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,
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.
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
23 * @file libavcodec/libmp3lame.c
24 * Interface to libmp3lame for mp3 encoding.
28 #include "mpegaudio.h"
29 #include <lame/lame.h>
31 #define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
32 typedef struct Mp3AudioContext
{
33 lame_global_flags
*gfp
;
35 uint8_t buffer
[BUFFER_SIZE
];
39 static av_cold
int MP3lame_encode_init(AVCodecContext
*avctx
)
41 Mp3AudioContext
*s
= avctx
->priv_data
;
43 if (avctx
->channels
> 2)
46 s
->stereo
= avctx
->channels
> 1 ? 1 : 0;
48 if ((s
->gfp
= lame_init()) == NULL
)
50 lame_set_in_samplerate(s
->gfp
, avctx
->sample_rate
);
51 lame_set_out_samplerate(s
->gfp
, avctx
->sample_rate
);
52 lame_set_num_channels(s
->gfp
, avctx
->channels
);
53 if(avctx
->compression_level
== FF_COMPRESSION_DEFAULT
) {
54 lame_set_quality(s
->gfp
, 5);
56 lame_set_quality(s
->gfp
, avctx
->compression_level
);
58 /* lame 3.91 doesn't work in mono */
59 lame_set_mode(s
->gfp
, JOINT_STEREO
);
60 lame_set_brate(s
->gfp
, avctx
->bit_rate
/1000);
61 if(avctx
->flags
& CODEC_FLAG_QSCALE
) {
62 lame_set_brate(s
->gfp
, 0);
63 lame_set_VBR(s
->gfp
, vbr_default
);
64 lame_set_VBR_q(s
->gfp
, avctx
->global_quality
/ (float)FF_QP2LAMBDA
);
66 lame_set_bWriteVbrTag(s
->gfp
,0);
67 lame_set_disable_reservoir(s
->gfp
, avctx
->flags2
& CODEC_FLAG2_BIT_RESERVOIR
? 0 : 1);
68 if (lame_init_params(s
->gfp
) < 0)
71 avctx
->frame_size
= lame_get_framesize(s
->gfp
);
73 avctx
->coded_frame
= avcodec_alloc_frame();
74 avctx
->coded_frame
->key_frame
= 1;
84 static const int sSampleRates
[3] = {
88 static const int sBitRates
[2][3][15] = {
89 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
90 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
91 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
93 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
94 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
95 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
99 static const int sSamplesPerFrame
[2][3] =
105 static const int sBitsPerSlot
[3] = {
111 static int mp3len(void *data
, int *samplesPerFrame
, int *sampleRate
)
113 uint32_t header
= AV_RB32(data
);
114 int layerID
= 3 - ((header
>> 17) & 0x03);
115 int bitRateID
= ((header
>> 12) & 0x0f);
116 int sampleRateID
= ((header
>> 10) & 0x03);
117 int bitsPerSlot
= sBitsPerSlot
[layerID
];
118 int isPadded
= ((header
>> 9) & 0x01);
119 static int const mode_tab
[4]= {2,3,1,0};
120 int mode
= mode_tab
[(header
>> 19) & 0x03];
122 int temp0
, temp1
, bitRate
;
124 if ( (( header
>> 21 ) & 0x7ff) != 0x7ff || mode
== 3 || layerID
==3 || sampleRateID
==3) {
128 if(!samplesPerFrame
) samplesPerFrame
= &temp0
;
129 if(!sampleRate
) sampleRate
= &temp1
;
131 // *isMono = ((header >> 6) & 0x03) == 0x03;
133 *sampleRate
= sSampleRates
[sampleRateID
]>>mode
;
134 bitRate
= sBitRates
[mpeg_id
][layerID
][bitRateID
] * 1000;
135 *samplesPerFrame
= sSamplesPerFrame
[mpeg_id
][layerID
];
136 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
138 return *samplesPerFrame
* bitRate
/ (bitsPerSlot
* *sampleRate
) + isPadded
;
141 static int MP3lame_encode_frame(AVCodecContext
*avctx
,
142 unsigned char *frame
, int buf_size
, void *data
)
144 Mp3AudioContext
*s
= avctx
->priv_data
;
148 /* lame 3.91 dies on '1-channel interleaved' data */
152 lame_result
= lame_encode_buffer_interleaved(
156 s
->buffer
+ s
->buffer_index
,
157 BUFFER_SIZE
- s
->buffer_index
160 lame_result
= lame_encode_buffer(
165 s
->buffer
+ s
->buffer_index
,
166 BUFFER_SIZE
- s
->buffer_index
170 lame_result
= lame_encode_flush(
172 s
->buffer
+ s
->buffer_index
,
173 BUFFER_SIZE
- s
->buffer_index
178 if(lame_result
==-1) {
179 /* output buffer too small */
180 av_log(avctx
, AV_LOG_ERROR
, "lame: output buffer too small (buffer index: %d, free bytes: %d)\n", s
->buffer_index
, BUFFER_SIZE
- s
->buffer_index
);
185 s
->buffer_index
+= lame_result
;
187 if(s
->buffer_index
<4)
190 len
= mp3len(s
->buffer
, NULL
, NULL
);
191 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
192 if(len
<= s
->buffer_index
){
193 memcpy(frame
, s
->buffer
, len
);
194 s
->buffer_index
-= len
;
196 memmove(s
->buffer
, s
->buffer
+len
, s
->buffer_index
);
197 //FIXME fix the audio codec API, so we do not need the memcpy()
198 /*for(i=0; i<len; i++){
199 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
206 static av_cold
int MP3lame_encode_close(AVCodecContext
*avctx
)
208 Mp3AudioContext
*s
= avctx
->priv_data
;
210 av_freep(&avctx
->coded_frame
);
217 AVCodec libmp3lame_encoder
= {
221 sizeof(Mp3AudioContext
),
223 MP3lame_encode_frame
,
224 MP3lame_encode_close
,
225 .capabilities
= CODEC_CAP_DELAY
,
226 .sample_fmts
= (const enum SampleFormat
[]){SAMPLE_FMT_S16
,SAMPLE_FMT_NONE
},
227 .long_name
= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),