[audio] Clean up jack and alsa dependencies

I had forgotten to test with shared libs and it turns out jack and alsa
were directly accessing symbols in the renderer (and in jack's case,
linking in a duplicate of the renderer).

Fixes #16.

include/snd_internal.h

@@ -77,6 +77,7 @@ struct snd_s {
 	int				frames;					//!< frames in buffer
 											//!< 1 frame = channels samples
 	int				submission_chunk;		//!< don't mix less than this #
+	unsigned        paintedtime;			//!< sound clock in samples
 	int				framepos;				//!< position of dma cursor
 	unsigned char	*buffer;				//!< destination for mixed sound
 	/** Transfer mixed samples to the output.
@@ -90,6 +91,9 @@ struct snd_s {
 	/** Optional data for the xfer function.
 	*/
 	void            *xfer_data;
+
+	void      (*finish_channels) (void);
+	void      (*paint_channels) (struct snd_s *snd, unsigned endtime);
 };
 
 /** Describes the sound data.
@@ -318,6 +322,12 @@ struct channel_s *SND_AllocChannel (snd_t *snd);
 */
 void SND_ChannelStop (snd_t *snd, channel_t *chan);
 
+/** Mark all channels as no longer in use.
+
+	For use by asynchronous output drivers.
+*/
+void SND_FinishChannels (void);
+
 /** Scan channels looking for stopped channels.
 	\param wait	if true, wait for the channels to be done. if false, force the
 				channels to be done. true is for threaded, false for
@@ -397,9 +407,6 @@ void SND_LocalSound (snd_t *snd, const char *s);
 	\ingroup sound_render
 */
 ///@{
-/** sound clock in samples
-*/
-extern unsigned snd_paintedtime;
 
 /** Mix all active channels into the output buffer.
 	\param endtime	sample time until when to mix

libs/audio/renderer/snd_channels.c

@@ -183,6 +183,18 @@ SND_ScanChannels (snd_t *snd, int wait)
 	}
 }
 
+void
+SND_FinishChannels (void)
+{
+	int         i;
+	channel_t  *ch;
+
+	for (i = 0; i < MAX_CHANNELS; i++) {
+		ch = &snd_channels[i];
+		ch->done = ch->stop = 1;
+	}
+}
+
 void
 SND_StopAllSounds (snd_t *snd)
 {

libs/audio/renderer/snd_dma.c

@@ -70,7 +70,10 @@ static general_data_t plugin_info_general_data;
 static snd_output_funcs_t *snd_output_funcs;
 static snd_output_data_t *snd_output_data;
 
-static snd_t    snd;
+static snd_t    snd = {
+	.finish_channels = SND_FinishChannels,
+	.paint_channels  = SND_PaintChannels,
+};
 static int      snd_shutdown = 0;
 
 static void
@@ -83,7 +86,7 @@ s_xfer_paint_buffer (snd_t *snd, portable_samplepair_t *paintbuffer, int count,
 	p = (float *) paintbuffer;
 	count *= snd->channels;
 	out_max = (snd->frames * snd->channels) - 1;
-	out_idx = snd_paintedtime * snd->channels;
+	out_idx = snd->paintedtime * snd->channels;
 	while (out_idx > out_max)
 		out_idx -= out_max + 1;
 	step = 3 - snd->channels;
@@ -164,10 +167,10 @@ s_get_soundtime (void)
 	if (framepos < oldframepos) {
 		buffers++;						// buffer wrapped
 
-		if (snd_paintedtime > 0x40000000) {	// time to chop things off to avoid
+		if (snd.paintedtime > 0x40000000) {	// time to chop things off to avoid
 			// 32 bit limits
 			buffers = 0;
-			snd_paintedtime = frames;
+			snd.paintedtime = frames;
 			s_stop_all_sounds ();
 		}
 	}
@@ -188,9 +191,9 @@ s_update_ (void)
 	s_get_soundtime ();
 
 	// check to make sure that we haven't overshot
-	if (snd_paintedtime < soundtime) {
+	if (snd.paintedtime < soundtime) {
 //		Sys_Printf ("S_Update_ : overflow\n");
-		snd_paintedtime = soundtime;
+		snd.paintedtime = soundtime;
 	}
 	// mix ahead of current position
 	endtime = soundtime + snd_mixahead->value * snd.speed;
@@ -492,6 +495,10 @@ static plugin_funcs_t plugin_info_funcs = {
 	.snd_render = &plugin_info_render_funcs,
 };
 
+snd_render_data_t snd_render_data = {
+	.paintedtime = &snd.paintedtime,
+};
+
 static plugin_data_t plugin_info_data = {
 	.general = &plugin_info_general_data,
 	.snd_render = &snd_render_data,

libs/audio/renderer/snd_mem.c

@@ -51,14 +51,6 @@
 
 #define SAMPLE_GAP	4
 
-snd_render_data_t snd_render_data = {
-	0,
-	0,
-	0,
-	&snd_paintedtime,
-	0,
-};
-
 static sfxbuffer_t *
 snd_fail (sfx_t *sfx)
 {

libs/audio/renderer/snd_mix.c

@@ -50,8 +50,6 @@
 
 cvar_t     *snd_volume;
 
-unsigned    snd_paintedtime;				// sample PAIRS
-
 portable_samplepair_t snd_paintbuffer[PAINTBUFFER_SIZE * 2];
 static int  max_overpaint;				// number of extra samples painted
 										// due to phase shift
@@ -118,11 +116,11 @@ SND_PaintChannels (snd_t *snd, unsigned endtime)
 		snd_paintbuffer[i].right = 0;
 	}
 
-	while (snd_paintedtime < endtime) {
+	while (snd->paintedtime < endtime) {
 		// if snd_paintbuffer is smaller than DMA buffer
 		end = endtime;
-		if (end - snd_paintedtime > PAINTBUFFER_SIZE)
-			end = snd_paintedtime + PAINTBUFFER_SIZE;
+		if (end - snd->paintedtime > PAINTBUFFER_SIZE)
+			end = snd->paintedtime + PAINTBUFFER_SIZE;
 
 		max_overpaint = 0;
 
@@ -144,9 +142,9 @@ SND_PaintChannels (snd_t *snd, unsigned endtime)
 			}
 
 			if (!ch->end)
-				ch->end = snd_paintedtime + sfx->length - ch->pos;
+				ch->end = snd->paintedtime + sfx->length - ch->pos;
 
-			ltime = snd_paintedtime;
+			ltime = snd->paintedtime;
 
 			while (ltime < end) {		// paint up to end
 				count = ((ch->end < end) ? ch->end : end) - ltime;
@@ -171,15 +169,15 @@ SND_PaintChannels (snd_t *snd, unsigned endtime)
 		}
 
 		// transfer out according to DMA format
-		snd->xfer (snd, snd_paintbuffer, end - snd_paintedtime,
+		snd->xfer (snd, snd_paintbuffer, end - snd->paintedtime,
 				   snd_volume->value);
 
-		memmove (snd_paintbuffer, snd_paintbuffer + end - snd_paintedtime,
+		memmove (snd_paintbuffer, snd_paintbuffer + end - snd->paintedtime,
 				 max_overpaint * sizeof (snd_paintbuffer[0]));
 		memset (snd_paintbuffer + max_overpaint, 0, sizeof (snd_paintbuffer)
 				- max_overpaint * sizeof (snd_paintbuffer[0]));
 
-		snd_paintedtime = end;
+		snd->paintedtime = end;
 	}
 }
 

libs/audio/targets/Makemodule.am

@@ -21,14 +21,15 @@ libs_audio_targets_snd_output_sdl_la_SOURCES=		libs/audio/targets/snd_sdl.c
 
 libs_audio_targets_snd_output_alsa_la_LDFLAGS=		$(plugin_ldflags)
 libs_audio_targets_snd_output_alsa_la_LIBADD=		$(snd_deps)
-libs_audio_targets_snd_output_alsa_la_DEPENDENCIES=$(snd_deps)
+libs_audio_targets_snd_output_alsa_la_DEPENDENCIES=	$(snd_deps)
 libs_audio_targets_snd_output_alsa_la_CFLAGS=		$(ALSA_CFLAGS)
 libs_audio_targets_snd_output_alsa_la_SOURCES=		libs/audio/targets/snd_alsa.c
 
-libs_audio_targets_snd_output_jack_la_LDFLAGS=			$(plugin_ldflags)
-libs_audio_targets_snd_output_jack_la_SOURCES=			libs/audio/targets/snd_jack.c $(snd_common) $(format_src)
-libs_audio_targets_snd_output_jack_la_LIBADD=			$(snd_libs) $(format_libs) $(JACK_LIBS)
+libs_audio_targets_snd_output_jack_la_LDFLAGS=		$(plugin_ldflags)
+libs_audio_targets_snd_output_jack_la_LIBADD=		$(snd_deps) $(JACK_LIBS)
 libs_audio_targets_snd_output_jack_la_DEPENDENCIES=	$(snd_libs)
+libs_audio_targets_snd_output_jack_la_CFLAGS=		$(JACK_CFLAGS)
+libs_audio_targets_snd_output_jack_la_SOURCES=		libs/audio/targets/snd_jack.c
 
 libs_audio_targets_snd_output_oss_la_LDFLAGS=		$(plugin_ldflags)
 libs_audio_targets_snd_output_oss_la_LIBADD=		$(snd_deps) $(OSS_LIBS)

libs/audio/targets/snd_alsa.c

@@ -202,7 +202,7 @@ alsa_xfer (snd_t *snd, portable_samplepair_t *paintbuffer, int count,
 	p = (float *) paintbuffer;
 	count *= snd->channels;
 	out_max = (snd->frames * snd->channels) - 1;
-	out_idx = snd_paintedtime * snd->channels;
+	out_idx = snd->paintedtime * snd->channels;
 	while (out_idx > out_max)
 		out_idx -= out_max + 1;
 	step = 3 - snd->channels;
@@ -288,7 +288,7 @@ alsa_process (snd_pcm_t *pcm, snd_t *snd)
 			ret = 0;
 		}
 		snd->buffer = packet.areas[0].addr;
-		SND_PaintChannels (snd, snd_paintedtime + packet.nframes);
+		snd->paint_channels (snd, snd->paintedtime + packet.nframes);
 		if ((res = qfsnd_pcm_mmap_commit (pcm, packet.offset,
 										  packet.nframes)) < 0
 			|| (snd_pcm_uframes_t) res != packet.nframes) {

libs/audio/targets/snd_jack.c

@@ -59,18 +59,6 @@ static cvar_t  *snd_jack_server;
 
 static int s_jack_connect (snd_t *snd);
 
-static void
-s_finish_channels (void)
-{
-	int         i;
-	channel_t  *ch;
-
-	for (i = 0; i < MAX_CHANNELS; i++) {
-		ch = &snd_channels[i];
-		ch->done = ch->stop = 1;
-	}
-}
-
 static void
 s_update (snd_t *snd)
 {
@@ -85,7 +73,7 @@ s_update (snd_t *snd)
 		if (!snd_shutdown) {
 			if (now - snd_alive_time > 1.0) {
 				Sys_Printf ("jackd client thread seems to have died\n");
-				s_finish_channels ();
+				snd->finish_channels ();
 				snd_shutdown = 1;
 			}
 		}
@@ -162,8 +150,8 @@ snd_jack_xfer (snd_t *snd, portable_samplepair_t *paintbuffer, int count,
 	}
 	for (i = 0; i < count; i++) {
 		/* max is +/- 1.0. need to implement clamping. */
-		*output[0]++ = volume * snd_paintbuffer[i].left;
-		*output[1]++ = volume * snd_paintbuffer[i].right;
+		*output[0]++ = volume * paintbuffer[i].left;
+		*output[1]++ = volume * paintbuffer[i].right;
 	}
 }
 
@@ -176,15 +164,17 @@ snd_jack_process (jack_nframes_t nframes, void *arg)
 	snd_alive = 1;
 	for (i = 0; i < 2; i++)
 		output[i] = (float *) jack_port_get_buffer (jack_out[i], nframes);
-	SND_PaintChannels (snd, snd_paintedtime + nframes);
+	snd->paint_channels (snd, snd->paintedtime + nframes);
 	return 0;
 }
 
 static void
 snd_jack_shutdown (void *arg)
 {
+	snd_t      *snd = arg;
+
 	snd_shutdown = 1;
-	s_finish_channels ();
+	snd->finish_channels ();
 }
 
 static void
@@ -217,12 +207,13 @@ s_jack_connect (snd_t *snd)
 	if (jack_set_xrun_callback (jack_handle, snd_jack_xrun, 0))
 		Sys_Printf ("Could not set xrun callback\n");
 	jack_set_process_callback (jack_handle, snd_jack_process, snd);
-	jack_on_shutdown (jack_handle, snd_jack_shutdown, 0);
+	jack_on_shutdown (jack_handle, snd_jack_shutdown, snd);
 	for (i = 0; i < 2; i++)
 		jack_out[i] = jack_port_register (jack_handle, va (0, "out_%d", i + 1),
 										  JACK_DEFAULT_AUDIO_TYPE,
 										  JackPortIsOutput, 0);
 	snd->speed = jack_get_sample_rate (jack_handle);
+	snd->channels = 2;
 	s_jack_activate ();
 	sound_started = 1;
 	Sys_Printf ("Connected to JACK: %d Sps\n", snd->speed);