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-rw-r--r--mp3streamout~.c291
1 files changed, 158 insertions, 133 deletions
diff --git a/mp3streamout~.c b/mp3streamout~.c
index 88fec11..5f089dc 100644
--- a/mp3streamout~.c
+++ b/mp3streamout~.c
@@ -70,8 +70,8 @@ extern void print_header_compact( struct frame* fr );
extern int head_check( unsigned long head, int check_layer );
#define MY_MP3_MALLOC_IN_SIZE 65536
- /* max size taken from lame readme */
-#define MY_MP3_MALLOC_OUT_SIZE 1.25*MY_MP3_MALLOC_IN_SIZE+7200
+/* max size taken from lame readme */
+#define MY_MP3_MALLOC_OUT_SIZE 1.25*MY_MP3_MALLOC_IN_SIZE+7200
#define MAXDATARATE 320 /* maximum mp3 data rate is 320kbit/s */
#define STRBUF_SIZE 32
@@ -84,12 +84,12 @@ typedef struct _mp3streamout
{
t_object x_obj;
- /* LAME stuff */
+ /* LAME stuff */
int x_lame; /* info about encoder status */
int x_lamechunk; /* chunk size for LAME encoder */
int x_mp3size; /* number of returned mp3 samples */
- /* buffer stuff */
+ /* buffer stuff */
unsigned short x_inp; /* in position for buffer */
unsigned short x_outp; /* out position for buffer*/
short *x_mp3inbuf; /* data to be sent to LAME */
@@ -98,13 +98,13 @@ typedef struct _mp3streamout
int x_bytesbuffered; /* number of unprocessed bytes in buffer */
int x_start;
- /* mp3 format stuff */
+ /* mp3 format stuff */
int x_samplerate;
int x_bitrate; /* bitrate of mp3 stream */
int x_mp3mode; /* mode (mono, joint stereo, stereo, dual mono) */
int x_mp3quality; /* quality of encoding */
- /* connection data */
+ /* connection data */
int x_fd; /* info about connection status */
int x_outpackets; /* mp3 packets sent */
@@ -114,7 +114,7 @@ typedef struct _mp3streamout
} t_mp3streamout;
- /* encode PCM data to mp3 stream */
+/* encode PCM data to mp3 stream */
static void mp3streamout_encode(t_mp3streamout *x)
{
unsigned short i, wp;
@@ -131,11 +131,11 @@ static void mp3streamout_encode(t_mp3streamout *x)
return;
}
- /* on start/reconnect set outpoint that it not interferes with inpoint */
+ /* on start/reconnect set outpoint that it not interferes with inpoint */
if(x->x_start == -1)
{
post("mp3streamout~: initializing buffers");
- /* we try to keep 2.5 times the data the encoder needs in the buffer */
+ /* we try to keep 2.5 times the data the encoder needs in the buffer */
if(x->x_inp > (2 * x->x_lamechunk))
{
x->x_outp = (short) x->x_inp - (2.5 * x->x_lamechunk);
@@ -150,10 +150,10 @@ static void mp3streamout_encode(t_mp3streamout *x)
i = MY_MP3_MALLOC_IN_SIZE - x->x_outp;
- /* read from buffer */
- if(x->x_lamechunk <= i)
+ /* read from buffer */
+ if(x->x_lamechunk <= i)
{
- /* enough data until end of buffer */
+ /* enough data until end of buffer */
for(n = 0; n < x->x_lamechunk; n++) /* fill encode buffer */
{
x->x_mp3inbuf[n] = x->x_buffer[n + x->x_outp];
@@ -174,15 +174,15 @@ static void mp3streamout_encode(t_mp3streamout *x)
x->x_outp = x->x_lamechunk - i;
}
- /* encode mp3 data */
+ /* encode mp3 data */
- x->x_mp3size = lame_encode_buffer_interleaved(x->lgfp, x->x_mp3inbuf,
- x->x_lamechunk/lame_get_num_channels(x->lgfp),
+ x->x_mp3size = lame_encode_buffer_interleaved(x->lgfp, x->x_mp3inbuf,
+ x->x_lamechunk/lame_get_num_channels(x->lgfp),
x->x_mp3outbuf, MY_MP3_MALLOC_OUT_SIZE);
x->x_mp3size+=lame_encode_flush( x->lgfp, x->x_mp3outbuf+x->x_mp3size, MY_MP3_MALLOC_OUT_SIZE-x->x_mp3size );
// post( "mp3streamout~ : encoding returned %d frames", x->x_mp3size );
- /* check result */
+ /* check result */
if(x->x_mp3size<0)
{
lame_close( x->lgfp );
@@ -191,21 +191,21 @@ static void mp3streamout_encode(t_mp3streamout *x)
}
}
- /* stream mp3 to the peer */
+/* stream mp3 to the peer */
static void mp3streamout_stream(t_mp3streamout *x)
{
- int count = -1, i;
+ int count = -1, i;
struct frame hframe;
/* header needs to be included in each packet */
for( i=0; i<x->x_mp3size; i++ )
{
- // track valid data
- if ( head_check( *((unsigned long*)x->x_mp3outbuf+i), 3 ) )
- {
- // post( "valid header emitted @ %d (byte %d)", i, i*sizeof(unsigned long) );
- }
+ // track valid data
+ if ( head_check( *((unsigned long*)x->x_mp3outbuf+i), 3 ) )
+ {
+ // post( "valid header emitted @ %d (byte %d)", i, i*sizeof(unsigned long) );
+ }
}
count = send(x->x_fd, x->x_mp3outbuf, x->x_mp3size, MSG_NOSIGNAL);
@@ -221,23 +221,23 @@ static void mp3streamout_stream(t_mp3streamout *x)
#endif
x->x_fd = -1;
outlet_float(x->x_obj.ob_outlet, 0);
- }
+ }
else
{
x->x_outpackets++;
if ( x->x_outpackets%100 == 0 )
{
- // post( "mp3streamout~ : emitted %d bytes (packets = %d)", count, x->x_outpackets );
+ // post( "mp3streamout~ : emitted %d bytes (packets = %d)", count, x->x_outpackets );
}
}
if((count > 0)&&(count != x->x_mp3size))
{
- error("mp3streamout~: %d bytes skipped", x->x_mp3size - count);
+ error("mp3streamout~: %d bytes skipped", x->x_mp3size - count);
}
}
-
- /* buffer data as channel interleaved PCM */
+
+/* buffer data as channel interleaved PCM */
static t_int *mp3streamout_perform(t_int *w)
{
t_float *in1 = (t_float *)(w[1]); /* left audio inlet */
@@ -247,76 +247,94 @@ static t_int *mp3streamout_perform(t_int *w)
unsigned short i,wp;
float in;
- /* copy the data into the buffer */
+ /* copy the data into the buffer */
i = MY_MP3_MALLOC_IN_SIZE - x->x_inp; /* space left at the end of buffer */
-
+
n *= 2; /* two channels go into one buffer */
- if( n <= i )
+ if( n <= i )
{
/* the place between inp and MY_MP3_MALLOC_IN_SIZE */
/* is big enough to hold the data */
- for(wp = 0; wp < n; wp++)
+ for(wp = 0; wp < n; wp++)
+ {
+ if(wp%2)
+ {
+ in = *(in2++); /* right channel / inlet */
+ }
+ else
+ {
+ in = *(in1++); /* left channel / inlet */
+ }
+ if (in > 1.0)
+ {
+ in = 1.0;
+ }
+ if (in < -1.0)
{
- if(wp%2)
- {
- in = *(in2++); /* right channel / inlet */
- }
- else
- {
- in = *(in1++); /* left channel / inlet */
- }
- if (in > 1.0) { in = 1.0; }
- if (in < -1.0) { in = -1.0; }
- x->x_buffer[wp + x->x_inp] = (short) (32767.0 * in);
+ in = -1.0;
}
- x->x_inp += n; /* n more samples written to buffer */
- }
- else
+ x->x_buffer[wp + x->x_inp] = (short) (32767.0 * in);
+ }
+ x->x_inp += n; /* n more samples written to buffer */
+ }
+ else
{
- /* the place between inp and MY_MP3_MALLOC_IN_SIZE is not */
- /* big enough to hold the data */
- /* writing will take place in two turns, one from */
- /* x->x_inp -> MY_MP3_MALLOC_IN_SIZE, then from 0 on */
+ /* the place between inp and MY_MP3_MALLOC_IN_SIZE is not */
+ /* big enough to hold the data */
+ /* writing will take place in two turns, one from */
+ /* x->x_inp -> MY_MP3_MALLOC_IN_SIZE, then from 0 on */
- for(wp = 0; wp < i; wp++) /* fill up to end of buffer */
+ for(wp = 0; wp < i; wp++) /* fill up to end of buffer */
+ {
+ if(wp%2)
+ {
+ in = *(in2++);
+ }
+ else
+ {
+ in = *(in1++);
+ }
+ if (in > 1.0)
+ {
+ in = 1.0;
+ }
+ if (in < -1.0)
+ {
+ in = -1.0;
+ }
+ x->x_buffer[wp + x->x_inp] = (short) (32767.0 * in);
+ }
+ for(wp = i; wp < n; wp++) /* write rest at start of buffer */
+ {
+ if(wp%2)
+ {
+ in = *(in2++);
+ }
+ else
+ {
+ in = *(in1++);
+ }
+ if (in > 1.0)
{
- if(wp%2)
- {
- in = *(in2++);
- }
- else
- {
- in = *(in1++);
- }
- if (in > 1.0) { in = 1.0; }
- if (in < -1.0) { in = -1.0; }
- x->x_buffer[wp + x->x_inp] = (short) (32767.0 * in);
+ in = 1.0;
}
- for(wp = i; wp < n; wp++) /* write rest at start of buffer */
+ if (in < -1.0)
{
- if(wp%2)
- {
- in = *(in2++);
- }
- else
- {
- in = *(in1++);
- }
- if (in > 1.0) { in = 1.0; }
- if (in < -1.0) { in = -1.0; }
- x->x_buffer[wp - i] = (short) (32767.0 * in);
+ in = -1.0;
}
- x->x_inp = n - i; /* new writeposition in buffer */
+ x->x_buffer[wp - i] = (short) (32767.0 * in);
+ }
+ x->x_inp = n - i; /* new writeposition in buffer */
}
if((x->x_fd >= 0)&&(x->x_lame >= 0))
- {
- /* count buffered samples when things are running */
+ {
+ /* count buffered samples when things are running */
x->x_bytesbuffered += n;
- /* encode and send to the peer */
+ /* encode and send to the peer */
if(x->x_bytesbuffered > x->x_lamechunk)
{
mp3streamout_encode(x); /* encode to mp3 */
@@ -336,7 +354,7 @@ static void mp3streamout_dsp(t_mp3streamout *x, t_signal **sp)
dsp_add(mp3streamout_perform, 4, sp[0]->s_vec, sp[1]->s_vec, x, sp[0]->s_n);
}
- /* initialize the lame library */
+/* initialize the lame library */
static void mp3streamout_tilde_lame_init(t_mp3streamout *x)
{
int ret;
@@ -357,11 +375,11 @@ static void mp3streamout_tilde_lame_init(t_mp3streamout *x)
}
#endif
{
- const char *lameVersion = get_lame_version();
- verbose(0, "mp3streamout~ : using lame version : %s", lameVersion );
+ const char *lameVersion = get_lame_version();
+ verbose(0, "mp3streamout~ : using lame version : %s", lameVersion );
}
- /* setting lame parameters */
+ /* setting lame parameters */
lame_set_num_channels( x->lgfp, 2);
lame_set_in_samplerate( x->lgfp, sys_getsr() );
lame_set_out_samplerate( x->lgfp, x->x_samplerate );
@@ -374,26 +392,29 @@ static void mp3streamout_tilde_lame_init(t_mp3streamout *x)
lame_set_disable_reservoir( x->lgfp, 0 );
lame_set_padding_type( x->lgfp, PAD_NO );
ret = lame_init_params( x->lgfp );
- if ( ret<0 ) {
- post( "mp3streamout~ : error : lame params initialization returned : %d", ret );
- } else {
- x->x_lame=1;
- /* magic formula copied from windows dll for MPEG-I */
- x->x_lamechunk = 2*1152;
-
- post( "mp3streamout~ : lame initialization done. (%d)", x->x_lame );
+ if ( ret<0 )
+ {
+ post( "mp3streamout~ : error : lame params initialization returned : %d", ret );
+ }
+ else
+ {
+ x->x_lame=1;
+ /* magic formula copied from windows dll for MPEG-I */
+ x->x_lamechunk = 2*1152;
+
+ post( "mp3streamout~ : lame initialization done. (%d)", x->x_lame );
}
lame_init_bitstream( x->lgfp );
}
- /* connect to the peer */
+/* connect to the peer */
static void mp3streamout_connect(t_mp3streamout *x, t_symbol *hostname, t_floatarg fportno)
{
struct sockaddr_in csocket;
struct hostent *hp;
int portno = fportno; /* get port from message box */
- /* variables used for communication with the peer */
+ /* variables used for communication with the peer */
const char *buf = 0;
char resp[STRBUF_SIZE];
unsigned int len;
@@ -418,7 +439,7 @@ static void mp3streamout_connect(t_mp3streamout *x, t_symbol *hostname, t_floata
return;
}
- /* connect socket using hostname provided in command line */
+ /* connect socket using hostname provided in command line */
csocket.sin_family = AF_INET;
hp = gethostbyname(hostname->s_name);
if (hp == 0)
@@ -433,10 +454,10 @@ static void mp3streamout_connect(t_mp3streamout *x, t_symbol *hostname, t_floata
}
memcpy((char *)&csocket.sin_addr, (char *)hp->h_addr, hp->h_length);
- /* assign client port number */
+ /* assign client port number */
csocket.sin_port = htons((unsigned short)portno);
- /* try to connect. */
+ /* try to connect. */
post("mp3streamout~: connecting to port %d", portno);
if (connect(sockfd, (struct sockaddr *) &csocket, sizeof (csocket)) < 0)
{
@@ -458,15 +479,16 @@ static void mp3streamout_connect(t_mp3streamout *x, t_symbol *hostname, t_floata
}
- /* close connection to the peer */
+/* close connection to the peer */
static void mp3streamout_disconnect(t_mp3streamout *x)
{
int err = -1;
if(x->x_lame >= 0)
{
- /* ignore remaining bytes */
- if ( x->x_mp3size = lame_encode_flush( x->lgfp, x->x_mp3outbuf, 0) < 0 ) {
+ /* ignore remaining bytes */
+ if ( x->x_mp3size = lame_encode_flush( x->lgfp, x->x_mp3outbuf, 0) < 0 )
+ {
post( "mp3streamout~ : warning : remaining encoded bytes" );
}
lame_close( x->lgfp );
@@ -487,43 +509,46 @@ static void mp3streamout_disconnect(t_mp3streamout *x)
}
}
- /* settings for mp3 encoding */
+/* settings for mp3 encoding */
static void mp3streamout_mpeg(t_mp3streamout *x, t_floatarg fbitrate,
- t_floatarg fmode, t_floatarg fquality)
+ t_floatarg fmode, t_floatarg fquality)
{
if ( fbitrate != 32 && fbitrate != 40 && fbitrate != 48 && fbitrate != 56 &&
- fbitrate != 64 && fbitrate != 80 && fbitrate != 96 && fbitrate != 112 &&
- fbitrate != 128 && fbitrate != 160 && fbitrate != 192 && fbitrate != 224 &&
- fbitrate != 256 && fbitrate != 320 ) {
- post( "mp3streamout~ : wrong bitrate." );
- return;
+ fbitrate != 64 && fbitrate != 80 && fbitrate != 96 && fbitrate != 112 &&
+ fbitrate != 128 && fbitrate != 160 && fbitrate != 192 && fbitrate != 224 &&
+ fbitrate != 256 && fbitrate != 320 )
+ {
+ post( "mp3streamout~ : wrong bitrate." );
+ return;
}
- if ( fmode <0 || fmode>2 ) {
- post( "mp3streamout~ : wrong mp3 mode." );
- if ( fmode == 3 )
- {
- post( "mp3streamout~ : mone is not supported by streamout~ for now." );
- }
- return;
+ if ( fmode <0 || fmode>2 )
+ {
+ post( "mp3streamout~ : wrong mp3 mode." );
+ if ( fmode == 3 )
+ {
+ post( "mp3streamout~ : mone is not supported by streamout~ for now." );
+ }
+ return;
}
/* there is a bug in lame 3.92 and quality below 5 will not work */
/* WAIT FOR A FIX */
- if ( fquality <5 || fquality>9 ) {
- post( "mp3streamout~ : wrong quality." );
- return;
+ if ( fquality <5 || fquality>9 )
+ {
+ post( "mp3streamout~ : wrong quality." );
+ return;
}
x->x_bitrate = fbitrate;
x->x_mp3mode = fmode;
x->x_mp3quality = (int)fquality;
post("mp3streamout~: setting mp3 stream to %dHz, %dkbit/s, mode %d, quality %d",
- x->x_samplerate, x->x_bitrate, x->x_mp3mode, x->x_mp3quality);
+ x->x_samplerate, x->x_bitrate, x->x_mp3mode, x->x_mp3quality);
mp3streamout_tilde_lame_init(x);
}
- /* print settings */
+/* print settings */
static void mp3streamout_print(t_mp3streamout *x)
{
- const char * buf = 0;
+ const char * buf = 0;
verbose(0, mp3streamout_version);
post(" LAME mp3 settings:\n"
@@ -531,18 +556,18 @@ static void mp3streamout_print(t_mp3streamout *x)
" bitrate: %d kbit/s", x->x_samplerate, x->x_bitrate);
switch(x->x_mp3mode)
{
- case 0 :
- buf = "stereo";
- break;
- case 1 :
- buf = "joint stereo";
- break;
- case 2 :
- buf = "dual channel";
- break;
- case 3 :
- buf = "mono";
- break;
+ case 0 :
+ buf = "stereo";
+ break;
+ case 1 :
+ buf = "joint stereo";
+ break;
+ case 2 :
+ buf = "dual channel";
+ break;
+ case 3 :
+ buf = "mono";
+ break;
}
post(" mode: %s\n"
" quality: %d", buf, x->x_mp3quality);
@@ -557,8 +582,8 @@ static void mp3streamout_print(t_mp3streamout *x)
}
}
- /* clean up */
-static void mp3streamout_free(t_mp3streamout *x)
+/* clean up */
+static void mp3streamout_free(t_mp3streamout *x)
{
if(x->x_lame >= 0)
@@ -606,7 +631,7 @@ void mp3streamout_tilde_setup(void)
{
verbose(0, mp3streamout_version);
mp3streamout_class = class_new(gensym("mp3streamout~"), (t_newmethod)mp3streamout_new, (t_method)mp3streamout_free,
- sizeof(t_mp3streamout), 0, 0);
+ sizeof(t_mp3streamout), 0, 0);
CLASS_MAINSIGNALIN(mp3streamout_class, t_mp3streamout, x_f );
class_addmethod(mp3streamout_class, (t_method)mp3streamout_dsp, gensym("dsp"), 0);
class_addmethod(mp3streamout_class, (t_method)mp3streamout_connect, gensym("connect"), A_SYMBOL, A_FLOAT, 0);
generated by cgit v1.2.1 (git 2.18.0) at 2024-05-10 13:50:31 +0000