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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option 

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3182 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
steve 2001-11-28 12:46:23 +00:00
parent b7ff737901
commit f6ee7f826c
4 changed files with 43 additions and 18 deletions
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4
cfg-mplayer.h
View File

@ -74,6 +74,9 @@ extern int pl_delay_len;
/* from libvo/aspect.c */
extern float monitor_aspect;
/* from dec_audio, currently used for ac3surround decoder only */
extern int audio_output_channels;
/*
* CONF_TYPE_FUNC_FULL :
* allows own implemtations for passing the params
@ -103,6 +106,7 @@ struct config conf[]={
{"dsp", "Use -ao oss:dsp_path!\n", CONF_TYPE_PRINT, CONF_NOCFG, 0, 0},
{"mixer", &mixer_device, CONF_TYPE_STRING, 0, 0, 0},
{"master", &mixer_usemaster, CONF_TYPE_FLAG, 0, 0, 1},
{"channels", &audio_output_channels, CONF_TYPE_INT, CONF_RANGE, 2, 6},
#ifdef HAVE_X11
{"display", &mDisplayName, CONF_TYPE_STRING, 0, 0, 0},
#endif

10
dec_audio.c
View File

@ -83,6 +83,9 @@ static struct mad_stream mad_stream;
static struct mad_frame mad_frame;
static struct mad_synth mad_synth;
/* used for ac3surround decoder - set using -channels option */
int audio_output_channels = 2;
// ensure buffer is filled with some data
static void mad_prepare_buffer(sh_audio_t* sh_audio, struct mad_stream* ms, int length)
@ -228,7 +231,8 @@ case AFM_ALAW:
break;
case AFM_AC3:
// Dolby AC3 audio:
sh_audio->audio_out_minsize=4*256*6;
// however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame
sh_audio->audio_out_minsize=audio_output_channels*2*256*6;
break;
case AFM_HWAC3:
// Dolby AC3 audio:
@ -329,7 +333,7 @@ case AFM_AC3: {
// Dolby AC3 audio:
dec_audio_sh=sh_audio; // save sh_audio for the callback:
ac3_config.fill_buffer_callback = ac3_fill_buffer;
ac3_config.num_output_ch = 2;
ac3_config.num_output_ch = audio_output_channels;
ac3_config.flags = 0;
if(gCpuCaps.hasMMX){
ac3_config.flags |= AC3_MMX_ENABLE;
@ -342,7 +346,7 @@ if(gCpuCaps.has3DNow){
if(sh_audio->ac3_frame){
ac3_frame_t* fr=(ac3_frame_t*)sh_audio->ac3_frame;
sh_audio->samplerate=fr->sampling_rate;
sh_audio->channels=2;
sh_audio->channels=ac3_config.num_output_ch;
// 1 frame: 6*256 samples 1 sec: sh_audio->samplerate samples
//sh_audio->i_bps=fr->frame_size*fr->sampling_rate/(6*256);
sh_audio->i_bps=fr->bit_rate*(1000/8);

1
help_mp-en.h
View File

@ -33,6 +33,7 @@ static char help_text[]=
#ifdef USE_FAKE_MONO
" -stereo <mode> select MPEG1 stereo output (0:stereo 1:left 2:right)\n"
#endif
" -channels <n> target number of audio output channels\n"
" -fs -vm -zoom fullscreen playing options (fullscr,vidmode chg,softw.scale)\n"
" -x <x> -y <y> scale image to <x> * <y> resolution [if -vo driver supports!]\n"
" -sub <file> specify subtitle file to use (see also -subfps, -subdelay)\n"

46
libao2/ao_oss.c
View File

@ -26,6 +26,8 @@ static ao_info_t info =
""
};
/* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */
LIBAO_EXTERN(oss)
static char *dsp="/dev/dsp";
@ -95,8 +97,8 @@ static int control(int cmd,int arg){
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
// printf("ao2: %d Hz %d chans %s\n",rate,channels,
// audio_out_format_name(format));
printf("ao2: %d Hz %d chans %s\n",rate,channels,
audio_out_format_name(format));
if (ao_subdevice)
dsp = ao_subdevice;
@ -124,13 +126,26 @@ static int init(int rate,int channels,int format,int flags){
audio_out_format_name(ao_data.format), audio_out_format_name(format));
if(format != AFMT_AC3) {
ao_data.channels=channels-1;
ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_data.channels);
// set rate
ao_data.samplerate=rate;
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
printf("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
// We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
ao_data.channels = channels;
if (ao_data.channels > 2) {
if (ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1) {
printf("audio_setup: Failed to set audio device to %d channels\n", ao_data.channels);
return 0;
}
}
else {
int c = ao_data.channels-1;
if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
printf("audio_setup: Failed to set audio device to %d channels\n", ao_data.channels);
return 0;
}
}
printf("audio_setup: using %d channels (requested: %d)\n", ao_data.channels, ao_data.channels);
// set rate
ao_data.samplerate=rate;
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
printf("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
}
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
@ -195,8 +210,13 @@ static void reset(){
ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
if(ao_data.format != AFMT_AC3) {
ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_data.channels);
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
if (ao_data.channels > 2)
ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
else {
int c = ao_data.channels-1;
ioctl (audio_fd, SNDCTL_DSP_STEREO, &c);
}
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
}
}
@ -267,7 +287,3 @@ static float get_delay(){
}
return ((float)ao_data.buffersize)/(float)ao_data.bps;
}