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]);
84 ff_af_queue_close(&s
->afq
);
90 static av_cold
int mp3lame_encode_init(AVCodecContext
*avctx
)
92 LAMEContext
*s
= avctx
->priv_data
;
97 /* initialize LAME and get defaults */
98 if (!(s
->gfp
= lame_init()))
99 return AVERROR(ENOMEM
);
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
);
106 lame_set_in_samplerate (s
->gfp
, avctx
->sample_rate
);
107 lame_set_out_samplerate(s
->gfp
, avctx
->sample_rate
);
109 /* algorithmic quality */
110 if (avctx
->compression_level
!= FF_COMPRESSION_DEFAULT
)
111 lame_set_quality(s
->gfp
, avctx
->compression_level
);
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
);
118 if (avctx
->bit_rate
) {
120 lame_set_VBR(s
->gfp
, vbr_abr
);
121 lame_set_VBR_mean_bitrate_kbps(s
->gfp
, avctx
->bit_rate
/ 1000);
123 lame_set_brate(s
->gfp
, avctx
->bit_rate
/ 1000);
127 /* do not get a Xing VBR header frame from LAME */
128 lame_set_bWriteVbrTag(s
->gfp
,0);
130 /* bit reservoir usage */
131 lame_set_disable_reservoir(s
->gfp
, !s
->reservoir
);
133 /* set specified parameters */
134 if (lame_init_params(s
->gfp
) < 0) {
139 /* get encoder delay */
140 avctx
->initial_padding
= lame_get_encoder_delay(s
->gfp
) + 528 + 1;
141 ff_af_queue_init(avctx
, &s
->afq
);
143 avctx
->frame_size
= lame_get_framesize(s
->gfp
);
145 /* allocate float sample buffers */
146 if (avctx
->sample_fmt
== AV_SAMPLE_FMT_FLTP
) {
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
);
158 ret
= realloc_buffer(s
);
162 s
->fdsp
= avpriv_float_dsp_alloc(avctx
->flags
& CODEC_FLAG_BITEXACT
);
164 ret
= AVERROR(ENOMEM
);
171 mp3lame_encode_close(avctx
);
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); \
183 static int mp3lame_encode_frame(AVCodecContext
*avctx
, AVPacket
*avpkt
,
184 const AVFrame
*frame
, int *got_packet_ptr
)
186 LAMEContext
*s
= avctx
->priv_data
;
193 switch (avctx
->sample_fmt
) {
194 case AV_SAMPLE_FMT_S16P
:
195 ENCODE_BUFFER(lame_encode_buffer
, int16_t, frame
->data
);
197 case AV_SAMPLE_FMT_S32P
:
198 ENCODE_BUFFER(lame_encode_buffer_int
, int32_t, frame
->data
);
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
);
205 for (ch
= 0; ch
< avctx
->channels
; ch
++) {
206 s
->fdsp
->vector_fmul_scalar(s
->samples_flt
[ch
],
207 (const float *)frame
->data
[ch
],
209 FFALIGN(frame
->nb_samples
, 8));
211 ENCODE_BUFFER(lame_encode_buffer_float
, float, s
->samples_flt
);
216 } else if (!s
->afq
.frame_alloc
) {
219 lame_result
= lame_encode_flush(s
->gfp
, s
->buffer
+ s
->buffer_index
,
220 s
->buffer_size
- s
->buffer_index
);
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
);
230 s
->buffer_index
+= lame_result
;
231 ret
= realloc_buffer(s
);
233 av_log(avctx
, AV_LOG_ERROR
, "error reallocating output buffer\n");
237 /* add current frame to the queue */
239 if ((ret
= ff_af_queue_add(&s
->afq
, frame
)) < 0)
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)
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");
253 if (avpriv_mpegaudio_decode_header(&hdr
, h
)) {
254 av_log(avctx
, AV_LOG_ERROR
, "free format output not supported\n");
257 len
= hdr
.frame_size
;
258 av_dlog(avctx
, "in:%d packet-len:%d index:%d\n", avctx
->frame_size
, len
,
260 if (len
<= s
->buffer_index
) {
261 if ((ret
= ff_alloc_packet2(avctx
, avpkt
, len
)) < 0)
263 memcpy(avpkt
->data
, s
->buffer
, len
);
264 s
->buffer_index
-= len
;
265 memmove(s
->buffer
, s
->buffer
+ len
, s
->buffer_index
);
267 /* Get the next frame pts/duration */
268 ff_af_queue_remove(&s
->afq
, avctx
->frame_size
, &avpkt
->pts
,
277 #define OFFSET(x) offsetof(LAMEContext, x)
278 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
279 static 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
},
286 static const AVClass libmp3lame_class
= {
287 .class_name
= "libmp3lame encoder",
288 .item_name
= av_default_item_name
,
290 .version
= LIBAVUTIL_VERSION_INT
,
293 static const AVCodecDefault libmp3lame_defaults
[] = {
298 static const int libmp3lame_sample_rates
[] = {
299 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
302 AVCodec 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
,
315 AV_SAMPLE_FMT_NONE
},
316 .supported_samplerates
= libmp3lame_sample_rates
,
317 .channel_layouts
= (const uint64_t[]) { AV_CH_LAYOUT_MONO
,
320 .priv_class
= &libmp3lame_class
,
321 .defaults
= libmp3lame_defaults
,