I thought I was going to be able to leave 1.4 alone, but that was not the case.

I ran into some problems with G.722 in 1.4, so I have merged in all of the fixes
in this area that I have made in trunk/1.6.0, and things are happy again.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114550 65c4cc65-6c06-0410-ace0-fbb531ad65f3

formats/format_pcm.c

@@ -102,7 +102,10 @@ static struct ast_frame *pcm_read(struct ast_filestream *s, int *whennext)
 		return NULL;
 	}
 	s->fr.datalen = res;
-	*whennext = s->fr.samples = res;
+	if (s->fmt->format == AST_FORMAT_G722)
+		*whennext = s->fr.samples = res * 2;
+	else
+		*whennext = s->fr.samples = res;
 	return &s->fr;
 }
 
@@ -380,24 +383,31 @@ static int au_rewrite(struct ast_filestream *s, const char *comment)
 static int au_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
 {
 	off_t min, max, cur;
-	long offset = 0, samples;
-	
-	samples = sample_offset;
+	long offset = 0, bytes;
+
+	if (fs->fmt->format == AST_FORMAT_G722)
+		bytes = sample_offset / 2;
+	else
+		bytes = sample_offset;
+
 	min = AU_HEADER_SIZE;
 	cur = ftello(fs->f);
 	fseek(fs->f, 0, SEEK_END);
 	max = ftello(fs->f);
+
 	if (whence == SEEK_SET)
-		offset = samples + min;
+		offset = bytes + min;
 	else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
-		offset = samples + cur;
+		offset = bytes + cur;
 	else if (whence == SEEK_END)
-		offset = max - samples;
+		offset = max - bytes;
         if (whence != SEEK_FORCECUR) {
 		offset = (offset > max) ? max : offset;
 	}
+
 	/* always protect the header space. */
 	offset = (offset < min) ? min : offset;
+
 	return fseeko(fs->f, offset, SEEK_SET);
 }
 

main/channel.c

@@ -1573,7 +1573,7 @@ static int generator_force(const void *data)
 	if (!tmp || !generate)
 		return 0;
 
-	res = generate(chan, tmp, 0, 160);
+	res = generate(chan, tmp, 0, ast_format_rate(chan->writeformat & AST_FORMAT_AUDIO_MASK) / 50);
 
 	chan->generatordata = tmp;
 
@@ -1889,6 +1889,7 @@ static void ast_read_generator_actions(struct ast_channel *chan, struct ast_fram
 		void *tmp = chan->generatordata;
 		int (*generate)(struct ast_channel *chan, void *tmp, int datalen, int samples) = NULL;
 		int res;
+		int samples;
 
 		if (chan->timingfunc) {
 			if (option_debug > 1)
@@ -1897,6 +1898,15 @@ static void ast_read_generator_actions(struct ast_channel *chan, struct ast_fram
 		}
 
 		chan->generatordata = NULL;     /* reset, to let writes go through */
+
+		if (f->subclass != chan->writeformat) {
+			float factor;
+			factor = ((float) ast_format_rate(chan->writeformat)) / ((float) ast_format_rate(f->subclass));
+			samples = (int) ( ((float) f->samples) * factor );
+		} else {
+			samples = f->samples;
+		}
+
 		if (chan->generator->generate) {
 			generate = chan->generator->generate;
 		}
@@ -1909,7 +1919,7 @@ static void ast_read_generator_actions(struct ast_channel *chan, struct ast_fram
 		 * avoidance not to work in deeper functions
 		 */
 		ast_channel_unlock(chan);
-		res = generate(chan, tmp, f->datalen, f->samples);
+		res = generate(chan, tmp, f->datalen, samples);
 		ast_channel_lock(chan);
 		chan->generatordata = tmp;
 		if (res) {

main/file.c

@@ -680,11 +680,21 @@ static enum fsread_res ast_readaudio_callback(struct ast_filestream *s)
 	}
 	if (whennext != s->lasttimeout) {
 #ifdef HAVE_ZAPTEL
-		if (s->owner->timingfd > -1)
-			ast_settimeout(s->owner, whennext, ast_fsread_audio, s);
-		else
+		if (s->owner->timingfd > -1) {
+			int zap_timer_samples = whennext;
+			int rate;
+			/* whennext is in samples, but zaptel timers operate in 8 kHz samples. */
+			if ((rate = ast_format_rate(s->fmt->format)) != 8000) {
+				float factor;
+				factor = ((float) rate) / ((float) 8000.0); 
+				zap_timer_samples = (int) ( ((float) zap_timer_samples) / factor );
+			}
+			ast_settimeout(s->owner, zap_timer_samples, ast_fsread_audio, s);
+		} else
 #endif		
-			s->owner->streamid = ast_sched_add(s->owner->sched, whennext/8, ast_fsread_audio, s);
+			s->owner->streamid = ast_sched_add(s->owner->sched, 
+				whennext / (ast_format_rate(s->fmt->format) / 1000), 
+				ast_fsread_audio, s);
 		s->lasttimeout = whennext;
 		return FSREAD_SUCCESS_NOSCHED;
 	}
@@ -728,7 +738,8 @@ static enum fsread_res ast_readvideo_callback(struct ast_filestream *s)
 	}
 
 	if (whennext != s->lasttimeout) {
-		s->owner->vstreamid = ast_sched_add(s->owner->sched, whennext / 8, 
+		s->owner->vstreamid = ast_sched_add(s->owner->sched, 
+			whennext / (ast_format_rate(s->fmt->format) / 1000), 
 			ast_fsread_video, s);
 		s->lasttimeout = whennext;
 		return FSREAD_SUCCESS_NOSCHED;

main/rtp.c

@@ -1313,7 +1313,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 		/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
 		ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
 		rtp->f.ts = timestamp / 8;
-		rtp->f.len = rtp->f.samples / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 );
+		rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000);
 	} else {
 		/* Video -- samples is # of samples vs. 90000 */
 		if (!rtp->lastividtimestamp)
@@ -2795,17 +2795,28 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
 			ast_smoother_feed(rtp->smoother, _f);
 		}
 
-		while((f = ast_smoother_read(rtp->smoother)) && (f->data))
+		while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) {
+			if (f->subclass == AST_FORMAT_G722) {
+				/* G.722 is silllllllllllllly */
+				f->samples /= 2;
+			}
+
 			ast_rtp_raw_write(rtp, f, codec);
+		}
 	} else {
-	        /* Don't buffer outgoing frames; send them one-per-packet: */
+		/* Don't buffer outgoing frames; send them one-per-packet: */
 		if (_f->offset < hdrlen) {
 			f = ast_frdup(_f);
 		} else {
 			f = _f;
 		}
-		if (f->data)
+		if (f->data) {
+			if (f->subclass == AST_FORMAT_G722) {
+				/* G.722 is silllllllllllllly */
+				f->samples /= 2;
+			}
 			ast_rtp_raw_write(rtp, f, codec);
+		}
 		if (f != _f)
 			ast_frfree(f);
 	}