1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
|
/*
* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/**
* @file mp3lameaudio.c
* Interface to libmp3lame for mp3 encoding.
*/
#include "avcodec.h"
#include "mpegaudio.h"
#include <lame/lame.h>
#define BUFFER_SIZE (2*MPA_FRAME_SIZE)
typedef struct Mp3AudioContext {
lame_global_flags *gfp;
int stereo;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
} Mp3AudioContext;
static int MP3lame_encode_init(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
if (avctx->channels > 2)
return -1;
s->stereo = avctx->channels > 1 ? 1 : 0;
if ((s->gfp = lame_init()) == NULL)
goto err;
lame_set_in_samplerate(s->gfp, avctx->sample_rate);
lame_set_out_samplerate(s->gfp, avctx->sample_rate);
lame_set_num_channels(s->gfp, avctx->channels);
/* lame 3.91 dies on quality != 5 */
lame_set_quality(s->gfp, 5);
/* lame 3.91 doesn't work in mono */
lame_set_mode(s->gfp, JOINT_STEREO);
lame_set_brate(s->gfp, avctx->bit_rate/1000);
if(avctx->flags & CODEC_FLAG_QSCALE) {
lame_set_brate(s->gfp, 0);
lame_set_VBR(s->gfp, vbr_default);
lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
}
lame_set_bWriteVbrTag(s->gfp,0);
if (lame_init_params(s->gfp) < 0)
goto err_close;
avctx->frame_size = lame_get_framesize(s->gfp);
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
err_close:
lame_close(s->gfp);
err:
return -1;
}
static const int sSampleRates[3] = {
44100, 48000, 32000
};
static const int sBitRates[2][3][15] = {
{ { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
{ 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
{ 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
},
{ { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
},
};
static const int sSamplesPerFrame[2][3] =
{
{ 384, 1152, 1152 },
{ 384, 1152, 576 }
};
static const int sBitsPerSlot[3] = {
32,
8,
8
};
static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
{
uint8_t *dataTmp = (uint8_t *)data;
uint32_t header = ( (uint32_t)dataTmp[0] << 24 ) | ( (uint32_t)dataTmp[1] << 16 ) | ( (uint32_t)dataTmp[2] << 8 ) | (uint32_t)dataTmp[3];
int layerID = 3 - ((header >> 17) & 0x03);
int bitRateID = ((header >> 12) & 0x0f);
int sampleRateID = ((header >> 10) & 0x03);
int bitsPerSlot = sBitsPerSlot[layerID];
int isPadded = ((header >> 9) & 0x01);
static int const mode_tab[4]= {2,3,1,0};
int mode= mode_tab[(header >> 19) & 0x03];
int mpeg_id= mode>0;
int temp0, temp1, bitRate;
if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
return -1;
}
if(!samplesPerFrame) samplesPerFrame= &temp0;
if(!sampleRate ) sampleRate = &temp1;
// *isMono = ((header >> 6) & 0x03) == 0x03;
*sampleRate = sSampleRates[sampleRateID]>>mode;
bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
}
int MP3lame_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
Mp3AudioContext *s = avctx->priv_data;
int len;
int lame_result;
/* lame 3.91 dies on '1-channel interleaved' data */
if(data){
if (s->stereo) {
lame_result = lame_encode_buffer_interleaved(
s->gfp,
data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
} else {
lame_result = lame_encode_buffer(
s->gfp,
data,
data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
}
}else{
lame_result= lame_encode_flush(
s->gfp,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
}
if(lame_result==-1) {
/* output buffer too small */
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);
return 0;
}
s->buffer_index += lame_result;
if(s->buffer_index<4)
return 0;
len= mp3len(s->buffer, NULL, NULL);
//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
if(len <= s->buffer_index){
memcpy(frame, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer+len, s->buffer_index);
//FIXME fix the audio codec API, so we dont need the memcpy()
/*for(i=0; i<len; i++){
av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
}*/
return len;
}else
return 0;
}
int MP3lame_encode_close(AVCodecContext *avctx)
{
Mp3AudioContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame);
lame_close(s->gfp);
return 0;
}
AVCodec mp3lame_encoder = {
"mp3",
CODEC_TYPE_AUDIO,
CODEC_ID_MP3,
sizeof(Mp3AudioContext),
MP3lame_encode_init,
MP3lame_encode_frame,
MP3lame_encode_close,
.capabilities= CODEC_CAP_DELAY,
};
|
