caaef9a183
after the commit ? so that I can either get careless, or waste time/bandwidth by running each commit twice... Anyways, files/audio_openbsd.c is a `caching' audio system. It's much smarter than the old one: uses get encoding to map the reasonable encodings, does not even cross-check alaw/ulaw when 8 bit/16 bit work okay. It does keep audio opened all the way instead of closing/reopening all the time. and it avoids gratuitous assumptions about PC broken hardware, such as not having support for 8 bits signed and other brain-damage. Missing part: it won't convert endianess on the fly when the only `right' modes have the wrong endianess... this might occur with e.g., a sb16 on an alpha.
126 lines
3.1 KiB
Plaintext
126 lines
3.1 KiB
Plaintext
--- audio.c.orig Mon Nov 23 17:16:24 1998
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+++ audio.c Tue Mar 23 22:25:08 1999
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@@ -68,42 +68,43 @@
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{
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int fmts;
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int i,j,k,k1=NUM_RATES-1;
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- struct audio_info_struct ai1 = *ai;
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if (param.outmode != DECODE_AUDIO) {
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memset(capabilities,1,sizeof(capabilities));
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return;
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}
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- memset(capabilities,0,sizeof(capabilities));
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+ memset(capabilities,-1,sizeof(capabilities));
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if(param.force_rate) {
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rates[NUM_RATES-1] = param.force_rate;
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k1 = NUM_RATES;
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}
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- if(audio_open(&ai1) < 0) {
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- perror("audio");
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- exit(1);
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- }
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+ if(param.verbose > 1) {
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+ struct audio_info_struct ai1 = *ai;
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+ if(audio_open(&ai1) < 0) {
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+ perror("audio");
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+ exit(1);
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+ }
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- for(i=0;i<NUM_CHANNELS;i++) {
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- for(j=0;j<NUM_RATES;j++) {
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- ai1.channels = channels[i];
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- ai1.rate = rates[j];
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- fmts = audio_get_formats(&ai1);
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- if(fmts < 0)
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- continue;
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- for(k=0;k<NUM_ENCODINGS;k++) {
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- if((fmts & encodings[k]) == encodings[k])
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- capabilities[i][k][j] = 1;
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+ for(i=0;i<NUM_CHANNELS;i++) {
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+ for(j=0;j<NUM_RATES;j++) {
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+ ai1.channels = channels[i];
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+ ai1.rate = rates[j];
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+ fmts = audio_get_formats(&ai1);
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+ if(fmts < 0)
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+ continue;
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+ for(k=0;k<NUM_ENCODINGS;k++) {
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+ if((fmts & encodings[k]) == encodings[k])
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+ capabilities[i][k][j] = 1;
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+ else
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+ capabilities[i][k][j] = 0;
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+ }
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}
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}
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- }
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-
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- audio_close(&ai1);
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- if(param.verbose > 1) {
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- fprintf(stderr,"\nAudio capabilties:\n |");
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+ audio_close(&ai1);
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+ fprintf(stderr,"\nAudio capabilities:\n |");
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for(j=0;j<NUM_ENCODINGS;j++) {
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fprintf(stderr," %5s |",audio_val2name[j].sname);
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}
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@@ -144,7 +145,19 @@
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if(rn >= 0) {
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for(i=f0;i<f2;i++) {
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- if(capabilities[c][i][rn]) {
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+ if(capabilities[c][i][rn] == (char)-1) {
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+ int fmts, k;
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+ ai->channels = channels[c];
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+ ai->rate = rates[rn];
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+ fmts = audio_get_formats(ai);
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+ for(k=0;k<NUM_ENCODINGS;k++) {
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+ if ((fmts & encodings[k]) == encodings[k])
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+ capabilities[c][k][rn] = 1;
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+ else
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+ capabilities[c][k][rn] = 0;
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+ }
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+ }
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+ if(capabilities[c][i][rn] == 1) {
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ai->rate = rates[rn];
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ai->format = encodings[i];
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ai->channels = channels[c];
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@@ -160,7 +173,11 @@
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* c=num of channels of stream
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* r=rate of stream
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*/
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-void audio_fit_capabilities(struct audio_info_struct *ai,int c,int r)
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+#ifdef OPENBSD
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+void audio_fit_capabilities(struct audio_info_struct *ai, int c, int r)
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+#else
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+static void do_audio_fit_capabilities(struct audio_info_struct *ai,int c,int r)
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+#endif
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{
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int rn;
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int f0=0;
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@@ -247,6 +264,23 @@
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exit(1);
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}
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+#ifndef OPENBSD
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+void audio_fit_capabilities(struct audio_info_struct *ai,int c,int r)
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+{
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+ struct audio_info_struct ai1 = *ai;
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+
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+ if (param.verbose <= 1 && audio_open(&ai1) >= 0) {
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+ do_audio_fit_capabilities(&ai1, c, r);
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+ ai->rate = ai1.rate;
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+ ai->channels = ai1.channels;
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+ ai->format = ai1.format;
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+ audio_close(&ai1);
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+ } else {
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+ do_audio_fit_capabilities(ai, c, r);
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+ }
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+}
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+#endif
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+
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char *audio_encoding_name(int format)
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{
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int i;
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