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
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_set_mode(s
->gfp
, s
->stereo ? JOINT_STEREO
: MONO
);
59 lame_set_brate(s
->gfp
, avctx
->bit_rate
/1000);
60 if(avctx
->flags
& CODEC_FLAG_QSCALE
) {
61 lame_set_brate(s
->gfp
, 0);
62 lame_set_VBR(s
->gfp
, vbr_default
);
63 lame_set_VBR_quality(s
->gfp
, avctx
->global_quality
/(float)FF_QP2LAMBDA
);
65 lame_set_bWriteVbrTag(s
->gfp
,0);
66 lame_set_disable_reservoir(s
->gfp
, avctx
->flags2
& CODEC_FLAG2_BIT_RESERVOIR ?
0 : 1);
67 if (lame_init_params(s
->gfp
) < 0)
70 avctx
->frame_size
= lame_get_framesize(s
->gfp
);
72 avctx
->coded_frame
= avcodec_alloc_frame();
73 avctx
->coded_frame
->key_frame
= 1;
83 static const int sSampleRates
[] = {
84 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
87 static const int sBitRates
[2][3][15] = {
88 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
89 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
90 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
92 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
93 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
94 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
98 static const int sSamplesPerFrame
[2][3] =
104 static const int sBitsPerSlot
[3] = {
110 static int mp3len(void *data
, int *samplesPerFrame
, int *sampleRate
)
112 uint32_t header
= AV_RB32(data
);
113 int layerID
= 3 - ((header
>> 17) & 0x03);
114 int bitRateID
= ((header
>> 12) & 0x0f);
115 int sampleRateID
= ((header
>> 10) & 0x03);
116 int bitsPerSlot
= sBitsPerSlot
[layerID
];
117 int isPadded
= ((header
>> 9) & 0x01);
118 static int const mode_tab
[4]= {2,3,1,0};
119 int mode
= mode_tab
[(header
>> 19) & 0x03];
121 int temp0
, temp1
, bitRate
;
123 if ( (( header
>> 21 ) & 0x7ff) != 0x7ff || mode
== 3 || layerID
==3 || sampleRateID
==3) {
127 if(!samplesPerFrame
) samplesPerFrame
= &temp0
;
128 if(!sampleRate
) sampleRate
= &temp1
;
130 // *isMono = ((header >> 6) & 0x03) == 0x03;
132 *sampleRate
= sSampleRates
[sampleRateID
]>>mode
;
133 bitRate
= sBitRates
[mpeg_id
][layerID
][bitRateID
] * 1000;
134 *samplesPerFrame
= sSamplesPerFrame
[mpeg_id
][layerID
];
135 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
137 return *samplesPerFrame
* bitRate
/ (bitsPerSlot
* *sampleRate
) + isPadded
;
140 static int MP3lame_encode_frame(AVCodecContext
*avctx
,
141 unsigned char *frame
, int buf_size
, void *data
)
143 Mp3AudioContext
*s
= avctx
->priv_data
;
147 /* lame 3.91 dies on '1-channel interleaved' data */
151 lame_result
= lame_encode_buffer_interleaved(
155 s
->buffer
+ s
->buffer_index
,
156 BUFFER_SIZE
- s
->buffer_index
159 lame_result
= lame_encode_buffer(
164 s
->buffer
+ s
->buffer_index
,
165 BUFFER_SIZE
- s
->buffer_index
169 lame_result
= lame_encode_flush(
171 s
->buffer
+ s
->buffer_index
,
172 BUFFER_SIZE
- s
->buffer_index
177 if(lame_result
==-1) {
178 /* output buffer too small */
179 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
);
184 s
->buffer_index
+= lame_result
;
186 if(s
->buffer_index
<4)
189 len
= mp3len(s
->buffer
, NULL
, NULL
);
190 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
191 if(len
<= s
->buffer_index
){
192 memcpy(frame
, s
->buffer
, len
);
193 s
->buffer_index
-= len
;
195 memmove(s
->buffer
, s
->buffer
+len
, s
->buffer_index
);
196 //FIXME fix the audio codec API, so we do not need the memcpy()
197 /*for(i=0; i<len; i++){
198 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
205 static av_cold
int MP3lame_encode_close(AVCodecContext
*avctx
)
207 Mp3AudioContext
*s
= avctx
->priv_data
;
209 av_freep(&avctx
->coded_frame
);
216 AVCodec libmp3lame_encoder
= {
220 sizeof(Mp3AudioContext
),
222 MP3lame_encode_frame
,
223 MP3lame_encode_close
,
224 .capabilities
= CODEC_CAP_DELAY
,
225 .sample_fmts
= (const enum AVSampleFormat
[]){AV_SAMPLE_FMT_S16
,AV_SAMPLE_FMT_NONE
},
226 .supported_samplerates
= sSampleRates
,
227 .long_name
= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),