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authoralex <alex@b3059339-0415-0410-9bf9-f77b7e298cf2>2004-12-27 17:30:15 +0000
committeralex <alex@b3059339-0415-0410-9bf9-f77b7e298cf2>2004-12-27 17:30:15 +0000
commit507121f7fe2d170dd8db99d3112602036ddef718 (patch)
tree38b26e115cfadde356b005496286f78307839440 /libmpcodecs
parent00f99a82a8f57573e3e6982cf9d014c9b9d8a68b (diff)
downloadmpv-507121f7fe2d170dd8db99d3112602036ddef718.tar.bz2
mpv-507121f7fe2d170dd8db99d3112602036ddef718.tar.xz
removing AFMT_ dependancy
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14246 b3059339-0415-0410-9bf9-f77b7e298cf2
Diffstat (limited to 'libmpcodecs')
-rw-r--r--libmpcodecs/ad_dvdpcm.c10
-rw-r--r--libmpcodecs/ad_hwac3.c2
-rw-r--r--libmpcodecs/ad_internal.h2
-rw-r--r--libmpcodecs/ad_pcm.c30
-rw-r--r--libmpcodecs/ad_ra1428.c2
-rw-r--r--libmpcodecs/ad_sample.c2
-rw-r--r--libmpcodecs/dec_audio.c22
7 files changed, 37 insertions, 33 deletions
diff --git a/libmpcodecs/ad_dvdpcm.c b/libmpcodecs/ad_dvdpcm.c
index acfbc7855a..1962594961 100644
--- a/libmpcodecs/ad_dvdpcm.c
+++ b/libmpcodecs/ad_dvdpcm.c
@@ -6,7 +6,6 @@
#include "mp_msg.h"
#include "help_mp.h"
#include "ad_internal.h"
-#include "../libaf/af_format.h"
static ad_info_t info =
{
@@ -35,26 +34,25 @@ static int init(sh_audio_t *sh)
}
switch ((h >> 6) & 3) {
case 0:
- sh->sample_format = AFMT_S16_BE;
+ sh->sample_format = AF_FORMAT_S16_BE;
sh->samplesize = 2;
break;
case 1:
mp_msg(MSGT_DECAUDIO, MSGL_INFO, MSGTR_SamplesWanted);
sh->i_bps = sh->channels * sh->samplerate * 5 / 2;
case 2:
- sh->sample_format = AFMT_AF_FLAGS | AF_FORMAT_I |
- AF_FORMAT_BE | AF_FORMAT_SI;
+ sh->sample_format = AF_FORMAT_S24_BE;
sh->samplesize = 3;
break;
default:
- sh->sample_format = AFMT_S16_BE;
+ sh->sample_format = AF_FORMAT_S16_BE;
sh->samplesize = 2;
}
} else {
// use defaults:
sh->channels=2;
sh->samplerate=48000;
- sh->sample_format = AFMT_S16_BE;
+ sh->sample_format = AF_FORMAT_S16_BE;
sh->samplesize = 2;
}
if (!sh->i_bps)
diff --git a/libmpcodecs/ad_hwac3.c b/libmpcodecs/ad_hwac3.c
index 764625eb3b..147568e746 100644
--- a/libmpcodecs/ad_hwac3.c
+++ b/libmpcodecs/ad_hwac3.c
@@ -105,7 +105,7 @@ static int preinit(sh_audio_t *sh)
sh->audio_in_minsize = 8192;
sh->channels = 2;
sh->samplesize = 2;
- sh->sample_format = AFMT_AC3;
+ sh->sample_format = AF_FORMAT_AC3;
return 1;
}
diff --git a/libmpcodecs/ad_internal.h b/libmpcodecs/ad_internal.h
index e5b67f5844..91890ebdb4 100644
--- a/libmpcodecs/ad_internal.h
+++ b/libmpcodecs/ad_internal.h
@@ -1,6 +1,6 @@
#include "codec-cfg.h"
-#include "../libao2/afmt.h"
+#include "../libaf/af_format.h"
#include "stream.h"
#include "demuxer.h"
diff --git a/libmpcodecs/ad_pcm.c b/libmpcodecs/ad_pcm.c
index 8288a69612..a2bcd17827 100644
--- a/libmpcodecs/ad_pcm.c
+++ b/libmpcodecs/ad_pcm.c
@@ -24,31 +24,31 @@ static int init(sh_audio_t *sh_audio)
sh_audio->channels=h->nChannels;
sh_audio->samplerate=h->nSamplesPerSec;
sh_audio->samplesize=(h->wBitsPerSample+7)/8;
- sh_audio->sample_format=AFMT_S16_LE; // default
+ sh_audio->sample_format=AF_FORMAT_S16_LE; // default
switch(sh_audio->format){ /* hardware formats: */
case 0x0:
case 0x1: // Microsoft PCM
switch (sh_audio->samplesize) {
- case 1: sh_audio->sample_format=AFMT_U8; break;
- case 2: sh_audio->sample_format=AFMT_S16_LE; break;
- case 3: sh_audio->sample_format=AFMT_S24_LE; break;
- case 4: sh_audio->sample_format=AFMT_S32_LE; break;
+ case 1: sh_audio->sample_format=AF_FORMAT_U8; break;
+ case 2: sh_audio->sample_format=AF_FORMAT_S16_LE; break;
+ case 3: sh_audio->sample_format=AF_FORMAT_S24_LE; break;
+ case 4: sh_audio->sample_format=AF_FORMAT_S32_LE; break;
}
break;
- case 0x6: sh_audio->sample_format=AFMT_A_LAW;break;
- case 0x7: sh_audio->sample_format=AFMT_MU_LAW;break;
- case 0x11: sh_audio->sample_format=AFMT_IMA_ADPCM;break;
- case 0x50: sh_audio->sample_format=AFMT_MPEG;break;
+ case 0x6: sh_audio->sample_format=AF_FORMAT_A_LAW;break;
+ case 0x7: sh_audio->sample_format=AF_FORMAT_MU_LAW;break;
+ case 0x11: sh_audio->sample_format=AF_FORMAT_IMA_ADPCM;break;
+ case 0x50: sh_audio->sample_format=AF_FORMAT_MPEG2;break;
/* case 0x2000: sh_audio->sample_format=AFMT_AC3; */
case 0x20776172: // 'raw '
- sh_audio->sample_format=AFMT_S16_BE;
- if(sh_audio->samplesize==1) sh_audio->sample_format=AFMT_U8;
+ sh_audio->sample_format=AF_FORMAT_S16_BE;
+ if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_U8;
break;
case 0x736F7774: // 'twos'
- sh_audio->sample_format=AFMT_S16_BE;
+ sh_audio->sample_format=AF_FORMAT_S16_BE;
// intended fall-through
case 0x74776F73: // 'swot'
- if(sh_audio->samplesize==1) sh_audio->sample_format=AFMT_S8;
+ if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_S8;
// Uncomment this if twos audio is broken for you
// (typically with movies made on sgi machines)
// This is just a workaround, the real bug is elsewhere
@@ -58,10 +58,10 @@ static int init(sh_audio_t *sh_audio)
#endif
break;
case 0x32336c66: // 'fl32', bigendian float32
- sh_audio->sample_format=AFMT_AF_FLAGS | AF_FORMAT_BE | AF_FORMAT_F;
+ sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
sh_audio->samplesize=4;
break;
- default: if(sh_audio->samplesize!=2) sh_audio->sample_format=AFMT_U8;
+ default: if(sh_audio->samplesize!=2) sh_audio->sample_format=AF_FORMAT_U8;
}
return 1;
}
diff --git a/libmpcodecs/ad_ra1428.c b/libmpcodecs/ad_ra1428.c
index 56dbdda394..0c75c17e88 100644
--- a/libmpcodecs/ad_ra1428.c
+++ b/libmpcodecs/ad_ra1428.c
@@ -25,7 +25,7 @@ static int preinit(sh_audio_t *sh) {
sh->samplerate=sh->wf->nSamplesPerSec;
sh->samplesize=sh->wf->wBitsPerSample/8;
sh->channels=sh->wf->nChannels;
- sh->sample_format=AFMT_S16_LE;
+ sh->sample_format=AF_FORMAT_S16_LE;
switch (sh->format) {
case mmioFOURCC('1','4','_','4'):
diff --git a/libmpcodecs/ad_sample.c b/libmpcodecs/ad_sample.c
index aa7912bba8..848152c68e 100644
--- a/libmpcodecs/ad_sample.c
+++ b/libmpcodecs/ad_sample.c
@@ -47,7 +47,7 @@ static int preinit(sh_audio_t *sh){
sh->samplesize=2; // bytes (not bits!) per sample per channel
sh->channels=2; // number of channels
sh->samplerate=44100; // samplerate
- sh->sample_format=AFMT_S16_LE; // sample format, see libao2/afmt.h
+ sh->sample_format=AF_FORMAT_S16_LE; // sample format, see libao2/afmt.h
sh->i_bps=64000/8; // input data rate (compressed bytes per second)
// Note: if you have VBR or unknown input rate, set it to some common or
diff --git a/libmpcodecs/dec_audio.c b/libmpcodecs/dec_audio.c
index e13dfdd5d0..4f615b3f84 100644
--- a/libmpcodecs/dec_audio.c
+++ b/libmpcodecs/dec_audio.c
@@ -14,7 +14,7 @@
#include "dec_audio.h"
#include "ad.h"
-#include "../libao2/afmt.h"
+#include "../libaf/af_format.h"
#include "../libaf/af.h"
@@ -267,13 +267,15 @@ int preinit_audio_filters(sh_audio_t *sh_audio,
// input format: same as codec's output format:
afs->input.rate = in_samplerate;
afs->input.nch = in_channels;
- afs->input.format = af_format_decode(in_format);
+// afs->input.format = af_format_decode(in_format);
+ afs->input.format = in_format;
afs->input.bps = in_bps;
// output format: same as ao driver's input format (if missing, fallback to input)
afs->output.rate = *out_samplerate ? *out_samplerate : afs->input.rate;
afs->output.nch = *out_channels ? *out_channels : afs->input.nch;
- afs->output.format = *out_format ? af_format_decode(*out_format) : afs->input.format;
+// afs->output.format = *out_format ? af_format_decode(*out_format) : afs->input.format;
+ afs->output.format = *out_format ? *out_format : afs->input.format;
afs->output.bps = out_bps ? out_bps : afs->input.bps;
// filter config:
@@ -291,11 +293,12 @@ int preinit_audio_filters(sh_audio_t *sh_audio,
*out_samplerate=afs->output.rate;
*out_channels=afs->output.nch;
- *out_format=af_format_encode((void*)(&afs->output));
+// *out_format=af_format_encode((void*)(&afs->output));
+ *out_format=afs->output.format;
mp_msg(MSGT_DECAUDIO, MSGL_INFO, "AF_pre: af format: %d bps, %d ch, %d hz, %s\n",
afs->output.bps, afs->output.nch, afs->output.rate,
- fmt2str(afs->output.format,strbuf,200));
+ af_fmt2str(afs->output.format,strbuf,200));
sh_audio->afilter=(void*)afs;
return 1;
@@ -315,13 +318,15 @@ int init_audio_filters(sh_audio_t *sh_audio,
// input format: same as codec's output format:
afs->input.rate = in_samplerate;
afs->input.nch = in_channels;
- afs->input.format = af_format_decode(in_format);
+// afs->input.format = af_format_decode(in_format);
+ afs->input.format = in_format;
afs->input.bps = in_bps;
// output format: same as ao driver's input format (if missing, fallback to input)
afs->output.rate = out_samplerate ? out_samplerate : afs->input.rate;
afs->output.nch = out_channels ? out_channels : afs->input.nch;
- afs->output.format = af_format_decode(out_format ? out_format : afs->input.format);
+// afs->output.format = af_format_decode(out_format ? out_format : afs->input.format);
+ afs->output.format = out_format ? out_format : afs->input.format;
afs->output.bps = out_bps ? out_bps : afs->input.bps;
// filter config:
@@ -404,7 +409,8 @@ int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
afd.len=declen;
afd.rate=sh_audio->samplerate;
afd.nch=sh_audio->channels;
- afd.format=af_format_decode(sh_audio->sample_format);
+// afd.format=af_format_decode(sh_audio->sample_format);
+ afd.format=sh_audio->sample_format;
afd.bps=sh_audio->samplesize;
//pafd=&afd;
// printf("\nAF: %d --> ",declen);