aarch64: Add assembly support for -fsanitize=hwaddress tagged globals.
[libav.git] / libavcodec / libvorbis.c
blob972ca6a59e5c9425f9f7610dcfe77154bca77aa6
1 /*
2 * copyright (c) 2002 Mark Hills <mark@pogo.org.uk>
4 * This file is part of Libav.
6 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 /**
22 * @file
23 * Vorbis encoding support via libvorbisenc.
24 * @author Mark Hills <mark@pogo.org.uk>
27 #include <vorbis/vorbisenc.h>
29 #include "libavutil/fifo.h"
30 #include "libavutil/opt.h"
31 #include "avcodec.h"
32 #include "audio_frame_queue.h"
33 #include "bytestream.h"
34 #include "internal.h"
35 #include "vorbis.h"
36 #include "vorbis_parser.h"
38 #undef NDEBUG
39 #include <assert.h>
41 /* Number of samples the user should send in each call.
42 * This value is used because it is the LCD of all possible frame sizes, so
43 * an output packet will always start at the same point as one of the input
44 * packets.
46 #define LIBVORBIS_FRAME_SIZE 64
48 #define BUFFER_SIZE (1024 * 64)
50 typedef struct LibvorbisContext {
51 AVClass *av_class; /**< class for AVOptions */
52 vorbis_info vi; /**< vorbis_info used during init */
53 vorbis_dsp_state vd; /**< DSP state used for analysis */
54 vorbis_block vb; /**< vorbis_block used for analysis */
55 AVFifoBuffer *pkt_fifo; /**< output packet buffer */
56 int eof; /**< end-of-file flag */
57 int dsp_initialized; /**< vd has been initialized */
58 vorbis_comment vc; /**< VorbisComment info */
59 ogg_packet op; /**< ogg packet */
60 double iblock; /**< impulse block bias option */
61 AVVorbisParseContext *vp; /**< parse context to get durations */
62 AudioFrameQueue afq; /**< frame queue for timestamps */
63 } LibvorbisContext;
65 static const AVOption options[] = {
66 { "iblock", "Sets the impulse block bias", offsetof(LibvorbisContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
67 { NULL }
70 static const AVCodecDefault defaults[] = {
71 { "b", "0" },
72 { NULL },
75 static const AVClass class = {
76 .class_name = "libvorbis",
77 .item_name = av_default_item_name,
78 .option = options,
79 .version = LIBAVUTIL_VERSION_INT,
83 static int vorbis_error_to_averror(int ov_err)
85 switch (ov_err) {
86 case OV_EFAULT: return AVERROR_BUG;
87 case OV_EINVAL: return AVERROR(EINVAL);
88 case OV_EIMPL: return AVERROR(EINVAL);
89 default: return AVERROR_UNKNOWN;
93 static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
95 LibvorbisContext *s = avctx->priv_data;
96 double cfreq;
97 int ret;
99 if (avctx->flags & AV_CODEC_FLAG_QSCALE || !avctx->bit_rate) {
100 /* variable bitrate
101 * NOTE: we use the oggenc range of -1 to 10 for global_quality for
102 * user convenience, but libvorbis uses -0.1 to 1.0.
104 float q = avctx->global_quality / (float)FF_QP2LAMBDA;
105 /* default to 3 if the user did not set quality or bitrate */
106 if (!(avctx->flags & AV_CODEC_FLAG_QSCALE))
107 q = 3.0;
108 if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
109 avctx->sample_rate,
110 q / 10.0)))
111 goto error;
112 } else {
113 int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
114 int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
116 /* average bitrate */
117 if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
118 avctx->sample_rate, maxrate,
119 avctx->bit_rate, minrate)))
120 goto error;
122 /* variable bitrate by estimate, disable slow rate management */
123 if (minrate == -1 && maxrate == -1)
124 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))
125 goto error;
128 /* cutoff frequency */
129 if (avctx->cutoff > 0) {
130 cfreq = avctx->cutoff / 1000.0;
131 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))
132 goto error;
135 /* impulse block bias */
136 if (s->iblock) {
137 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
138 goto error;
141 if ((ret = vorbis_encode_setup_init(vi)))
142 goto error;
144 return 0;
145 error:
146 return vorbis_error_to_averror(ret);
149 /* How many bytes are needed for a buffer of length 'l' */
150 static int xiph_len(int l)
152 return 1 + l / 255 + l;
155 static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
157 LibvorbisContext *s = avctx->priv_data;
159 /* notify vorbisenc this is EOF */
160 if (s->dsp_initialized)
161 vorbis_analysis_wrote(&s->vd, 0);
163 vorbis_block_clear(&s->vb);
164 vorbis_dsp_clear(&s->vd);
165 vorbis_info_clear(&s->vi);
167 av_fifo_free(s->pkt_fifo);
168 ff_af_queue_close(&s->afq);
169 av_freep(&avctx->extradata);
171 av_vorbis_parse_free(&s->vp);
173 return 0;
176 static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
178 LibvorbisContext *s = avctx->priv_data;
179 ogg_packet header, header_comm, header_code;
180 uint8_t *p;
181 unsigned int offset;
182 int ret;
184 vorbis_info_init(&s->vi);
185 if ((ret = libvorbis_setup(&s->vi, avctx))) {
186 av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
187 goto error;
189 if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) {
190 av_log(avctx, AV_LOG_ERROR, "analysis init failed\n");
191 ret = vorbis_error_to_averror(ret);
192 goto error;
194 s->dsp_initialized = 1;
195 if ((ret = vorbis_block_init(&s->vd, &s->vb))) {
196 av_log(avctx, AV_LOG_ERROR, "dsp init failed\n");
197 ret = vorbis_error_to_averror(ret);
198 goto error;
201 vorbis_comment_init(&s->vc);
202 vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
204 if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
205 &header_code))) {
206 ret = vorbis_error_to_averror(ret);
207 goto error;
210 avctx->extradata_size = 1 + xiph_len(header.bytes) +
211 xiph_len(header_comm.bytes) +
212 header_code.bytes;
213 p = avctx->extradata = av_malloc(avctx->extradata_size +
214 AV_INPUT_BUFFER_PADDING_SIZE);
215 if (!p) {
216 ret = AVERROR(ENOMEM);
217 goto error;
219 p[0] = 2;
220 offset = 1;
221 offset += av_xiphlacing(&p[offset], header.bytes);
222 offset += av_xiphlacing(&p[offset], header_comm.bytes);
223 memcpy(&p[offset], header.packet, header.bytes);
224 offset += header.bytes;
225 memcpy(&p[offset], header_comm.packet, header_comm.bytes);
226 offset += header_comm.bytes;
227 memcpy(&p[offset], header_code.packet, header_code.bytes);
228 offset += header_code.bytes;
229 assert(offset == avctx->extradata_size);
231 s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size);
232 if (!s->vp) {
233 av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
234 return ret;
237 vorbis_comment_clear(&s->vc);
239 avctx->frame_size = LIBVORBIS_FRAME_SIZE;
240 ff_af_queue_init(avctx, &s->afq);
242 s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
243 if (!s->pkt_fifo) {
244 ret = AVERROR(ENOMEM);
245 goto error;
248 return 0;
249 error:
250 libvorbis_encode_close(avctx);
251 return ret;
254 static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
255 const AVFrame *frame, int *got_packet_ptr)
257 LibvorbisContext *s = avctx->priv_data;
258 ogg_packet op;
259 int ret, duration;
261 /* send samples to libvorbis */
262 if (frame) {
263 const int samples = frame->nb_samples;
264 float **buffer;
265 int c, channels = s->vi.channels;
267 buffer = vorbis_analysis_buffer(&s->vd, samples);
268 for (c = 0; c < channels; c++) {
269 int co = (channels > 8) ? c :
270 ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
271 memcpy(buffer[c], frame->extended_data[co],
272 samples * sizeof(*buffer[c]));
274 if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
275 av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
276 return vorbis_error_to_averror(ret);
278 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
279 return ret;
280 } else {
281 if (!s->eof)
282 if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
283 av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
284 return vorbis_error_to_averror(ret);
286 s->eof = 1;
289 /* retrieve available packets from libvorbis */
290 while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {
291 if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)
292 break;
293 if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)
294 break;
296 /* add any available packets to the output packet buffer */
297 while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {
298 if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
299 av_log(avctx, AV_LOG_ERROR, "packet buffer is too small");
300 return AVERROR_BUG;
302 av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
303 av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
305 if (ret < 0) {
306 av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
307 break;
310 if (ret < 0) {
311 av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");
312 return vorbis_error_to_averror(ret);
315 /* check for available packets */
316 if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
317 return 0;
319 av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
321 if ((ret = ff_alloc_packet(avpkt, op.bytes))) {
322 av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
323 return ret;
325 av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
327 avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
329 duration = av_vorbis_parse_frame(s->vp, avpkt->data, avpkt->size);
330 if (duration > 0) {
331 /* we do not know encoder delay until we get the first packet from
332 * libvorbis, so we have to update the AudioFrameQueue counts */
333 if (!avctx->initial_padding) {
334 avctx->initial_padding = duration;
335 s->afq.remaining_delay += duration;
336 s->afq.remaining_samples += duration;
338 ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
341 *got_packet_ptr = 1;
342 return 0;
345 AVCodec ff_libvorbis_encoder = {
346 .name = "libvorbis",
347 .long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
348 .type = AVMEDIA_TYPE_AUDIO,
349 .id = AV_CODEC_ID_VORBIS,
350 .priv_data_size = sizeof(LibvorbisContext),
351 .init = libvorbis_encode_init,
352 .encode2 = libvorbis_encode_frame,
353 .close = libvorbis_encode_close,
354 .capabilities = AV_CODEC_CAP_DELAY,
355 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
356 AV_SAMPLE_FMT_NONE },
357 .priv_class = &class,
358 .defaults = defaults,
359 .wrapper_name = "libvorbis",