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-rw-r--r--cfg-common.h4
-rw-r--r--cfg-mplayer.h4
-rw-r--r--libmpcodecs/dec_audio.c138
-rw-r--r--libmpcodecs/dec_audio.h5
-rw-r--r--libmpdemux/stheader.h12
-rw-r--r--mencoder.c57
-rw-r--r--mplayer.c35
7 files changed, 218 insertions, 37 deletions
diff --git a/cfg-common.h b/cfg-common.h
index 6fa1478c0b..8071ec876a 100644
--- a/cfg-common.h
+++ b/cfg-common.h
@@ -91,6 +91,7 @@
// force video/audio rate:
{"fps", &force_fps, CONF_TYPE_FLOAT, CONF_MIN, 0, 0, NULL},
{"srate", &force_srate, CONF_TYPE_INT, CONF_RANGE, 1000, 8*48000, NULL},
+ {"channels", &audio_output_channels, CONF_TYPE_INT, CONF_RANGE, 1, 6, NULL},
// ------------------------- codec/vfilter options --------------------
@@ -187,6 +188,9 @@ extern float movie_aspect;
extern int softzoom;
extern int flip;
+/* from dec_audio, currently used for ac3surround decoder only */
+extern int audio_output_channels;
+
#ifdef STREAMING
/* defined in network.c */
extern char *network_username;
diff --git a/cfg-mplayer.h b/cfg-mplayer.h
index f2e583f9a9..4fdf346371 100644
--- a/cfg-mplayer.h
+++ b/cfg-mplayer.h
@@ -111,9 +111,6 @@ extern int nortc;
/* from libvo/aspect.c */
extern float monitor_aspect;
-/* from dec_audio, currently used for ac3surround decoder only */
-extern int audio_output_channels;
-
/* Options related to audio out plugins */
struct config ao_plugin_conf[]={
{"list", &ao_plugin_cfg.plugin_list, CONF_TYPE_STRING, 0, 0, 0, NULL},
@@ -173,7 +170,6 @@ static config_t mplayer_opts[]={
{"dsp", "Use -ao oss:dsp_path!\n", CONF_TYPE_PRINT, CONF_NOCFG, 0, 0, NULL},
{"mixer", &mixer_device, CONF_TYPE_STRING, 0, 0, 0, NULL},
{"master", "Option -master has been removed, use -aop list=volume instead.\n", CONF_TYPE_PRINT, 0, 0, 0, NULL},
- {"channels", &audio_output_channels, CONF_TYPE_INT, CONF_RANGE, 2, 6, NULL},
// override audio buffer size (used only by -ao oss, anyway obsolete...)
{"abs", &ao_data.buffersize, CONF_TYPE_INT, CONF_MIN, 0, 0, NULL},
diff --git a/libmpcodecs/dec_audio.c b/libmpcodecs/dec_audio.c
index 29ceefbf5b..51353ed0b2 100644
--- a/libmpcodecs/dec_audio.c
+++ b/libmpcodecs/dec_audio.c
@@ -18,6 +18,8 @@ extern int verbose; // defined in mplayer.c
#include "ad.h"
#include "../libao2/afmt.h"
+#include "../libaf/af.h"
+
#ifdef USE_FAKE_MONO
int fakemono=0;
#endif
@@ -118,6 +120,10 @@ int init_audio_codec(sh_audio_t *sh_audio)
sh_audio->samplerate,sh_audio->channels,
sh_audio->samplesize*8,sh_audio->sample_format,
sh_audio->i_bps,sh_audio->o_bps,sh_audio->i_bps*8*0.001);
+
+ sh_audio->a_out_buffer_size=sh_audio->a_buffer_size;
+ sh_audio->a_out_buffer=sh_audio->a_buffer;
+ sh_audio->a_out_buffer_len=sh_audio->a_buffer_len;
return 1;
}
@@ -210,23 +216,149 @@ return 1; // success
void uninit_audio(sh_audio_t *sh_audio)
{
+ if(sh_audio->afilter){
+ mp_msg(MSGT_DECAUDIO,MSGL_V,"Uninit audio filters...\n");
+ af_uninit(sh_audio->afilter);
+ sh_audio->afilter=NULL;
+ }
if(sh_audio->inited){
mp_msg(MSGT_DECAUDIO,MSGL_V,MSGTR_UninitAudioStr,sh_audio->codec->drv);
mpadec->uninit(sh_audio);
sh_audio->inited=0;
}
+ if(sh_audio->a_out_buffer!=sh_audio->a_buffer) free(sh_audio->a_out_buffer);
+ sh_audio->a_out_buffer=NULL;
if(sh_audio->a_buffer) free(sh_audio->a_buffer);
sh_audio->a_buffer=NULL;
if(sh_audio->a_in_buffer) free(sh_audio->a_in_buffer);
sh_audio->a_in_buffer=NULL;
}
+ /* Init audio filters */
+int init_audio_filters(sh_audio_t *sh_audio,
+ int in_samplerate, int in_channels, int in_format, int in_bps,
+ int out_samplerate, int out_channels, int out_format, int out_bps,
+ int out_minsize, int out_maxsize){
+ af_stream_t* afs=malloc(sizeof(af_stream_t));
+ memset(afs,0,sizeof(af_stream_t));
+
+ // input format: same as codec's output format:
+ afs->input.rate = in_samplerate;
+ afs->input.nch = in_channels;
+ 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 ? out_format : afs->input.format;
+ afs->output.bps = out_bps ? out_bps : afs->input.bps;
+
+ // filter config:
+ afs->cfg.force = 0;
+ afs->cfg.list = NULL;
+
+ mp_msg(MSGT_DECAUDIO, MSGL_INFO, "Building audio filter chain for %dHz/%dch/%dbit -> %dHz/%dch/%dbit...\n",
+ afs->input.rate,afs->input.nch,afs->input.bps*8,
+ afs->output.rate,afs->output.nch,afs->output.bps*8);
+
+ // let's autoprobe it!
+ if(0 != af_init(afs)){
+ free(afs);
+ return 0; // failed :(
+ }
+
+ // allocate the a_out_* buffers:
+ if(out_maxsize<out_minsize) out_maxsize=out_minsize;
+ if(out_maxsize<8192) out_maxsize=MAX_OUTBURST; // not sure this is ok
+
+ sh_audio->a_out_buffer_size=out_maxsize;
+ sh_audio->a_out_buffer=malloc(sh_audio->a_out_buffer_size);
+ memset(sh_audio->a_out_buffer,0,sh_audio->a_out_buffer_size);
+ sh_audio->a_out_buffer_len=0;
+
+ // ok!
+ sh_audio->afilter=(void*)afs;
+ return 1;
+}
+
int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
- if(sh_audio->inited)
- return mpadec->decode_audio(sh_audio,buf,minlen,maxlen);
+ int declen;
+ af_data_t afd; // filter input
+ af_data_t* pafd; // filter output
+
+ if(!sh_audio->inited) return -1; // no codec
+ if(!sh_audio->afilter){
+ // no filter, just decode:
+ // FIXME: don't drop initial decoded data in a_buffer!
+ return mpadec->decode_audio(sh_audio,buf,minlen,maxlen);
+ }
+
+// declen=af_inputlen(sh_audio->afilter,minlen);
+ declen=af_calc_insize_constrained(sh_audio->afilter,minlen,maxlen,
+ sh_audio->a_buffer_size-sh_audio->audio_out_minsize);
+
+ mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\ndecaudio: minlen=%d maxlen=%d declen=%d (max=%d)\n",
+ minlen, maxlen, declen, sh_audio->a_buffer_size);
+
+ if(declen<=0) return -1; // error!
+
+ // limit declen to buffer size: - DONE by af_calc_insize_constrained
+// if(declen>sh_audio->a_buffer_size) declen=sh_audio->a_buffer_size;
+
+ // decode if needed:
+ while(declen>sh_audio->a_buffer_len){
+ int len=declen-sh_audio->a_buffer_len;
+ int maxlen=sh_audio->a_buffer_size-sh_audio->a_buffer_len;
+
+ mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"decaudio: decoding %d bytes, max: %d (%d)\n",
+ len, maxlen, sh_audio->audio_out_minsize);
+
+ if(maxlen<sh_audio->audio_out_minsize) break; // don't overflow buffer!
+ // not enough decoded data waiting, decode 'len' bytes more:
+ len=mpadec->decode_audio(sh_audio,
+ sh_audio->a_buffer+sh_audio->a_buffer_len, len, maxlen);
+ if(len<=0) break; // EOF?
+ sh_audio->a_buffer_len+=len;
+ }
+ if(declen>sh_audio->a_buffer_len)
+ declen=sh_audio->a_buffer_len; // still no enough data (EOF) :(
+
+ // round to whole samples:
+// declen/=sh_audio->samplesize*sh_audio->channels;
+// declen*=sh_audio->samplesize*sh_audio->channels;
+
+ // run the filters:
+ afd.audio=sh_audio->a_buffer;
+ afd.len=declen;
+ afd.rate=sh_audio->samplerate;
+ afd.nch=sh_audio->channels;
+ afd.format=sh_audio->sample_format;
+ afd.bps=sh_audio->samplesize;
+ //pafd=&afd;
+// printf("\nAF: %d --> ",declen);
+ pafd=af_play(sh_audio->afilter,&afd);
+// printf("%d \n",pafd->len);
+
+ if(!pafd) return -1; // error
+
+ mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"decaudio: declen=%d out=%d (max %d)\n",
+ declen, pafd->len, maxlen);
+
+ // copy filter==>out:
+ if(maxlen < pafd->len)
+ mp_msg(MSGT_DECAUDIO,MSGL_WARN,"%i bytes of audio data lost due to buffer overflow, len = %i", pafd->len - maxlen,pafd->len);
else
- return -1;
+ maxlen=pafd->len;
+ memmove(buf, pafd->audio, maxlen);
+
+ // remove processed data from decoder buffer:
+ sh_audio->a_buffer_len-=declen;
+ if(sh_audio->a_buffer_len>0)
+ memmove(sh_audio->a_buffer, sh_audio->a_buffer+declen, sh_audio->a_buffer_len);
+
+ return maxlen;
}
void resync_audio_stream(sh_audio_t *sh_audio)
diff --git a/libmpcodecs/dec_audio.h b/libmpcodecs/dec_audio.h
index d4d5b67917..cc44607dab 100644
--- a/libmpcodecs/dec_audio.h
+++ b/libmpcodecs/dec_audio.h
@@ -9,3 +9,8 @@ extern int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m
extern void resync_audio_stream(sh_audio_t *sh_audio);
extern void skip_audio_frame(sh_audio_t *sh_audio);
extern void uninit_audio(sh_audio_t *sh_audio);
+
+extern int init_audio_filters(sh_audio_t *sh_audio,
+ int in_samplerate, int in_channels, int in_format, int in_bps,
+ int out_samplerate, int out_channels, int out_format, int out_bps,
+ int out_minsize, int out_maxsize); \ No newline at end of file
diff --git a/libmpdemux/stheader.h b/libmpdemux/stheader.h
index 559af7d67f..12dff42c10 100644
--- a/libmpdemux/stheader.h
+++ b/libmpdemux/stheader.h
@@ -55,15 +55,21 @@ typedef struct {
int o_bps; // == samplerate*samplesize*channels (uncompr. bytes/sec)
int i_bps; // == bitrate (compressed bytes/sec)
// in buffers:
- int audio_in_minsize;
+ int audio_in_minsize; // max. compressed packet size (== min. in buffer size)
char* a_in_buffer;
int a_in_buffer_len;
int a_in_buffer_size;
- // out buffers:
- int audio_out_minsize;
+ // decoder buffers:
+ int audio_out_minsize; // max. uncompressed packet size (==min. out buffsize)
char* a_buffer;
int a_buffer_len;
int a_buffer_size;
+ // output buffers:
+ char* a_out_buffer;
+ int a_out_buffer_len;
+ int a_out_buffer_size;
+// void* audio_out; // the audio_out handle, used for this audio stream
+ void* afilter; // the audio filter stream
// win32-compatible codec parameters:
AVIStreamHeader audio;
WAVEFORMATEX* wf;
diff --git a/mencoder.c b/mencoder.c
index d2ad95705c..ff283af118 100644
--- a/mencoder.c
+++ b/mencoder.c
@@ -45,6 +45,8 @@ static char* banner_text=
#include "libvo/video_out.h"
+#include "libao2/afmt.h"
+
#include "libmpcodecs/mp_image.h"
#include "libmpcodecs/dec_audio.h"
#include "libmpcodecs/dec_video.h"
@@ -249,19 +251,19 @@ static int dec_audio(sh_audio_t *sh_audio,unsigned char* buffer,int total){
while(size<total && !at_eof){
int len=total-size;
if(len>MAX_OUTBURST) len=MAX_OUTBURST;
- if(len>sh_audio->a_buffer_size) len=sh_audio->a_buffer_size;
- if(len>sh_audio->a_buffer_len){
+ if(len>sh_audio->a_out_buffer_size) len=sh_audio->a_out_buffer_size;
+ if(len>sh_audio->a_out_buffer_len){
int ret=decode_audio(sh_audio,
- &sh_audio->a_buffer[sh_audio->a_buffer_len],
- len-sh_audio->a_buffer_len,
- sh_audio->a_buffer_size-sh_audio->a_buffer_len);
- if(ret>0) sh_audio->a_buffer_len+=ret; else at_eof=1;
+ &sh_audio->a_out_buffer[sh_audio->a_out_buffer_len],
+ len-sh_audio->a_out_buffer_len,
+ sh_audio->a_out_buffer_size-sh_audio->a_out_buffer_len);
+ if(ret>0) sh_audio->a_out_buffer_len+=ret; else at_eof=1;
}
- if(len>sh_audio->a_buffer_len) len=sh_audio->a_buffer_len;
- memcpy(buffer+size,sh_audio->a_buffer,len);
- sh_audio->a_buffer_len-=len; size+=len;
- if(sh_audio->a_buffer_len>0)
- memcpy(sh_audio->a_buffer,&sh_audio->a_buffer[len],sh_audio->a_buffer_len);
+ if(len>sh_audio->a_out_buffer_len) len=sh_audio->a_out_buffer_len;
+ memcpy(buffer+size,sh_audio->a_out_buffer,len);
+ sh_audio->a_out_buffer_len-=len; size+=len;
+ if(sh_audio->a_out_buffer_len>0)
+ memcpy(sh_audio->a_out_buffer,&sh_audio->a_out_buffer[len],sh_audio->a_out_buffer_len);
}
return size;
}
@@ -694,15 +696,25 @@ case ACODEC_PCM:
printf("CBR PCM audio selected\n");
mux_a->h.dwSampleSize=2*sh_audio->channels;
mux_a->h.dwScale=1;
- mux_a->h.dwRate=sh_audio->samplerate;
+ mux_a->h.dwRate=force_srate?force_srate:sh_audio->samplerate;
mux_a->wf=malloc(sizeof(WAVEFORMATEX));
mux_a->wf->nBlockAlign=mux_a->h.dwSampleSize;
mux_a->wf->wFormatTag=0x1; // PCM
- mux_a->wf->nChannels=sh_audio->channels;
- mux_a->wf->nSamplesPerSec=sh_audio->samplerate;
+ mux_a->wf->nChannels=audio_output_channels?audio_output_channels:sh_audio->channels;
+ mux_a->wf->nSamplesPerSec=mux_a->h.dwRate;
mux_a->wf->nAvgBytesPerSec=mux_a->h.dwSampleSize*mux_a->wf->nSamplesPerSec;
mux_a->wf->wBitsPerSample=16;
mux_a->wf->cbSize=0; // FIXME for l3codeca.acm
+ // setup filter:
+ if(!init_audio_filters(sh_audio,
+ sh_audio->samplerate,
+ sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize,
+ mux_a->wf->nSamplesPerSec, mux_a->wf->nChannels,
+ (mux_a->wf->wBitsPerSample==8)? AFMT_U8:AFMT_S16_LE,
+ mux_a->wf->wBitsPerSample/8,
+ 16384, mux_a->wf->nAvgBytesPerSec)){
+ mp_msg(MSGT_CPLAYER,MSGL_ERR,"Couldn't find matching filter / ao format!\n");
+ }
break;
case ACODEC_VBRMP3:
printf("MP3 audio selected\n");
@@ -712,8 +724,9 @@ case ACODEC_VBRMP3:
if(sizeof(MPEGLAYER3WAVEFORMAT)!=30) mp_msg(MSGT_MENCODER,MSGL_WARN,"sizeof(MPEGLAYER3WAVEFORMAT)==%d!=30, maybe broken C compiler?\n",sizeof(MPEGLAYER3WAVEFORMAT));
mux_a->wf=malloc(sizeof(MPEGLAYER3WAVEFORMAT)); // should be 30
mux_a->wf->wFormatTag=0x55; // MP3
- mux_a->wf->nChannels= sh_audio->channels;
- mux_a->wf->nSamplesPerSec=force_srate?force_srate:sh_audio->samplerate;
+ mux_a->wf->nChannels= (lame_param_mode<0) ? sh_audio->channels :
+ ((lame_param_mode==3) ? 1 : 2);
+ mux_a->wf->nSamplesPerSec=mux_a->h.dwRate;
mux_a->wf->nAvgBytesPerSec=192000/8; // FIXME!
mux_a->wf->nBlockAlign=(mux_a->h.dwRate<32000)?576:1152; // required for l3codeca.acm + WMP 6.4
mux_a->wf->wBitsPerSample=0; //16;
@@ -724,6 +737,15 @@ case ACODEC_VBRMP3:
((MPEGLAYER3WAVEFORMAT*)(mux_a->wf))->nBlockSize=(mux_a->h.dwRate<32000)?576:1152; // ???
((MPEGLAYER3WAVEFORMAT*)(mux_a->wf))->nFramesPerBlock=1;
((MPEGLAYER3WAVEFORMAT*)(mux_a->wf))->nCodecDelay=0;
+ // setup filter:
+ if(!init_audio_filters(sh_audio,
+ sh_audio->samplerate,
+ sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize,
+ mux_a->wf->nSamplesPerSec, mux_a->wf->nChannels,
+ AFMT_S16_LE, 2,
+ 4608, mux_a->h.dwRate*mux_a->wf->nChannels*2)){
+ mp_msg(MSGT_CPLAYER,MSGL_ERR,"Couldn't find matching filter / ao format!\n");
+ }
break;
}
@@ -748,7 +770,8 @@ case ACODEC_VBRMP3:
lame=lame_init();
lame_set_bWriteVbrTag(lame,0);
-lame_set_in_samplerate(lame,sh_audio->samplerate);
+lame_set_in_samplerate(lame,mux_a->wf->nSamplesPerSec);
+//lame_set_in_samplerate(lame,sh_audio->samplerate); // if resampling done by lame
lame_set_num_channels(lame,mux_a->wf->nChannels);
lame_set_out_samplerate(lame,mux_a->wf->nSamplesPerSec);
lame_set_quality(lame,lame_param_algqual); // 0 = best q
diff --git a/mplayer.c b/mplayer.c
index 378467178c..fd5d0a0dca 100644
--- a/mplayer.c
+++ b/mplayer.c
@@ -1237,6 +1237,7 @@ if(sh_audio){
if(!(audio_out=init_best_audio_out(audio_driver_list,
(ao_plugin_cfg.plugin_list), // plugin flag
force_srate?force_srate:sh_audio->samplerate,
+ audio_output_channels?audio_output_channels:
sh_audio->channels,sh_audio->sample_format,0))){
// FAILED:
mp_msg(MSGT_CPLAYER,MSGL_ERR,MSGTR_CannotInitAO);
@@ -1244,7 +1245,7 @@ if(sh_audio){
} else {
// SUCCESS:
inited_flags|=INITED_AO;
- mp_msg(MSGT_CPLAYER,MSGL_INFO,"AO: [%s] %iHz %dch %s\n",
+ mp_msg(MSGT_CPLAYER,MSGL_INFO,"AO: [%s] %dHz %dch %s\n",
audio_out->info->short_name,
force_srate?force_srate:sh_audio->samplerate,
sh_audio->channels,
@@ -1253,6 +1254,19 @@ if(sh_audio){
audio_out->info->name, audio_out->info->author);
if(strlen(audio_out->info->comment) > 0)
mp_msg(MSGT_CPLAYER,MSGL_V,MSGTR_AOComment, audio_out->info->comment);
+ // init audio filters:
+#if 1
+ current_module="af_init";
+ if(!init_audio_filters(sh_audio,
+ sh_audio->samplerate,
+ sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize,
+ ao_data.samplerate, ao_data.channels, ao_data.format,
+ audio_out_format_bits(ao_data.format)/8, /* ao_data.bps, */
+ ao_data.outburst*4, ao_data.buffersize)){
+ mp_msg(MSGT_CPLAYER,MSGL_ERR,"Couldn't find matching filter / ao format, -> nosound\n");
+ sh_audio=d_audio->sh=NULL; // -> nosound
+ }
+#endif
}
}
@@ -1338,24 +1352,25 @@ while(sh_audio){
// Fill buffer if needed:
current_module="decode_audio"; // Enter AUDIO decoder module
t=GetTimer();
- while(sh_audio->a_buffer_len<playsize && !d_audio->eof){
- int ret=decode_audio(sh_audio,&sh_audio->a_buffer[sh_audio->a_buffer_len],
- playsize-sh_audio->a_buffer_len,sh_audio->a_buffer_size-sh_audio->a_buffer_len);
+ while(sh_audio->a_out_buffer_len<playsize && !d_audio->eof){
+ int ret=decode_audio(sh_audio,&sh_audio->a_out_buffer[sh_audio->a_out_buffer_len],
+ playsize-sh_audio->a_out_buffer_len,sh_audio->a_out_buffer_size-sh_audio->a_out_buffer_len);
if(ret<=0) break; // EOF?
- sh_audio->a_buffer_len+=ret;
+ sh_audio->a_out_buffer_len+=ret;
}
t=GetTimer()-t;
tt = t*0.000001f; audio_time_usage+=tt;
- if(playsize>sh_audio->a_buffer_len) playsize=sh_audio->a_buffer_len;
+ if(playsize>sh_audio->a_out_buffer_len) playsize=sh_audio->a_out_buffer_len;
// play audio:
current_module="play_audio";
- playsize=audio_out->play(sh_audio->a_buffer,playsize,0);
+ playsize=audio_out->play(sh_audio->a_out_buffer,playsize,0);
if(playsize>0){
- sh_audio->a_buffer_len-=playsize;
- memmove(sh_audio->a_buffer,&sh_audio->a_buffer[playsize],sh_audio->a_buffer_len);
- sh_audio->timer+=playsize/(float)(sh_audio->o_bps);
+ sh_audio->a_out_buffer_len-=playsize;
+ memmove(sh_audio->a_out_buffer,&sh_audio->a_out_buffer[playsize],sh_audio->a_out_buffer_len);
+ sh_audio->timer+=playsize/((float)((ao_data.bps && sh_audio->afilter) ?
+ ao_data.bps : sh_audio->o_bps));
}
break;