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
21 #include <vorbis/vorbisenc.h>
23 #include "libavutil/avassert.h"
24 #include "libavutil/fifo.h"
25 #include "libavutil/opt.h"
27 #include "audio_frame_queue.h"
30 #include "vorbis_parser.h"
33 /* Number of samples the user should send in each call.
34 * This value is used because it is the LCD of all possible frame sizes, so
35 * an output packet will always start at the same point as one of the input
38 #define LIBVORBIS_FRAME_SIZE 64
40 #define BUFFER_SIZE (1024 * 64)
42 typedef struct LibvorbisEncContext {
43 AVClass *av_class; /**< class for AVOptions */
44 vorbis_info vi; /**< vorbis_info used during init */
45 vorbis_dsp_state vd; /**< DSP state used for analysis */
46 vorbis_block vb; /**< vorbis_block used for analysis */
47 AVFifoBuffer *pkt_fifo; /**< output packet buffer */
48 int eof; /**< end-of-file flag */
49 int dsp_initialized; /**< vd has been initialized */
50 vorbis_comment vc; /**< VorbisComment info */
51 double iblock; /**< impulse block bias option */
52 AVVorbisParseContext *vp; /**< parse context to get durations */
53 AudioFrameQueue afq; /**< frame queue for timestamps */
54 } LibvorbisEncContext;
56 static const AVOption options[] = {
57 { "iblock", "Sets the impulse block bias", offsetof(LibvorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
61 static const AVCodecDefault defaults[] = {
66 static const AVClass vorbis_class = {
67 .class_name = "libvorbis",
68 .item_name = av_default_item_name,
70 .version = LIBAVUTIL_VERSION_INT,
73 static int vorbis_error_to_averror(int ov_err)
76 case OV_EFAULT: return AVERROR_BUG;
77 case OV_EINVAL: return AVERROR(EINVAL);
78 case OV_EIMPL: return AVERROR(EINVAL);
79 default: return AVERROR_UNKNOWN;
83 static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
85 LibvorbisEncContext *s = avctx->priv_data;
89 if (avctx->flags & AV_CODEC_FLAG_QSCALE || !avctx->bit_rate) {
91 * NOTE: we use the oggenc range of -1 to 10 for global_quality for
92 * user convenience, but libvorbis uses -0.1 to 1.0.
94 float q = avctx->global_quality / (float)FF_QP2LAMBDA;
95 /* default to 3 if the user did not set quality or bitrate */
96 if (!(avctx->flags & AV_CODEC_FLAG_QSCALE))
98 if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
103 int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
104 int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
106 /* average bitrate */
107 if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
108 avctx->sample_rate, maxrate,
109 avctx->bit_rate, minrate)))
112 /* variable bitrate by estimate, disable slow rate management */
113 if (minrate == -1 && maxrate == -1)
114 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
115 goto error; /* should not happen */
118 /* cutoff frequency */
119 if (avctx->cutoff > 0) {
120 cfreq = avctx->cutoff / 1000.0;
121 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
122 goto error; /* should not happen */
125 /* impulse block bias */
127 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
131 if (avctx->channels == 3 &&
132 avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
133 avctx->channels == 4 &&
134 avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
135 avctx->channel_layout != AV_CH_LAYOUT_QUAD ||
136 avctx->channels == 5 &&
137 avctx->channel_layout != AV_CH_LAYOUT_5POINT0 &&
138 avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||
139 avctx->channels == 6 &&
140 avctx->channel_layout != AV_CH_LAYOUT_5POINT1 &&
141 avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK ||
142 avctx->channels == 7 &&
143 avctx->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) ||
144 avctx->channels == 8 &&
145 avctx->channel_layout != AV_CH_LAYOUT_7POINT1) {
146 if (avctx->channel_layout) {
148 av_get_channel_layout_string(name, sizeof(name), avctx->channels,
149 avctx->channel_layout);
150 av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
151 "output stream will have incorrect "
152 "channel layout.\n", name);
154 av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
155 "will use Vorbis channel layout for "
156 "%d channels.\n", avctx->channels);
160 if ((ret = vorbis_encode_setup_init(vi)))
165 return vorbis_error_to_averror(ret);
168 /* How many bytes are needed for a buffer of length 'l' */
169 static int xiph_len(int l)
171 return 1 + l / 255 + l;
174 static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
176 LibvorbisEncContext *s = avctx->priv_data;
178 /* notify vorbisenc this is EOF */
179 if (s->dsp_initialized)
180 vorbis_analysis_wrote(&s->vd, 0);
182 vorbis_block_clear(&s->vb);
183 vorbis_dsp_clear(&s->vd);
184 vorbis_info_clear(&s->vi);
186 av_fifo_freep(&s->pkt_fifo);
187 ff_af_queue_close(&s->afq);
188 av_freep(&avctx->extradata);
190 av_vorbis_parse_free(&s->vp);
195 static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
197 LibvorbisEncContext *s = avctx->priv_data;
198 ogg_packet header, header_comm, header_code;
203 vorbis_info_init(&s->vi);
204 if ((ret = libvorbis_setup(&s->vi, avctx))) {
205 av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
208 if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
209 av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
210 ret = vorbis_error_to_averror(ret);
213 s->dsp_initialized = 1;
214 if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
215 av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
216 ret = vorbis_error_to_averror(ret);
220 vorbis_comment_init(&s->vc);
221 if (!(avctx->flags & AV_CODEC_FLAG_BITEXACT))
222 vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
224 if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
226 ret = vorbis_error_to_averror(ret);
230 avctx->extradata_size = 1 + xiph_len(header.bytes) +
231 xiph_len(header_comm.bytes) +
233 p = avctx->extradata = av_malloc(avctx->extradata_size +
234 AV_INPUT_BUFFER_PADDING_SIZE);
236 ret = AVERROR(ENOMEM);
241 offset += av_xiphlacing(&p[offset], header.bytes);
242 offset += av_xiphlacing(&p[offset], header_comm.bytes);
243 memcpy(&p[offset], header.packet, header.bytes);
244 offset += header.bytes;
245 memcpy(&p[offset], header_comm.packet, header_comm.bytes);
246 offset += header_comm.bytes;
247 memcpy(&p[offset], header_code.packet, header_code.bytes);
248 offset += header_code.bytes;
249 av_assert0(offset == avctx->extradata_size);
251 s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size);
253 av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
257 vorbis_comment_clear(&s->vc);
259 avctx->frame_size = LIBVORBIS_FRAME_SIZE;
260 ff_af_queue_init(avctx, &s->afq);
262 s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
264 ret = AVERROR(ENOMEM);
270 libvorbis_encode_close(avctx);
274 static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
275 const AVFrame *frame, int *got_packet_ptr)
277 LibvorbisEncContext *s = avctx->priv_data;
281 /* send samples to libvorbis */
283 const int samples = frame->nb_samples;
285 int c, channels = s->vi.channels;
287 buffer = vorbis_analysis_buffer(&s->vd, samples);
288 for (c = 0; c < channels; c++) {
289 int co = (channels > 8) ? c :
290 ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
291 memcpy(buffer[c], frame->extended_data[co],
292 samples * sizeof(*buffer[c]));
294 if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
295 av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
296 return vorbis_error_to_averror(ret);
298 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
301 if (!s->eof && s->afq.frame_alloc)
302 if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
303 av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
304 return vorbis_error_to_averror(ret);
309 /* retrieve available packets from libvorbis */
310 while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
311 if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
313 if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
316 /* add any available packets to the output packet buffer */
317 while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
318 if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
319 av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
322 av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
323 av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
326 av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
331 av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
332 return vorbis_error_to_averror(ret);
335 /* check for available packets */
336 if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
339 av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
341 if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes, 0)) < 0)
343 av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
345 avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
347 duration = av_vorbis_parse_frame(s->vp, avpkt->data, avpkt->size);
349 /* we do not know encoder delay until we get the first packet from
350 * libvorbis, so we have to update the AudioFrameQueue counts */
351 if (!avctx->initial_padding && s->afq.frames) {
352 avctx->initial_padding = duration;
353 av_assert0(!s->afq.remaining_delay);
354 s->afq.frames->duration += duration;
355 if (s->afq.frames->pts != AV_NOPTS_VALUE)
356 s->afq.frames->pts -= duration;
357 s->afq.remaining_samples += duration;
359 ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
366 AVCodec ff_libvorbis_encoder = {
368 .long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
369 .type = AVMEDIA_TYPE_AUDIO,
370 .id = AV_CODEC_ID_VORBIS,
371 .priv_data_size = sizeof(LibvorbisEncContext),
372 .init = libvorbis_encode_init,
373 .encode2 = libvorbis_encode_frame,
374 .close = libvorbis_encode_close,
375 .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
376 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
377 AV_SAMPLE_FMT_NONE },
378 .priv_class = &vorbis_class,
379 .defaults = defaults,