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https://github.com/mpv-player/mpv
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Support playback of AFMT_S16_LE audio data on a big endian machine
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@1309 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
jkeil
parent
a26b39974f
commit
82eac1391a
219
libao2/ao_sun.c
219
libao2/ao_sun.c
@ -67,21 +67,21 @@ extern int verbose;
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// convert an OSS audio format specification into a sun audio encoding
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static int oss2sunfmt(int oss_format)
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{
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switch (oss_format){
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case AFMT_MU_LAW:
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return AUDIO_ENCODING_ULAW;
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case AFMT_A_LAW:
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return AUDIO_ENCODING_ALAW;
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case AFMT_S16_LE:
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return AUDIO_ENCODING_LINEAR;
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case AFMT_U8:
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return AUDIO_ENCODING_LINEAR8;
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switch (oss_format){
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case AFMT_MU_LAW:
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return AUDIO_ENCODING_ULAW;
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case AFMT_A_LAW:
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return AUDIO_ENCODING_ALAW;
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case AFMT_S16_LE:
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return AUDIO_ENCODING_LINEAR;
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case AFMT_U8:
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return AUDIO_ENCODING_LINEAR8;
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#ifdef AUDIO_ENCODING_DVI // Missing on NetBSD...
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case AFMT_IMA_ADPCM:
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return AUDIO_ENCODING_DVI;
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case AFMT_IMA_ADPCM:
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return AUDIO_ENCODING_DVI;
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#endif
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default:
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return AUDIO_ENCODING_NONE;
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default:
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return AUDIO_ENCODING_NONE;
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}
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}
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@ -101,7 +101,11 @@ static int realtime_samplecounter_available(char *dev)
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unsigned increment;
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unsigned min_increment;
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len = 44100 * 4 / 4; // amount of data for 0.25sec of 44.1khz, stereo, 16bit
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len = 44100 * 4 / 4; /* amount of data for 0.25sec of 44.1khz, stereo,
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* 16bit. 44kbyte can be sent to all supported
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* sun audio devices without blocking in the
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* "write" below.
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*/
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silence = calloc(1, len);
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if (silence == NULL)
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goto error;
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@ -172,6 +176,15 @@ static int realtime_samplecounter_available(char *dev)
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last_samplecnt = info.play.samples;
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}
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/*
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* For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes
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* chunks (== 4096 samples) to the audio device. If we see a minimum
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* sample counter increment from the soundcard driver of less than
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* 2000 samples, we assume that the driver provides a useable realtime
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* sample counter in the AUDIO_INFO play.samples field. Timing based
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* on sample counts should be much more accurate than counting whole
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* 16kbyte chunks.
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*/
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if (min_increment < 2000)
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rtsc_ok = RTSC_ENABLED;
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@ -198,92 +211,100 @@ error:
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// to set/get/query special features/parameters
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static int control(int cmd,int arg){
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switch(cmd){
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switch(cmd){
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case AOCONTROL_SET_DEVICE:
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audio_dev=(char*)arg;
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return CONTROL_OK;
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audio_dev=(char*)arg;
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return CONTROL_OK;
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case AOCONTROL_QUERY_FORMAT:
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(int rate,int channels,int format,int flags){
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audio_info_t info;
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int byte_per_sec;
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audio_info_t info;
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int byte_per_sec;
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if (ao_subdevice) audio_dev = ao_subdevice;
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if (ao_subdevice) audio_dev = ao_subdevice;
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if (enable_sample_timing == RTSC_UNKNOWN
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&& !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) {
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enable_sample_timing = realtime_samplecounter_available(audio_dev);
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}
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printf("ao2: %d Hz %d chans %s [0x%X]\n",
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rate,channels,audio_out_format_name(format),format);
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audio_fd=open(audio_dev, O_WRONLY);
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if(audio_fd<0){
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printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno));
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return 0;
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}
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ioctl(audio_fd, AUDIO_DRAIN, 0);
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AUDIO_INITINFO(&info);
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info.play.encoding = oss2sunfmt(ao_format = format);
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info.play.precision = (format==AFMT_S16_LE? AUDIO_PRECISION_16:AUDIO_PRECISION_8);
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info.play.channels = ao_channels = channels;
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info.play.sample_rate = ao_samplerate = rate;
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if(ioctl (audio_fd, AUDIO_SETINFO, &info)<0)
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printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n",channels,audio_out_format_name(format),rate);
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bytes_per_sample = channels * info.play.precision / 8;
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byte_per_sec = bytes_per_sample * rate;
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ao_outburst = byte_per_sec > 100000 ? 16384 : 8192;
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if(ao_buffersize==-1){
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// Measuring buffer size:
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void* data;
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ao_buffersize=0;
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#ifdef HAVE_AUDIO_SELECT
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data=malloc(ao_outburst); memset(data,0,ao_outburst);
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while(ao_buffersize<0x40000){
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fd_set rfds;
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struct timeval tv;
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FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
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tv.tv_sec=0; tv.tv_usec = 0;
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if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
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write(audio_fd,data,ao_outburst);
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ao_buffersize+=ao_outburst;
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if (enable_sample_timing == RTSC_UNKNOWN
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&& !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) {
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enable_sample_timing = realtime_samplecounter_available(audio_dev);
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}
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free(data);
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if(ao_buffersize==0){
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printf("\n *** Your audio driver DOES NOT support select() ***\n");
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printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
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return 0;
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printf("ao2: %d Hz %d chans %s [0x%X]\n",
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rate,channels,audio_out_format_name(format),format);
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audio_fd=open(audio_dev, O_WRONLY);
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if(audio_fd<0){
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printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno));
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return 0;
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}
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#ifdef __svr4__
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// remove the 0 bytes from the above ao_buffersize measurement from the
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// audio driver's STREAMS queue
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ioctl(audio_fd, I_FLUSH, FLUSHW);
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#endif
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ioctl(audio_fd, AUDIO_DRAIN, 0);
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AUDIO_INITINFO(&info);
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info.play.encoding = oss2sunfmt(ao_format = format);
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info.play.precision = (format==AFMT_S16_LE? AUDIO_PRECISION_16:AUDIO_PRECISION_8);
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info.play.channels = ao_channels = channels;
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info.play.sample_rate = ao_samplerate = rate;
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if(ioctl (audio_fd, AUDIO_SETINFO, &info)<0)
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printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n",
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channels, audio_out_format_name(format), rate);
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bytes_per_sample = channels * info.play.precision / 8;
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byte_per_sec = bytes_per_sample * rate;
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ao_outburst = byte_per_sec > 100000 ? 16384 : 8192;
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#ifdef __not_used__
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/*
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* hmm, ao_buffersize is currently not used in this driver, do there's
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* no need to measure it
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*/
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if(ao_buffersize==-1){
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// Measuring buffer size:
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void* data;
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ao_buffersize=0;
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#ifdef HAVE_AUDIO_SELECT
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data = malloc(ao_outburst);
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memset(data, format==AFMT_U8 ? 0x80 : 0, ao_outburst);
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while(ao_buffersize<0x40000){
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fd_set rfds;
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struct timeval tv;
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FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
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tv.tv_sec=0; tv.tv_usec = 0;
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if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
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write(audio_fd,data,ao_outburst);
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ao_buffersize+=ao_outburst;
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}
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free(data);
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if(ao_buffersize==0){
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printf("\n *** Your audio driver DOES NOT support select() ***\n");
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printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
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return 0;
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}
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#ifdef __svr4__
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// remove the 0 bytes from the above ao_buffersize measurement from the
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// audio driver's STREAMS queue
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ioctl(audio_fd, I_FLUSH, FLUSHW);
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#endif
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}
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ioctl(audio_fd, AUDIO_DRAIN, 0);
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#endif
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}
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#endif /* __not_used__ */
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AUDIO_INITINFO(&info);
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info.play.samples = 0;
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info.play.eof = 0;
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info.play.error = 0;
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ioctl (audio_fd, AUDIO_SETINFO, &info);
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AUDIO_INITINFO(&info);
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info.play.samples = 0;
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info.play.eof = 0;
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info.play.error = 0;
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ioctl (audio_fd, AUDIO_SETINFO, &info);
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queued_bursts = 0;
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queued_samples = 0;
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queued_bursts = 0;
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queued_samples = 0;
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return 1;
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return 1;
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}
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// close audio device
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@ -369,15 +390,35 @@ static int get_space(){
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// it should round it down to outburst*n
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// return: number of bytes played
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static int play(void* data,int len,int flags){
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if (len < ao_outburst) return 0;
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len /= ao_outburst;
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len = write(audio_fd, data, len*ao_outburst);
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len *= ao_outburst;
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#if WORDS_BIGENDIAN
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{
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static void *swab_buf;
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static int swab_len;
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if (ao_format == AFMT_S16_LE && len > swab_len) {
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if (swab_buf)
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swab_buf = realloc(swab_buf, len);
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else
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swab_buf = malloc(len);
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swab_len = len;
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if (swab_buf == NULL) return 0;
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}
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swab(data, swab_buf, len);
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data = swab_buf;
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}
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#endif
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len = write(audio_fd, data, len);
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if(len > 0) {
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queued_samples += len / bytes_per_sample;
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if (write(audio_fd,data,0) < 0)
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perror("ao_sun: send EOF audio record");
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else
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queued_bursts ++;
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queued_samples += len / bytes_per_sample;
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if (write(audio_fd,data,0) < 0)
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perror("ao_sun: send EOF audio record");
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else
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queued_bursts ++;
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}
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return len;
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}
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