2 * Interface to libmp3lame for mp3 encoding
3 * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Lesser General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
15 * You should have received a copy of the GNU Lesser General Public
16 * License along with this library; if not, write to the Free Software
17 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 * @file mp3lameaudio.c
22 * Interface to libmp3lame for mp3 encoding.
26 #include "mpegaudio.h"
27 #include <lame/lame.h>
29 #define BUFFER_SIZE (2*MPA_FRAME_SIZE)
30 typedef struct Mp3AudioContext
{
31 lame_global_flags
*gfp
;
33 uint8_t buffer
[BUFFER_SIZE
];
37 static int MP3lame_encode_init(AVCodecContext
*avctx
)
39 Mp3AudioContext
*s
= avctx
->priv_data
;
41 if (avctx
->channels
> 2)
44 s
->stereo
= avctx
->channels
> 1 ? 1 : 0;
46 if ((s
->gfp
= lame_init()) == NULL
)
48 lame_set_in_samplerate(s
->gfp
, avctx
->sample_rate
);
49 lame_set_out_samplerate(s
->gfp
, avctx
->sample_rate
);
50 lame_set_num_channels(s
->gfp
, avctx
->channels
);
51 /* lame 3.91 dies on quality != 5 */
52 lame_set_quality(s
->gfp
, 5);
53 /* lame 3.91 doesn't work in mono */
54 lame_set_mode(s
->gfp
, JOINT_STEREO
);
55 lame_set_brate(s
->gfp
, avctx
->bit_rate
/1000);
56 if(avctx
->flags
& CODEC_FLAG_QSCALE
) {
57 lame_set_brate(s
->gfp
, 0);
58 lame_set_VBR(s
->gfp
, vbr_default
);
59 lame_set_VBR_q(s
->gfp
, avctx
->global_quality
/ (float)FF_QP2LAMBDA
);
61 lame_set_bWriteVbrTag(s
->gfp
,0);
62 if (lame_init_params(s
->gfp
) < 0)
65 avctx
->frame_size
= lame_get_framesize(s
->gfp
);
67 avctx
->coded_frame
= avcodec_alloc_frame();
68 avctx
->coded_frame
->key_frame
= 1;
78 static const int sSampleRates
[3] = {
82 static const int sBitRates
[2][3][15] = {
83 { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
84 { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
85 { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
87 { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
88 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
89 { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
93 static const int sSamplesPerFrame
[2][3] =
99 static const int sBitsPerSlot
[3] = {
105 static int mp3len(void *data
, int *samplesPerFrame
, int *sampleRate
)
107 uint8_t *dataTmp
= (uint8_t *)data
;
108 uint32_t header
= ( (uint32_t)dataTmp
[0] << 24 ) | ( (uint32_t)dataTmp
[1] << 16 ) | ( (uint32_t)dataTmp
[2] << 8 ) | (uint32_t)dataTmp
[3];
109 int layerID
= 3 - ((header
>> 17) & 0x03);
110 int bitRateID
= ((header
>> 12) & 0x0f);
111 int sampleRateID
= ((header
>> 10) & 0x03);
112 int bitsPerSlot
= sBitsPerSlot
[layerID
];
113 int isPadded
= ((header
>> 9) & 0x01);
114 static int const mode_tab
[4]= {2,3,1,0};
115 int mode
= mode_tab
[(header
>> 19) & 0x03];
117 int temp0
, temp1
, bitRate
;
119 if ( (( header
>> 21 ) & 0x7ff) != 0x7ff || mode
== 3 || layerID
==3 || sampleRateID
==3) {
123 if(!samplesPerFrame
) samplesPerFrame
= &temp0
;
124 if(!sampleRate
) sampleRate
= &temp1
;
126 // *isMono = ((header >> 6) & 0x03) == 0x03;
128 *sampleRate
= sSampleRates
[sampleRateID
]>>mode
;
129 bitRate
= sBitRates
[mpeg_id
][layerID
][bitRateID
] * 1000;
130 *samplesPerFrame
= sSamplesPerFrame
[mpeg_id
][layerID
];
131 //av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
133 return *samplesPerFrame
* bitRate
/ (bitsPerSlot
* *sampleRate
) + isPadded
;
136 int MP3lame_encode_frame(AVCodecContext
*avctx
,
137 unsigned char *frame
, int buf_size
, void *data
)
139 Mp3AudioContext
*s
= avctx
->priv_data
;
143 /* lame 3.91 dies on '1-channel interleaved' data */
147 lame_result
= lame_encode_buffer_interleaved(
151 s
->buffer
+ s
->buffer_index
,
152 BUFFER_SIZE
- s
->buffer_index
155 lame_result
= lame_encode_buffer(
160 s
->buffer
+ s
->buffer_index
,
161 BUFFER_SIZE
- s
->buffer_index
165 lame_result
= lame_encode_flush(
167 s
->buffer
+ s
->buffer_index
,
168 BUFFER_SIZE
- s
->buffer_index
172 if(lame_result
==-1) {
173 /* output buffer too small */
174 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
);
178 s
->buffer_index
+= lame_result
;
180 if(s
->buffer_index
<4)
183 len
= mp3len(s
->buffer
, NULL
, NULL
);
184 //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
185 if(len
<= s
->buffer_index
){
186 memcpy(frame
, s
->buffer
, len
);
187 s
->buffer_index
-= len
;
189 memmove(s
->buffer
, s
->buffer
+len
, s
->buffer_index
);
190 //FIXME fix the audio codec API, so we dont need the memcpy()
191 /*for(i=0; i<len; i++){
192 av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
199 int MP3lame_encode_close(AVCodecContext
*avctx
)
201 Mp3AudioContext
*s
= avctx
->priv_data
;
203 av_freep(&avctx
->coded_frame
);
210 AVCodec mp3lame_encoder
= {
214 sizeof(Mp3AudioContext
),
216 MP3lame_encode_frame
,
217 MP3lame_encode_close
,
218 .capabilities
= CODEC_CAP_DELAY
,