musicout.c.patch

linux环境下用纯c写的RTP协议的通用开发库

5c5
<  * $Id: musicout.c.patch,v 1.2 2000/06/29 23:02:54 evangemh Exp $
---
>  * $Id: musicout.c.patch,v 1.2 2000/06/29 23:02:54 evangemh Exp $
7d6
<  * $Log: musicout.c.patch,v $
<  * Revision 1.2  2000/06/29 23:02:54  evangemh
<  * .
<  *
92a92,118
> #include <sys/types.h>
> #include <sys/stat.h>
> #include <fcntl.h>
> 
> #ifdef SOLARIS
> #include <sys/audioio.h>
> #endif
> #ifdef LINUX
> #include <linux/soundcard.h>
> #endif
> 
> #include <rtp.h>
> #include <netinet/in.h>
> #include <event.h>
> #include <syn.h>
> #include <payload.h>
> #include <frame.h>
> 
> /* 
>  * Upcall functions needed by RTP library
>  * to extract MPEG audio frames.
>  */
> extern void out_audio();
> extern bool aligned(struct rtp *, int);
> extern int dataoffset(struct rtp *, int);
> extern mediatime_t frameduration(char *, int);
> extern int framelength(char *, int);
110,111d135
< static void GetArguments();
<   
124a149,151
> /* to implement rebuffering and resetting player on signal for testing */
> extern long totbit, offset,buf_bit_idx,buf_byte_idx;
> 
127,132c154,155
< main(argc, argv)
< int argc;
< char **argv;
< {
< /*typedef short PCM[2][3][SBLIMIT];*/
< typedef short PCM[2][SSLIMIT][SBLIMIT];
---
> int plaympeg(struct synsession *pssn, ssrc_t ssrc, int audiodev) {
>   typedef short PCM[2][SSLIMIT][SBLIMIT];
134c157
< typedef unsigned int SAM[2][3][SBLIMIT];
---
>   typedef unsigned int SAM[2][3][SBLIMIT];
136c159
< typedef double FRA[2][3][SBLIMIT];
---
>   typedef double FRA[2][3][SBLIMIT];
138c161
< typedef double VE[2][HAN_SIZE];
---
>   typedef double VE[2][HAN_SIZE];
155d177
<     Arguments_t       Arguments;
158,159c180,181
< III_scalefac_t III_scalefac;
< III_side_info_t III_side_info;
---
>   III_scalefac_t III_scalefac;
>   III_side_info_t III_side_info;
161,163c183,184
< #ifdef  MACINTOSH
<     console_options.nrows = MAC_WINDOW_SIZE;
<     argc = ccommand(&argv);
---
> #ifdef SOLARIS
>   audio_info_t		audio_info;
164a186,191
>   struct timespec	now,packtime;
>   mediatime_t		ts;
>   struct framereadparam	freadparam;
>   int			 rv, tmp;
>   unsigned int span;
>   layer *rtp_info;
179,183c206,220
<     Arguments.topSb = 0;
<     GetArguments(argc, argv, &Arguments);
<     if ((musicout = fopen(Arguments.decoded_file_name, "w+b")) == NULL) {
<           printf ("Could not create \"%s\".\n", Arguments.decoded_file_name);
<           exit(1);
---
>   /* get up the bitstream */
>   bzero((char *) &freadparam, sizeof(struct framereadparam));
>   bzero((char *) &bs, sizeof(bs));
>   bs.buf = (char *) malloc(2048);
>   bs.format = BINARY;
>   bs.mode = READ_MODE;
>   bs.eob = FALSE;
>   bs.eobs = FALSE;
> 
>   sample_frames = 0;
> 
>   while (TRUE) {
>     synctl(pssn, SYN_CTL_STREAM_GETBUFSPAN, (char *) &span, ssrc);
>     if (span == 0) {
>       synrebuffer(pssn, ssrc);
185d221
<     open_bit_stream_r(&bs, Arguments.encoded_file_name, BUFFER_SIZE);
187,191c223,231
<     if (Arguments.need_aiff)
<        if (aiff_seek_to_sound_data(musicout) == -1) {
<           printf("Could not seek to PCM sound data in \"%s\".\n",
<                  Arguments.decoded_file_name);
<           exit(1);
---
>     /*
>      * Read a frame.
>      */
>     rv = synread(pssn, ssrc, &packtime, &ts, bs.buf, 2048, &freadparam);
> 
> #ifdef LINUX
>     clock_gettime(CLOCK_REALTIME, &now);
>     if (timecmp(now, packtime) < 0) {
>       ioctl(audiodev, SNDCTL_DSP_SYNC);
192a233
> #endif
194c235,236
<     sample_frames = 0;
---
>     if (rv < 0)
>       break;
196c238,242
<     while (!end_bs(&bs)) {
---
>     bs.buf_size = rv;
>     bs.buf_byte_idx = 0;
>     bs.buf_bit_idx = 8;
>     bs.eob = FALSE;
>     bs.eobs = FALSE;
200,203d245
<        if(frameNum > 0)        /* don't want to print on 1st loop; no lay */
<           if(frameBits%bitsPerSlot)
<              fprintf(stderr,"Got %ld bits = %ld slots plus %ld\n",
<                      frameBits, frameBits/bitsPerSlot, frameBits%bitsPerSlot);
210,214d251
<           /* finally write out the buffer */
<           if (info.lay != 1) out_fifo(*pcm_sample, 3, &fr_ps, done,
<                                       musicout, &sample_frames);
<           else               out_fifo(*pcm_sample, 1, &fr_ps, done,
<                                       musicout, &sample_frames);
232d268
<        if(frameNum == 0) WriteHdr(&fr_ps, stdout);  /* printout layer/mode */
234,239c270,293
< #ifdef ESPS
< if (frameNum == 0 && Arguments.need_esps) {
< 	esps_write_header(musicout,(long) sample_frames, (double)
< 		s_freq[info.version][info.sampling_frequency] * 1000,
< 		(int) stereo, Arguments.decoded_file_name );
< } /* MI */
---
>     if(frameNum == 0) {
> 
>       rtp_info = fr_ps.header;
> 	 
>       /*
>        * Set the audiodevice's clock rate; 
>        * all other parameters should already be set.
>        */
> 
> #ifdef SOLARIS	   
>       if (ioctl(audiodev, AUDIO_GETINFO, &audio_info) < 0) {
> 	perror("ioctl1");
> 	exit(1);
>       }
> 	   
>       audio_info.play.sample_rate = (int) (s_freq[rtp_info->version][rtp_info->sampling_frequency] * 1000);
>       audio_info.play.channels = fr_ps.stereo;
>       audio_info.play.precision = 16;
>       audio_info.play.encoding = AUDIO_ENCODING_LINEAR;
> 	   
>       if (ioctl(audiodev, AUDIO_SETINFO, &audio_info) < 0) {
> 	perror("ioctl2");
> 	exit(1);
>       }
240a295,333
> #ifdef LINUX
>       if (ioctl(audiodev, SNDCTL_DSP_RESET) < 0) {
> 	perror("ioctl DSP_RESET");
> 	exit(1);
>       }
> 	   
>       tmp = APF_NETWORK;
>       if (ioctl(audiodev, SNDCTL_DSP_PROFILE, &tmp) < 0) {	
> 	perror("ioctl DSP_PROFILE");
> 	exit(1);
>       }
> 
>       if (ioctl(audiodev, SNDCTL_DSP_SYNC) < 0) {	
> 	perror("ioctl DSP_SAMPLESZ");
> 	exit(1);
>       }
> 	   
>       tmp = AFMT_S16_LE;	   
>       if (ioctl(audiodev, SNDCTL_DSP_SETFMT, &tmp) < 0) {
> 	perror("ioctl DSP_SETFMT");
> 	exit(1);
>       }
> 
>       tmp = (int) (s_freq[rtp_info->version][rtp_info->sampling_frequency] * 1000);
>       if(ioctl(audiodev, SNDCTL_DSP_SPEED, &tmp) < 0) {
> 	printf("ioctl SNDCTL_DSP_SPEED");
> 	exit(1);
>       }
> 
>       tmp = fr_ps.stereo;
>       if (ioctl(audiodev, SNDCTL_DSP_CHANNELS, &tmp) < 0) {
> 	printf("ioctl SNDCLT_DSP_CHANNELS");
> 	exit(1);
>       }
> #endif
>       printf("Audio device reconfigured.\n");
>     }
> 
>     frameNum++;
242d334
<        fprintf(stderr, "{%4lu}", frameNum++); fflush(stderr); 
270,274d361
<                 if(Arguments.topSb>0)        /* clear channels to 0 */
<                    for(j=Arguments.topSb; j<fr_ps.sblimit; ++j)
<                       for(k=0; k<stereo; ++k)
<                          (*fraction)[k][0][j] = 0;
< 
279,280c366
<                 out_fifo(*pcm_sample, 1, &fr_ps, done,
<                          musicout, &sample_frames);