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 "codec_internal.h"
38 #include "encode.h"
39 #include "mpegaudio.h"
40 #include "mpegaudiodecheader.h"
41
42 #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.
43
44 typedef struct LAMEContext {
45 AVClass *class;
46 AVCodecContext *avctx;
47 lame_global_flags *gfp;
48 uint8_t *buffer;
49 int buffer_index;
50 int buffer_size;
51 int reservoir;
52 int joint_stereo;
53 int abr;
54 int delay_sent;
55 float *samples_flt[2];
56 AudioFrameQueue afq;
57 AVFloatDSPContext *fdsp;
58 } LAMEContext;
59
60
realloc_buffer(LAMEContext *s)61 static int realloc_buffer(LAMEContext *s)
62 {
63 if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
64 int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
65
66 ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
67 new_size);
68 if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
69 s->buffer_size = s->buffer_index = 0;
70 return err;
71 }
72 s->buffer_size = new_size;
73 }
74 return 0;
75 }
76
mp3lame_encode_close(AVCodecContext *avctx)77 static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
78 {
79 LAMEContext *s = avctx->priv_data;
80
81 av_freep(&s->samples_flt[0]);
82 av_freep(&s->samples_flt[1]);
83 av_freep(&s->buffer);
84 av_freep(&s->fdsp);
85
86 ff_af_queue_close(&s->afq);
87
88 lame_close(s->gfp);
89 return 0;
90 }
91
mp3lame_encode_init(AVCodecContext *avctx)92 static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
93 {
94 LAMEContext *s = avctx->priv_data;
95 int ret;
96
97 s->avctx = avctx;
98
99 /* initialize LAME and get defaults */
100 if (!(s->gfp = lame_init()))
101 return AVERROR(ENOMEM);
102
103
104 lame_set_num_channels(s->gfp, avctx->ch_layout.nb_channels);
105 lame_set_mode(s->gfp, avctx->ch_layout.nb_channels > 1 ?
106 s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
107
108 /* sample rate */
109 lame_set_in_samplerate (s->gfp, avctx->sample_rate);
110 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
111
112 /* algorithmic quality */
113 if (avctx->compression_level != FF_COMPRESSION_DEFAULT)
114 lame_set_quality(s->gfp, avctx->compression_level);
115
116 /* rate control */
117 if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
118 lame_set_VBR(s->gfp, vbr_default);
119 lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
120 } else {
121 if (avctx->bit_rate) {
122 if (s->abr) { // ABR
123 lame_set_VBR(s->gfp, vbr_abr);
124 lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
125 } else // CBR
126 lame_set_brate(s->gfp, avctx->bit_rate / 1000);
127 }
128 }
129
130 /* lowpass cutoff frequency */
131 if (avctx->cutoff)
132 lame_set_lowpassfreq(s->gfp, avctx->cutoff);
133
134 /* do not get a Xing VBR header frame from LAME */
135 lame_set_bWriteVbrTag(s->gfp,0);
136
137 /* bit reservoir usage */
138 lame_set_disable_reservoir(s->gfp, !s->reservoir);
139
140 /* set specified parameters */
141 if (lame_init_params(s->gfp) < 0) {
142 ret = AVERROR_EXTERNAL;
143 goto error;
144 }
145
146 /* get encoder delay */
147 avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
148 ff_af_queue_init(avctx, &s->afq);
149
150 avctx->frame_size = lame_get_framesize(s->gfp);
151
152 /* allocate float sample buffers */
153 if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
154 int ch;
155 for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
156 s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
157 sizeof(*s->samples_flt[ch]));
158 if (!s->samples_flt[ch]) {
159 ret = AVERROR(ENOMEM);
160 goto error;
161 }
162 }
163 }
164
165 ret = realloc_buffer(s);
166 if (ret < 0)
167 goto error;
168
169 s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
170 if (!s->fdsp) {
171 ret = AVERROR(ENOMEM);
172 goto error;
173 }
174
175
176 return 0;
177 error:
178 mp3lame_encode_close(avctx);
179 return ret;
180 }
181
182 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
183 lame_result = func(s->gfp, \
184 (const buf_type *)buf_name[0], \
185 (const buf_type *)buf_name[1], frame->nb_samples, \
186 s->buffer + s->buffer_index, \
187 s->buffer_size - s->buffer_index); \
188 } while (0)
189
mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)190 static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
191 const AVFrame *frame, int *got_packet_ptr)
192 {
193 LAMEContext *s = avctx->priv_data;
194 MPADecodeHeader hdr;
195 int len, ret, ch, discard_padding;
196 int lame_result;
197 uint32_t h;
198
199 if (frame) {
200 switch (avctx->sample_fmt) {
201 case AV_SAMPLE_FMT_S16P:
202 ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
203 break;
204 case AV_SAMPLE_FMT_S32P:
205 ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
206 break;
207 case AV_SAMPLE_FMT_FLTP:
208 if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
209 av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
210 return AVERROR(EINVAL);
211 }
212 for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
213 s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
214 (const float *)frame->data[ch],
215 32768.0f,
216 FFALIGN(frame->nb_samples, 8));
217 }
218 ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
219 break;
220 default:
221 return AVERROR_BUG;
222 }
223 } else if (!s->afq.frame_alloc) {
224 lame_result = 0;
225 } else {
226 lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
227 s->buffer_size - s->buffer_index);
228 }
229 if (lame_result < 0) {
230 if (lame_result == -1) {
231 av_log(avctx, AV_LOG_ERROR,
232 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
233 s->buffer_index, s->buffer_size - s->buffer_index);
234 }
235 return AVERROR(ENOMEM);
236 }
237 s->buffer_index += lame_result;
238 ret = realloc_buffer(s);
239 if (ret < 0) {
240 av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
241 return ret;
242 }
243
244 /* add current frame to the queue */
245 if (frame) {
246 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
247 return ret;
248 }
249
250 /* Move 1 frame from the LAME buffer to the output packet, if available.
251 We have to parse the first frame header in the output buffer to
252 determine the frame size. */
253 if (s->buffer_index < 4)
254 return 0;
255 h = AV_RB32(s->buffer);
256
257 ret = avpriv_mpegaudio_decode_header(&hdr, h);
258 if (ret < 0) {
259 av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
260 return AVERROR_BUG;
261 } else if (ret) {
262 av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
263 return AVERROR_INVALIDDATA;
264 }
265 len = hdr.frame_size;
266 ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
267 s->buffer_index);
268 if (len <= s->buffer_index) {
269 if ((ret = ff_get_encode_buffer(avctx, avpkt, len, 0)) < 0)
270 return ret;
271 memcpy(avpkt->data, s->buffer, len);
272 s->buffer_index -= len;
273 memmove(s->buffer, s->buffer + len, s->buffer_index);
274
275 /* Get the next frame pts/duration */
276 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
277 &avpkt->duration);
278
279 discard_padding = avctx->frame_size - avpkt->duration;
280 // Check if subtraction resulted in an overflow
281 if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
282 av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
283 av_packet_unref(avpkt);
284 return AVERROR(EINVAL);
285 }
286 if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
287 uint8_t* side_data = av_packet_new_side_data(avpkt,
288 AV_PKT_DATA_SKIP_SAMPLES,
289 10);
290 if(!side_data) {
291 av_packet_unref(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 *got_packet_ptr = 1;
302 }
303 return 0;
304 }
305
306 #define OFFSET(x) offsetof(LAMEContext, x)
307 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
308 static const AVOption options[] = {
309 { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
310 { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
311 { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
312 { NULL },
313 };
314
315 static const AVClass libmp3lame_class = {
316 .class_name = "libmp3lame encoder",
317 .item_name = av_default_item_name,
318 .option = options,
319 .version = LIBAVUTIL_VERSION_INT,
320 };
321
322 static const FFCodecDefault libmp3lame_defaults[] = {
323 { "b", "0" },
324 { NULL },
325 };
326
327 static const int libmp3lame_sample_rates[] = {
328 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
329 };
330
331 const FFCodec ff_libmp3lame_encoder = {
332 .p.name = "libmp3lame",
333 .p.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
334 .p.type = AVMEDIA_TYPE_AUDIO,
335 .p.id = AV_CODEC_ID_MP3,
336 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
337 AV_CODEC_CAP_SMALL_LAST_FRAME,
338 .priv_data_size = sizeof(LAMEContext),
339 .init = mp3lame_encode_init,
340 FF_CODEC_ENCODE_CB(mp3lame_encode_frame),
341 .close = mp3lame_encode_close,
342 .p.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 .p.supported_samplerates = libmp3lame_sample_rates,
347 #if FF_API_OLD_CHANNEL_LAYOUT
348 .p.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
349 AV_CH_LAYOUT_STEREO,
350 0 },
351 #endif
352 .p.ch_layouts = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_MONO,
353 AV_CHANNEL_LAYOUT_STEREO,
354 { 0 },
355 },
356 .p.priv_class = &libmp3lame_class,
357 .defaults = libmp3lame_defaults,
358 .p.wrapper_name = "libmp3lame",
359 };
360