);
237 vorbis_info_clear(&context->vi);
239 av_freep(&avccontext->coded_frame);
240 av_freep(&avccontext->extradata);
246 AVCodec ff_libvorbis_encoder = {
250 sizeof(OggVorbisContext),
251 oggvorbis_encode_init,
252 oggvorbis_encode_frame,
253 oggvorbis_encode_close,
254 .capabilities= CODEC_CAP_DELAY,
255 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
256 .long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),