2 * Interface to libmp3lame for mp3 encoding
3 * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
5 * This file is part of FFmpeg.
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.
12 * FFmpeg is distributed in the hope that it will be useful,
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
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
24 * Interface to libmp3lame for mp3 encoding.
27 #include <lame/lame.h>
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"
36 #include "audio_frame_queue.h"
38 #include "mpegaudio.h"
39 #include "mpegaudiodecheader.h"
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.
43 typedef struct LAMEContext
{
45 AVCodecContext
*avctx
;
46 lame_global_flags
*gfp
;
53 float *samples_flt
[2];
55 AVFloatDSPContext fdsp
;
59 static int realloc_buffer(LAMEContext
*s
)
61 if (!s
->buffer
|| s
->buffer_size
- s
->buffer_index
< BUFFER_SIZE
) {
62 int new_size
= s
->buffer_index
+ 2 * BUFFER_SIZE
, err
;
64 av_dlog(s
->avctx
, "resizing output buffer: %d -> %d\n", s
->buffer_size
,
66 if ((err
= av_reallocp(&s
->buffer
, new_size
)) < 0) {
67 s
->buffer_size
= s
->buffer_index
= 0;
70 s
->buffer_size
= new_size
;
75 static av_cold
int mp3lame_encode_close(AVCodecContext
*avctx
)
77 LAMEContext
*s
= avctx
->priv_data
;
79 av_freep(&s
->samples_flt
[0]);
80 av_freep(&s
->samples_flt
[1]);
83 ff_af_queue_close(&s
->afq
);
89 static av_cold
int mp3lame_encode_init(AVCodecContext
*avctx
)
91 LAMEContext
*s
= avctx
->priv_data
;
96 /* initialize LAME and get defaults */
97 if (!(s
->gfp
= lame_init()))
98 return AVERROR(ENOMEM
);
101 lame_set_num_channels(s
->gfp
, avctx
->channels
);
102 lame_set_mode(s
->gfp
, avctx
->channels
> 1 ? s
->joint_stereo
? JOINT_STEREO
: STEREO
: MONO
);
105 lame_set_in_samplerate (s
->gfp
, avctx
->sample_rate
);
106 lame_set_out_samplerate(s
->gfp
, avctx
->sample_rate
);
108 /* algorithmic quality */
109 if (avctx
->compression_level
== FF_COMPRESSION_DEFAULT
)
110 lame_set_quality(s
->gfp
, 5);
112 lame_set_quality(s
->gfp
, avctx
->compression_level
);
115 if (avctx
->flags
& 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
);
119 if (avctx
->bit_rate
) {
121 lame_set_VBR(s
->gfp
, vbr_abr
);
122 lame_set_VBR_mean_bitrate_kbps(s
->gfp
, avctx
->bit_rate
/ 1000);
124 lame_set_brate(s
->gfp
, avctx
->bit_rate
/ 1000);
128 /* do not get a Xing VBR header frame from LAME */
129 lame_set_bWriteVbrTag(s
->gfp
,0);
131 /* bit reservoir usage */
132 lame_set_disable_reservoir(s
->gfp
, !s
->reservoir
);
134 /* set specified parameters */
135 if (lame_init_params(s
->gfp
) < 0) {
140 /* get encoder delay */
141 avctx
->delay
= lame_get_encoder_delay(s
->gfp
) + 528 + 1;
142 ff_af_queue_init(avctx
, &s
->afq
);
144 avctx
->frame_size
= lame_get_framesize(s
->gfp
);
146 /* allocate float sample buffers */
147 if (avctx
->sample_fmt
== AV_SAMPLE_FMT_FLTP
) {
149 for (ch
= 0; ch
< avctx
->channels
; ch
++) {
150 s
->samples_flt
[ch
] = av_malloc(avctx
->frame_size
*
151 sizeof(*s
->samples_flt
[ch
]));
152 if (!s
->samples_flt
[ch
]) {
153 ret
= AVERROR(ENOMEM
);
159 ret
= realloc_buffer(s
);
163 avpriv_float_dsp_init(&s
->fdsp
, avctx
->flags
& CODEC_FLAG_BITEXACT
);
167 mp3lame_encode_close(avctx
);
171 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
172 lame_result = func(s->gfp, \
173 (const buf_type *)buf_name[0], \
174 (const buf_type *)buf_name[1], frame->nb_samples, \
175 s->buffer + s->buffer_index, \
176 s->buffer_size - s->buffer_index); \
179 static int mp3lame_encode_frame(AVCodecContext
*avctx
, AVPacket
*avpkt
,
180 const AVFrame
*frame
, int *got_packet_ptr
)
182 LAMEContext
*s
= avctx
->priv_data
;
189 switch (avctx
->sample_fmt
) {
190 case AV_SAMPLE_FMT_S16P
:
191 ENCODE_BUFFER(lame_encode_buffer
, int16_t, frame
->data
);
193 case AV_SAMPLE_FMT_S32P
:
194 ENCODE_BUFFER(lame_encode_buffer_int
, int32_t, frame
->data
);
196 case AV_SAMPLE_FMT_FLTP
:
197 if (frame
->linesize
[0] < 4 * FFALIGN(frame
->nb_samples
, 8)) {
198 av_log(avctx
, AV_LOG_ERROR
, "inadequate AVFrame plane padding\n");
199 return AVERROR(EINVAL
);
201 for (ch
= 0; ch
< avctx
->channels
; ch
++) {
202 s
->fdsp
.vector_fmul_scalar(s
->samples_flt
[ch
],
203 (const float *)frame
->data
[ch
],
205 FFALIGN(frame
->nb_samples
, 8));
207 ENCODE_BUFFER(lame_encode_buffer_float
, float, s
->samples_flt
);
212 } else if (!s
->afq
.frame_alloc
) {
215 lame_result
= lame_encode_flush(s
->gfp
, s
->buffer
+ s
->buffer_index
,
216 s
->buffer_size
- s
->buffer_index
);
218 if (lame_result
< 0) {
219 if (lame_result
== -1) {
220 av_log(avctx
, AV_LOG_ERROR
,
221 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
222 s
->buffer_index
, s
->buffer_size
- s
->buffer_index
);
226 s
->buffer_index
+= lame_result
;
227 ret
= realloc_buffer(s
);
229 av_log(avctx
, AV_LOG_ERROR
, "error reallocating output buffer\n");
233 /* add current frame to the queue */
235 if ((ret
= ff_af_queue_add(&s
->afq
, frame
)) < 0)
239 /* Move 1 frame from the LAME buffer to the output packet, if available.
240 We have to parse the first frame header in the output buffer to
241 determine the frame size. */
242 if (s
->buffer_index
< 4)
244 h
= AV_RB32(s
->buffer
);
245 if (ff_mpa_check_header(h
) < 0) {
246 av_log(avctx
, AV_LOG_ERROR
, "Invalid mp3 header at start of buffer\n");
249 if (avpriv_mpegaudio_decode_header(&hdr
, h
)) {
250 av_log(avctx
, AV_LOG_ERROR
, "free format output not supported\n");
253 len
= hdr
.frame_size
;
254 av_dlog(avctx
, "in:%d packet-len:%d index:%d\n", avctx
->frame_size
, len
,
256 if (len
<= s
->buffer_index
) {
257 if ((ret
= ff_alloc_packet2(avctx
, avpkt
, len
)) < 0)
259 memcpy(avpkt
->data
, s
->buffer
, len
);
260 s
->buffer_index
-= len
;
261 memmove(s
->buffer
, s
->buffer
+ len
, s
->buffer_index
);
263 /* Get the next frame pts/duration */
264 ff_af_queue_remove(&s
->afq
, avctx
->frame_size
, &avpkt
->pts
,
273 #define OFFSET(x) offsetof(LAMEContext, x)
274 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
275 static const AVOption options
[] = {
276 { "reservoir", "use bit reservoir", OFFSET(reservoir
), AV_OPT_TYPE_INT
, { .i64
= 1 }, 0, 1, AE
},
277 { "joint_stereo", "use joint stereo", OFFSET(joint_stereo
), AV_OPT_TYPE_INT
, { .i64
= 1 }, 0, 1, AE
},
278 { "abr", "use ABR", OFFSET(abr
), AV_OPT_TYPE_INT
, { .i64
= 0 }, 0, 1, AE
},
282 static const AVClass libmp3lame_class
= {
283 .class_name
= "libmp3lame encoder",
284 .item_name
= av_default_item_name
,
286 .version
= LIBAVUTIL_VERSION_INT
,
289 static const AVCodecDefault libmp3lame_defaults
[] = {
294 static const int libmp3lame_sample_rates
[] = {
295 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
298 AVCodec ff_libmp3lame_encoder
= {
299 .name
= "libmp3lame",
300 .long_name
= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
301 .type
= AVMEDIA_TYPE_AUDIO
,
302 .id
= AV_CODEC_ID_MP3
,
303 .priv_data_size
= sizeof(LAMEContext
),
304 .init
= mp3lame_encode_init
,
305 .encode2
= mp3lame_encode_frame
,
306 .close
= mp3lame_encode_close
,
307 .capabilities
= CODEC_CAP_DELAY
| CODEC_CAP_SMALL_LAST_FRAME
,
308 .sample_fmts
= (const enum AVSampleFormat
[]) { AV_SAMPLE_FMT_S32P
,
311 AV_SAMPLE_FMT_NONE
},
312 .supported_samplerates
= libmp3lame_sample_rates
,
313 .channel_layouts
= (const uint64_t[]) { AV_CH_LAYOUT_MONO
,
316 .priv_class
= &libmp3lame_class
,
317 .defaults
= libmp3lame_defaults
,