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path: root/libavcodec/libmp3lame.c (plain)
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1/*
2 * Interface to libmp3lame for mp3 encoding
3 * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22/**
23 * @file
24 * Interface to libmp3lame for mp3 encoding.
25 */
26
27#include <lame/lame.h>
28
29#include "libavutil/channel_layout.h"
30#include "libavutil/common.h"
31#include "libavutil/float_dsp.h"
32#include "libavutil/intreadwrite.h"
33#include "libavutil/log.h"
34#include "libavutil/opt.h"
35#include "avcodec.h"
36#include "audio_frame_queue.h"
37#include "internal.h"
38#include "mpegaudio.h"
39#include "mpegaudiodecheader.h"
40
41#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
42
43typedef struct LAMEContext {
44 AVClass *class;
45 AVCodecContext *avctx;
46 lame_global_flags *gfp;
47 uint8_t *buffer;
48 int buffer_index;
49 int buffer_size;
50 int reservoir;
51 int joint_stereo;
52 int abr;
53 int delay_sent;
54 float *samples_flt[2];
55 AudioFrameQueue afq;
56 AVFloatDSPContext *fdsp;
57} LAMEContext;
58
59
60static int realloc_buffer(LAMEContext *s)
61{
62 if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
63 int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
64
65 ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
66 new_size);
67 if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
68 s->buffer_size = s->buffer_index = 0;
69 return err;
70 }
71 s->buffer_size = new_size;
72 }
73 return 0;
74}
75
76static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
77{
78 LAMEContext *s = avctx->priv_data;
79
80 av_freep(&s->samples_flt[0]);
81 av_freep(&s->samples_flt[1]);
82 av_freep(&s->buffer);
83 av_freep(&s->fdsp);
84
85 ff_af_queue_close(&s->afq);
86
87 lame_close(s->gfp);
88 return 0;
89}
90
91static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
92{
93 LAMEContext *s = avctx->priv_data;
94 int ret;
95
96 s->avctx = avctx;
97
98 /* initialize LAME and get defaults */
99 if (!(s->gfp = lame_init()))
100 return AVERROR(ENOMEM);
101
102
103 lame_set_num_channels(s->gfp, avctx->channels);
104 lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
105
106 /* sample rate */
107 lame_set_in_samplerate (s->gfp, avctx->sample_rate);
108 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
109
110 /* algorithmic quality */
111 if (avctx->compression_level != FF_COMPRESSION_DEFAULT)
112 lame_set_quality(s->gfp, avctx->compression_level);
113
114 /* rate control */
115 if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
116 lame_set_VBR(s->gfp, vbr_default);
117 lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
118 } else {
119 if (avctx->bit_rate) {
120 if (s->abr) { // ABR
121 lame_set_VBR(s->gfp, vbr_abr);
122 lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
123 } else // CBR
124 lame_set_brate(s->gfp, avctx->bit_rate / 1000);
125 }
126 }
127
128 /* lowpass cutoff frequency */
129 if (avctx->cutoff)
130 lame_set_lowpassfreq(s->gfp, avctx->cutoff);
131
132 /* do not get a Xing VBR header frame from LAME */
133 lame_set_bWriteVbrTag(s->gfp,0);
134
135 /* bit reservoir usage */
136 lame_set_disable_reservoir(s->gfp, !s->reservoir);
137
138 /* set specified parameters */
139 if (lame_init_params(s->gfp) < 0) {
140 ret = -1;
141 goto error;
142 }
143
144 /* get encoder delay */
145 avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
146 ff_af_queue_init(avctx, &s->afq);
147
148 avctx->frame_size = lame_get_framesize(s->gfp);
149
150 /* allocate float sample buffers */
151 if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
152 int ch;
153 for (ch = 0; ch < avctx->channels; ch++) {
154 s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
155 sizeof(*s->samples_flt[ch]));
156 if (!s->samples_flt[ch]) {
157 ret = AVERROR(ENOMEM);
158 goto error;
159 }
160 }
161 }
162
163 ret = realloc_buffer(s);
164 if (ret < 0)
165 goto error;
166
167 s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
168 if (!s->fdsp) {
169 ret = AVERROR(ENOMEM);
170 goto error;
171 }
172
173
174 return 0;
175error:
176 mp3lame_encode_close(avctx);
177 return ret;
178}
179
180#define ENCODE_BUFFER(func, buf_type, buf_name) do { \
181 lame_result = func(s->gfp, \
182 (const buf_type *)buf_name[0], \
183 (const buf_type *)buf_name[1], frame->nb_samples, \
184 s->buffer + s->buffer_index, \
185 s->buffer_size - s->buffer_index); \
186} while (0)
187
188static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
189 const AVFrame *frame, int *got_packet_ptr)
190{
191 LAMEContext *s = avctx->priv_data;
192 MPADecodeHeader hdr;
193 int len, ret, ch, discard_padding;
194 int lame_result;
195 uint32_t h;
196
197 if (frame) {
198 switch (avctx->sample_fmt) {
199 case AV_SAMPLE_FMT_S16P:
200 ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
201 break;
202 case AV_SAMPLE_FMT_S32P:
203 ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
204 break;
205 case AV_SAMPLE_FMT_FLTP:
206 if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
207 av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
208 return AVERROR(EINVAL);
209 }
210 for (ch = 0; ch < avctx->channels; ch++) {
211 s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
212 (const float *)frame->data[ch],
213 32768.0f,
214 FFALIGN(frame->nb_samples, 8));
215 }
216 ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
217 break;
218 default:
219 return AVERROR_BUG;
220 }
221 } else if (!s->afq.frame_alloc) {
222 lame_result = 0;
223 } else {
224 lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
225 s->buffer_size - s->buffer_index);
226 }
227 if (lame_result < 0) {
228 if (lame_result == -1) {
229 av_log(avctx, AV_LOG_ERROR,
230 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
231 s->buffer_index, s->buffer_size - s->buffer_index);
232 }
233 return -1;
234 }
235 s->buffer_index += lame_result;
236 ret = realloc_buffer(s);
237 if (ret < 0) {
238 av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
239 return ret;
240 }
241
242 /* add current frame to the queue */
243 if (frame) {
244 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
245 return ret;
246 }
247
248 /* Move 1 frame from the LAME buffer to the output packet, if available.
249 We have to parse the first frame header in the output buffer to
250 determine the frame size. */
251 if (s->buffer_index < 4)
252 return 0;
253 h = AV_RB32(s->buffer);
254
255 ret = avpriv_mpegaudio_decode_header(&hdr, h);
256 if (ret < 0) {
257 av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
258 return AVERROR_BUG;
259 } else if (ret) {
260 av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
261 return -1;
262 }
263 len = hdr.frame_size;
264 ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
265 s->buffer_index);
266 if (len <= s->buffer_index) {
267 if ((ret = ff_alloc_packet2(avctx, avpkt, len, 0)) < 0)
268 return ret;
269 memcpy(avpkt->data, s->buffer, len);
270 s->buffer_index -= len;
271 memmove(s->buffer, s->buffer + len, s->buffer_index);
272
273 /* Get the next frame pts/duration */
274 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
275 &avpkt->duration);
276
277 discard_padding = avctx->frame_size - avpkt->duration;
278 // Check if subtraction resulted in an overflow
279 if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
280 av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
281 av_packet_unref(avpkt);
282 av_free(avpkt);
283 return AVERROR(EINVAL);
284 }
285 if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
286 uint8_t* side_data = av_packet_new_side_data(avpkt,
287 AV_PKT_DATA_SKIP_SAMPLES,
288 10);
289 if(!side_data) {
290 av_packet_unref(avpkt);
291 av_free(avpkt);
292 return AVERROR(ENOMEM);
293 }
294 if (!s->delay_sent) {
295 AV_WL32(side_data, avctx->initial_padding);
296 s->delay_sent = 1;
297 }
298 AV_WL32(side_data + 4, discard_padding);
299 }
300
301 avpkt->size = len;
302 *got_packet_ptr = 1;
303 }
304 return 0;
305}
306
307#define OFFSET(x) offsetof(LAMEContext, x)
308#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
309static const AVOption options[] = {
310 { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
311 { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
312 { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
313 { NULL },
314};
315
316static const AVClass libmp3lame_class = {
317 .class_name = "libmp3lame encoder",
318 .item_name = av_default_item_name,
319 .option = options,
320 .version = LIBAVUTIL_VERSION_INT,
321};
322
323static const AVCodecDefault libmp3lame_defaults[] = {
324 { "b", "0" },
325 { NULL },
326};
327
328static const int libmp3lame_sample_rates[] = {
329 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
330};
331
332AVCodec ff_libmp3lame_encoder = {
333 .name = "libmp3lame",
334 .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
335 .type = AVMEDIA_TYPE_AUDIO,
336 .id = AV_CODEC_ID_MP3,
337 .priv_data_size = sizeof(LAMEContext),
338 .init = mp3lame_encode_init,
339 .encode2 = mp3lame_encode_frame,
340 .close = mp3lame_encode_close,
341 .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
342 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
343 AV_SAMPLE_FMT_FLTP,
344 AV_SAMPLE_FMT_S16P,
345 AV_SAMPLE_FMT_NONE },
346 .supported_samplerates = libmp3lame_sample_rates,
347 .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
348 AV_CH_LAYOUT_STEREO,
349 0 },
350 .priv_class = &libmp3lame_class,
351 .defaults = libmp3lame_defaults,
352};
353
generated by cgit at 2024-05-17 20:22:48 (GMT)