FFmpeg: libavcodec/libvorbisenc.c Source File

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libvorbisenc.c

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1 /*

3 *

4 * This file is part of FFmpeg.

5 *

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.

10 *

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.

15 *

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

19 */

21 #include <vorbis/vorbisenc.h>

23 #include "libavutil/avassert.h"

24 #include "libavutil/fifo.h"

25 #include "libavutil/opt.h"

26 #include "avcodec.h"

27 #include "audio_frame_queue.h"

28 #include "internal.h"

29 #include "vorbis.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

36 * packets.

37 */

38 #define OGGVORBIS_FRAME_SIZE 64

40 #define BUFFER_SIZE (1024 * 64)

42 typedef struct OggVorbisEncContext {

43 AVClass *av_class; /**< class for AVOptions */

44 AVFrame frame;

45 vorbis_info vi; /**< vorbis_info used during init */

46 vorbis_dsp_state vd; /**< DSP state used for analysis */

47 vorbis_block vb; /**< vorbis_block used for analysis */

48 AVFifoBuffer *pkt_fifo; /**< output packet buffer */

49 int eof; /**< end-of-file flag */

50 int dsp_initialized; /**< vd has been initialized */

51 vorbis_comment vc; /**< VorbisComment info */

52 double iblock; /**< impulse block bias option */

53 VorbisParseContext vp; /**< parse context to get durations */

54 AudioFrameQueue afq; /**< frame queue for timestamps */

55 } OggVorbisEncContext;

57 static const AVOption options[] = {

58 { "iblock", "Sets the impulse block bias", offsetof(OggVorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },

59 { NULL }

60 };

62 static const AVCodecDefault defaults[] = {

63 { "b", "0" },

64 { NULL },

65 };

67 static const AVClass vorbis_class = {

68 .class_name = "libvorbis",

69 .item_name = av_default_item_name,

70 .option = options,

71 .version = LIBAVUTIL_VERSION_INT,

72 };

74 static int vorbis_error_to_averror(int ov_err)

75 {

76 switch (ov_err) {

77 case OV_EFAULT: return AVERROR_BUG;

78 case OV_EINVAL: return AVERROR(EINVAL);

79 case OV_EIMPL: return AVERROR(EINVAL);

80 default: return AVERROR_UNKNOWN;

81 }

82 }

84 static av_cold int oggvorbis_init_encoder(vorbis_info *vi,

85 AVCodecContext *avctx)

86 {

87 OggVorbisEncContext *s = avctx->priv_data;

88 double cfreq;

89 int ret;

91 if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) {

92 /* variable bitrate

93 * NOTE: we use the oggenc range of -1 to 10 for global_quality for

94 * user convenience, but libvorbis uses -0.1 to 1.0.

95 */

96 float q = avctx->global_quality / (float)FF_QP2LAMBDA;

97 /* default to 3 if the user did not set quality or bitrate */

98 if (!(avctx->flags & CODEC_FLAG_QSCALE))

99 q = 3.0;

100 if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,

101 avctx->sample_rate,

102 q / 10.0)))

103 goto error;

104 } else {

105 int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;

106 int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;

107

108 /* average bitrate */

109 if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,

110 avctx->sample_rate, maxrate,

111 avctx->bit_rate, minrate)))

112 goto error;

113

114 /* variable bitrate by estimate, disable slow rate management */

115 if (minrate == -1 && maxrate == -1)

116 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL)))

117 goto error; /* should not happen */

118 }

119

120 /* cutoff frequency */

121 if (avctx->cutoff > 0) {

122 cfreq = avctx->cutoff / 1000.0;

123 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq)))

124 goto error; /* should not happen */

125 }

126

127 /* impulse block bias */

128 if (s->iblock) {

129 if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))

130 goto error;

131 }

132

133 if (avctx->channels == 3 &&

134 avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||

135 avctx->channels == 4 &&

136 avctx->channel_layout != AV_CH_LAYOUT_2_2 &&

137 avctx->channel_layout != AV_CH_LAYOUT_QUAD ||

138 avctx->channels == 5 &&

139 avctx->channel_layout != AV_CH_LAYOUT_5POINT0 &&

140 avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||

141 avctx->channels == 6 &&

142 avctx->channel_layout != AV_CH_LAYOUT_5POINT1 &&

143 avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK ||

144 avctx->channels == 7 &&

145 avctx->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) ||

146 avctx->channels == 8 &&

147 avctx->channel_layout != AV_CH_LAYOUT_7POINT1) {

148 if (avctx->channel_layout) {

149 char name[32];

150 av_get_channel_layout_string(name, sizeof(name), avctx->channels,

151 avctx->channel_layout);

152 av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "

153 "output stream will have incorrect "

154 "channel layout.\n", name);

155 } else {

156 av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "

157 "will use Vorbis channel layout for "

158 "%d channels.\n", avctx->channels);

159 }

160 }

161

162 if ((ret = vorbis_encode_setup_init(vi)))

163 goto error;

164

165 return 0;

166 error:

167 return vorbis_error_to_averror(ret);

168 }

169

170 /* How many bytes are needed for a buffer of length 'l' */

171 static int xiph_len(int l)

172 {

173 return 1 + l / 255 + l;

174 }

175

176 static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)

177 {

178 OggVorbisEncContext *s = avctx->priv_data;

179

180 /* notify vorbisenc this is EOF */

181 if (s->dsp_initialized)

182 vorbis_analysis_wrote(&s->vd, 0);

183

184 vorbis_block_clear(&s->vb);

185 vorbis_dsp_clear(&s->vd);

186 vorbis_info_clear(&s->vi);

187

188 av_fifo_free(s->pkt_fifo);

189 ff_af_queue_close(&s->afq);

190 av_freep(&avctx->extradata);

191

192 return 0;

193 }

194

195 static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)

196 {

197 OggVorbisEncContext *s = avctx->priv_data;

198 ogg_packet header, header_comm, header_code;

199 uint8_t *p;

200 unsigned int offset;

201 int ret;

202

203 vorbis_info_init(&s->vi);

204 if ((ret = oggvorbis_init_encoder(&s->vi, avctx))) {

205 av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");

206 goto error;

207 }

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);

211 goto error;

212 }

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);

217 goto error;

218 }

219

220 vorbis_comment_init(&s->vc);

221 if (!(avctx->flags & CODEC_FLAG_BITEXACT))

222 vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);

223

224 if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,

225 &header_code))) {

226 ret = vorbis_error_to_averror(ret);

227 goto error;

228 }

229

230 avctx->extradata_size = 1 + xiph_len(header.bytes) +

231 xiph_len(header_comm.bytes) +

232 header_code.bytes;

233 p = avctx->extradata = av_malloc(avctx->extradata_size +

234 FF_INPUT_BUFFER_PADDING_SIZE);

235 if (!p) {

236 ret = AVERROR(ENOMEM);

237 goto error;

238 }

239 p[0] = 2;

240 offset = 1;

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);

250

251 if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {

252 av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");

253 return ret;

254 }

255

256 vorbis_comment_clear(&s->vc);

257

258 avctx->frame_size = OGGVORBIS_FRAME_SIZE;

259 ff_af_queue_init(avctx, &s->afq);

260

261 s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);

262 if (!s->pkt_fifo) {

263 ret = AVERROR(ENOMEM);

264 goto error;

265 }

266

267 return 0;

268 error:

269 oggvorbis_encode_close(avctx);

270 return ret;

271 }

272

273 static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

274 const AVFrame *frame, int *got_packet_ptr)

275 {

276 OggVorbisEncContext *s = avctx->priv_data;

277 ogg_packet op;

278 int ret, duration;

279

280 /* send samples to libvorbis */

281 if (frame) {

282 const int samples = frame->nb_samples;

283 float **buffer;

284 int c, channels = s->vi.channels;

285

286 buffer = vorbis_analysis_buffer(&s->vd, samples);

287 for (c = 0; c < channels; c++) {

288 int co = (channels > 8) ? c :

289 ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];

290 memcpy(buffer[c], frame->extended_data[co],

291 samples * sizeof(*buffer[c]));

292 }

293 if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {

294 av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");

295 return vorbis_error_to_averror(ret);

296 }

297 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)

298 return ret;

299 } else {

300 if (!s->eof)

301 if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {

302 av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");

303 return vorbis_error_to_averror(ret);

304 }

305 s->eof = 1;

306 }

307

308 /* retrieve available packets from libvorbis */

309 while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) {

310 if ((ret = vorbis_analysis(&s->vb, NULL)) < 0)

311 break;

312 if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0)

313 break;

314

315 /* add any available packets to the output packet buffer */

316 while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) {

317 if (av_fifo_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {

318 av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");

319 return AVERROR_BUG;

320 }

321 av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);

322 av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);

323 }

324 if (ret < 0) {

325 av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");

326 break;

327 }

328 }

329 if (ret < 0) {

330 av_log(avctx, AV_LOG_ERROR, "error getting available packets\n");

331 return vorbis_error_to_averror(ret);

332 }

333

334 /* check for available packets */

335 if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))

336 return 0;

337

338 av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);

339

340 if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes)) < 0)

341 return ret;

342 av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);

343

344 avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);

345

346 duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);

347 if (duration > 0) {

348 /* we do not know encoder delay until we get the first packet from

349 * libvorbis, so we have to update the AudioFrameQueue counts */

350 if (!avctx->delay && s->afq.frames) {

351 avctx->delay = duration;

352 av_assert0(!s->afq.remaining_delay);

353 s->afq.frames->duration += duration;

354 s->afq.frames->pts -= duration;

355 s->afq.remaining_samples += duration;

356 }

357 ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);

358 }

359

360 *got_packet_ptr = 1;

361 return 0;

362 }

363

364 AVCodec ff_libvorbis_encoder = {

365 .name = "libvorbis",

366 .type = AVMEDIA_TYPE_AUDIO,

367 .id = AV_CODEC_ID_VORBIS,

368 .priv_data_size = sizeof(OggVorbisEncContext),

369 .init = oggvorbis_encode_init,

370 .encode2 = oggvorbis_encode_frame,

371 .close = oggvorbis_encode_close,

372 .capabilities = CODEC_CAP_DELAY,

373 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

374 AV_SAMPLE_FMT_NONE },

375 .long_name = NULL_IF_CONFIG_SMALL("libvorbis"),

376 .priv_class = &vorbis_class,

377 .defaults = defaults,

378 };

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