Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / libmp3lame.c
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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 float *samples_flt[2];
54 AudioFrameQueue afq;
f6fa7814 55 AVFloatDSPContext *fdsp;
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56} LAMEContext;
57
58
59static int realloc_buffer(LAMEContext *s)
60{
61 if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
62 int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
63
64 av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
65 new_size);
66 if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
67 s->buffer_size = s->buffer_index = 0;
68 return err;
69 }
70 s->buffer_size = new_size;
71 }
72 return 0;
73}
74
75static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
76{
77 LAMEContext *s = avctx->priv_data;
78
79 av_freep(&s->samples_flt[0]);
80 av_freep(&s->samples_flt[1]);
81 av_freep(&s->buffer);
f6fa7814 82 av_freep(&s->fdsp);
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83
84 ff_af_queue_close(&s->afq);
85
86 lame_close(s->gfp);
87 return 0;
88}
89
90static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
91{
92 LAMEContext *s = avctx->priv_data;
93 int ret;
94
95 s->avctx = avctx;
96
97 /* initialize LAME and get defaults */
98 if (!(s->gfp = lame_init()))
99 return AVERROR(ENOMEM);
100
101
102 lame_set_num_channels(s->gfp, avctx->channels);
103 lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
104
105 /* sample rate */
106 lame_set_in_samplerate (s->gfp, avctx->sample_rate);
107 lame_set_out_samplerate(s->gfp, avctx->sample_rate);
108
109 /* algorithmic quality */
f6fa7814 110 if (avctx->compression_level != FF_COMPRESSION_DEFAULT)
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111 lame_set_quality(s->gfp, avctx->compression_level);
112
113 /* rate control */
114 if (avctx->flags & CODEC_FLAG_QSCALE) { // VBR
115 lame_set_VBR(s->gfp, vbr_default);
116 lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
117 } else {
118 if (avctx->bit_rate) {
119 if (s->abr) { // ABR
120 lame_set_VBR(s->gfp, vbr_abr);
121 lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
122 } else // CBR
123 lame_set_brate(s->gfp, avctx->bit_rate / 1000);
124 }
125 }
126
127 /* do not get a Xing VBR header frame from LAME */
128 lame_set_bWriteVbrTag(s->gfp,0);
129
130 /* bit reservoir usage */
131 lame_set_disable_reservoir(s->gfp, !s->reservoir);
132
133 /* set specified parameters */
134 if (lame_init_params(s->gfp) < 0) {
135 ret = -1;
136 goto error;
137 }
138
139 /* get encoder delay */
f6fa7814 140 avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
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141 ff_af_queue_init(avctx, &s->afq);
142
143 avctx->frame_size = lame_get_framesize(s->gfp);
144
145 /* allocate float sample buffers */
146 if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
147 int ch;
148 for (ch = 0; ch < avctx->channels; ch++) {
149 s->samples_flt[ch] = av_malloc(avctx->frame_size *
150 sizeof(*s->samples_flt[ch]));
151 if (!s->samples_flt[ch]) {
152 ret = AVERROR(ENOMEM);
153 goto error;
154 }
155 }
156 }
157
158 ret = realloc_buffer(s);
159 if (ret < 0)
160 goto error;
161
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162 s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
163 if (!s->fdsp) {
164 ret = AVERROR(ENOMEM);
165 goto error;
166 }
167
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168
169 return 0;
170error:
171 mp3lame_encode_close(avctx);
172 return ret;
173}
174
175#define ENCODE_BUFFER(func, buf_type, buf_name) do { \
176 lame_result = func(s->gfp, \
177 (const buf_type *)buf_name[0], \
178 (const buf_type *)buf_name[1], frame->nb_samples, \
179 s->buffer + s->buffer_index, \
180 s->buffer_size - s->buffer_index); \
181} while (0)
182
183static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
184 const AVFrame *frame, int *got_packet_ptr)
185{
186 LAMEContext *s = avctx->priv_data;
187 MPADecodeHeader hdr;
188 int len, ret, ch;
189 int lame_result;
190 uint32_t h;
191
192 if (frame) {
193 switch (avctx->sample_fmt) {
194 case AV_SAMPLE_FMT_S16P:
195 ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
196 break;
197 case AV_SAMPLE_FMT_S32P:
198 ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
199 break;
200 case AV_SAMPLE_FMT_FLTP:
201 if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
202 av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
203 return AVERROR(EINVAL);
204 }
205 for (ch = 0; ch < avctx->channels; ch++) {
f6fa7814 206 s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
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207 (const float *)frame->data[ch],
208 32768.0f,
209 FFALIGN(frame->nb_samples, 8));
210 }
211 ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
212 break;
213 default:
214 return AVERROR_BUG;
215 }
216 } else if (!s->afq.frame_alloc) {
217 lame_result = 0;
218 } else {
219 lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
220 s->buffer_size - s->buffer_index);
221 }
222 if (lame_result < 0) {
223 if (lame_result == -1) {
224 av_log(avctx, AV_LOG_ERROR,
225 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
226 s->buffer_index, s->buffer_size - s->buffer_index);
227 }
228 return -1;
229 }
230 s->buffer_index += lame_result;
231 ret = realloc_buffer(s);
232 if (ret < 0) {
233 av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
234 return ret;
235 }
236
237 /* add current frame to the queue */
238 if (frame) {
239 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
240 return ret;
241 }
242
243 /* Move 1 frame from the LAME buffer to the output packet, if available.
244 We have to parse the first frame header in the output buffer to
245 determine the frame size. */
246 if (s->buffer_index < 4)
247 return 0;
248 h = AV_RB32(s->buffer);
249 if (ff_mpa_check_header(h) < 0) {
250 av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
251 return AVERROR_BUG;
252 }
253 if (avpriv_mpegaudio_decode_header(&hdr, h)) {
254 av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
255 return -1;
256 }
257 len = hdr.frame_size;
258 av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
259 s->buffer_index);
260 if (len <= s->buffer_index) {
261 if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0)
262 return ret;
263 memcpy(avpkt->data, s->buffer, len);
264 s->buffer_index -= len;
265 memmove(s->buffer, s->buffer + len, s->buffer_index);
266
267 /* Get the next frame pts/duration */
268 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
269 &avpkt->duration);
270
271 avpkt->size = len;
272 *got_packet_ptr = 1;
273 }
274 return 0;
275}
276
277#define OFFSET(x) offsetof(LAMEContext, x)
278#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
279static const AVOption options[] = {
280 { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
281 { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
282 { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
283 { NULL },
284};
285
286static const AVClass libmp3lame_class = {
287 .class_name = "libmp3lame encoder",
288 .item_name = av_default_item_name,
289 .option = options,
290 .version = LIBAVUTIL_VERSION_INT,
291};
292
293static const AVCodecDefault libmp3lame_defaults[] = {
294 { "b", "0" },
295 { NULL },
296};
297
298static const int libmp3lame_sample_rates[] = {
299 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
300};
301
302AVCodec ff_libmp3lame_encoder = {
303 .name = "libmp3lame",
304 .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
305 .type = AVMEDIA_TYPE_AUDIO,
306 .id = AV_CODEC_ID_MP3,
307 .priv_data_size = sizeof(LAMEContext),
308 .init = mp3lame_encode_init,
309 .encode2 = mp3lame_encode_frame,
310 .close = mp3lame_encode_close,
311 .capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
312 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
313 AV_SAMPLE_FMT_FLTP,
314 AV_SAMPLE_FMT_S16P,
315 AV_SAMPLE_FMT_NONE },
316 .supported_samplerates = libmp3lame_sample_rates,
317 .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
318 AV_CH_LAYOUT_STEREO,
319 0 },
320 .priv_class = &libmp3lame_class,
321 .defaults = libmp3lame_defaults,
322};