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
|
------------------------------------------------------------------------
r35228 | cigaes | 2012-10-04 15:04:42 -0300 (Thu, 04 Oct 2012) | 5 lines
ad_ffmpeg: basic support for planar formats.
Upgrade to avcodec_decode_audio4().
Planar formats are immediately converted to packet formats.
A lot of optimizations are still possible.
------------------------------------------------------------------------
Index: libmpcodecs/ad_ffmpeg.c
===================================================================
--- libmpcodecs/ad_ffmpeg.c (revision 35227)
+++ libmpcodecs/ad_ffmpeg.c (revision 35228)
@@ -57,7 +57,7 @@
{
int broken_srate = 0;
int samplerate = lavc_context->sample_rate;
- int sample_format = samplefmt2affmt(lavc_context->sample_fmt);
+ int sample_format = samplefmt2affmt(av_get_packed_sample_fmt(lavc_context->sample_fmt));
if (!sample_format)
sample_format = sh_audio->sample_format;
if(sh_audio->wf){
@@ -169,10 +169,10 @@
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
switch (lavc_context->sample_fmt) {
- case AV_SAMPLE_FMT_U8:
- case AV_SAMPLE_FMT_S16:
- case AV_SAMPLE_FMT_S32:
- case AV_SAMPLE_FMT_FLT:
+ case AV_SAMPLE_FMT_U8: case AV_SAMPLE_FMT_U8P:
+ case AV_SAMPLE_FMT_S16: case AV_SAMPLE_FMT_S16P:
+ case AV_SAMPLE_FMT_S32: case AV_SAMPLE_FMT_S32P:
+ case AV_SAMPLE_FMT_FLT: case AV_SAMPLE_FMT_FLTP:
break;
default:
return 0;
@@ -202,10 +202,68 @@
return CONTROL_UNKNOWN;
}
+static av_always_inline void copy_samples_planar(unsigned bps,
+ unsigned nb_samples,
+ unsigned nb_channels,
+ unsigned char *dst,
+ unsigned char **src)
+{
+ unsigned s, c, o = 0;
+
+ for (s = 0; s < nb_samples; s++) {
+ for (c = 0; c < nb_channels; c++) {
+ memcpy(dst, src[c] + o, bps);
+ dst += bps;
+ }
+ o += bps;
+ }
+}
+
+static int copy_samples(AVCodecContext *avc, AVFrame *frame,
+ unsigned char *buf, int max_size)
+{
+ int channels = avc->channels;
+ int sample_size = av_get_bytes_per_sample(avc->sample_fmt);
+ int size = channels * sample_size * frame->nb_samples;
+
+ if (size > max_size) {
+ av_log(avc, AV_LOG_ERROR,
+ "Buffer overflow while decoding a single frame\n");
+ return AVERROR(EINVAL); /* same as avcodec_decode_audio3 */
+ }
+ /* TODO reorder channels at the same time */
+ if (av_sample_fmt_is_planar(avc->sample_fmt)) {
+ switch (sample_size) {
+ case 1:
+ copy_samples_planar(1, frame->nb_samples, channels,
+ buf, frame->extended_data);
+ break;
+ case 2:
+ copy_samples_planar(2, frame->nb_samples, channels,
+ buf, frame->extended_data);
+ break;
+ case 4:
+ copy_samples_planar(4, frame->nb_samples, channels,
+ buf, frame->extended_data);
+ default:
+ copy_samples_planar(sample_size, frame->nb_samples, channels,
+ buf, frame->extended_data);
+ }
+ } else {
+ memcpy(buf, frame->data[0], size);
+ }
+ return size;
+}
+
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned char *start=NULL;
- int y,len=-1;
+ int y,len=-1, got_frame;
+ AVFrame *frame = avcodec_alloc_frame();
+
+ if (!frame)
+ return AVERROR(ENOMEM);
+
while(len<minlen){
AVPacket pkt;
int len2=maxlen;
@@ -230,7 +288,7 @@
sh_audio->pts = pts;
sh_audio->pts_bytes = 0;
}
- y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
+ y=avcodec_decode_audio4(sh_audio->context, frame, &got_frame, &pkt);
//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
// LATM may need many packets to find mux info
if (y == AVERROR(EAGAIN))
@@ -238,6 +296,11 @@
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
if(!sh_audio->parser && y<x)
sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
+ if (!got_frame)
+ continue;
+ len2 = copy_samples(sh_audio->context, frame, buf, maxlen);
+ if (len2 < 0)
+ return len2;
if(len2>0){
if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
int samplesize = av_get_bytes_per_sample(((AVCodecContext *)
@@ -258,5 +321,7 @@
if (setup_format(sh_audio, sh_audio->context))
break;
}
+
+ av_free(frame);
return len;
}
|
GitWeb