[asterisk-commits] russell: branch russell/jack r96173 - /team/russell/jack/apps/app_jack.c

[SVN commits to the Asterisk project]

Author: russell
Date: Wed Jan  2 22:31:49 2008
New Revision: 96173

URL: http://svn.digium.com/view/asterisk?view=rev&rev=96173
Log:
Simplify the audio path from jack back to Asterisk.  The smoother isn't really
necessary.  Now, the code just waits until there is enough data in the ringbuffer
to fill a 20 ms frame before pulling it out.

Modified:
    team/russell/jack/apps/app_jack.c

Modified: team/russell/jack/apps/app_jack.c
URL: http://svn.digium.com/view/asterisk/team/russell/jack/apps/app_jack.c?view=diff&rev=96173&r1=96172&r2=96173
==============================================================================
--- team/russell/jack/apps/app_jack.c (original)
+++ team/russell/jack/apps/app_jack.c Wed Jan  2 22:31:49 2008
@@ -77,7 +77,6 @@
 	jack_port_t *output_port;
 	jack_ringbuffer_t *input_rb;
 	jack_ringbuffer_t *output_rb;
-	struct ast_smoother *smoother;
 	void *output_resampler;
 	double output_resample_factor;
 	void *input_resampler;
@@ -196,6 +195,8 @@
 		int total_out_buf_used = 0;
 		float f_buf[nframes + 1];
 
+		memset(f_buf, 0, sizeof(f_buf));
+
 		while (total_in_buf_used < nframes) {
 			int in_buf_used;
 			int out_buf_used;
@@ -307,11 +308,6 @@
 		jack_data->output_rb = NULL;
 	}
 
-	if (jack_data->smoother) {
-		ast_smoother_free(jack_data->smoother);
-		jack_data->smoother = NULL;
-	}
-
 	if (jack_data->output_resampler) {
 		resample_close(jack_data->output_resampler);
 		jack_data->output_resampler = NULL;
@@ -331,11 +327,6 @@
 	ast_channel_lock(chan);
 	chan_name = ast_strdupa(chan->name);
 	ast_channel_unlock(chan);
-
-	if (!(jack_data->smoother = ast_smoother_new(320))) { /* 20 ms frames */
-		destroy_jack_data(jack_data);
-		return -1;
-	}
 
 	if (!(jack_data->output_rb = jack_ringbuffer_create(RINGBUFFER_SIZE))) {
 		destroy_jack_data(jack_data);
@@ -458,37 +449,38 @@
  * \brief handle jack audio
  *
  * Read data from the input ringbuffer, which is the properly resampled audio
- * that was read from the jack input port.  Feed it through a smoother and write
- * frames to the channel.
+ * that was read from the jack input port.  Write it to the channel in 20 ms frames.
  */
 static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_data)
 {	
 	short buf[160];
-	int res;	
-	struct ast_frame *fr;
+	struct ast_frame f = {
+		.frametype = AST_FRAME_VOICE,
+		.subclass = AST_FORMAT_SLINEAR,
+		.src = "JACK",
+		.data = buf,
+		.datalen = sizeof(buf),
+		.samples = ARRAY_LEN(buf),
+	};
 
 	for (;;) {
-		struct ast_frame f = {
-			.frametype = AST_FRAME_VOICE,
-			.subclass = AST_FORMAT_SLINEAR,
-			.src = "JACK",
-			.data = buf,
-		};
+		size_t res;
+
+		res = jack_ringbuffer_read_space(jack_data->input_rb);
+
+		if (res < sizeof(buf)) {
+			/* Not enough data ready for another frame, move on ... */
+			break;
+		}
 
 		res = jack_ringbuffer_read(jack_data->input_rb, (char *) buf, sizeof(buf));
 
-		if (!res)
+		if (res < sizeof(buf)) {
+			ast_log(LOG_ERROR, "Error reading from ringbuffer, even though it said there was enough data\n");
 			break;
-
-		f.datalen = res;
-		f.samples = res / sizeof(short);
-
-		ast_smoother_feed(jack_data->smoother, &f);
-	}
-
-	while ((fr = ast_smoother_read(jack_data->smoother))) {
-		ast_write(chan, fr);
-		ast_frfree(fr);
+		}
+
+		ast_write(chan, &f);
 	}
 }