[Asterisk-code-review] app_amd: Fixed timeout issue (asterisk[master])

[Kevin Harwell]

Kevin Harwell has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/13252 )

Change subject: app_amd: Fixed timeout issue
......................................................................

app_amd: Fixed timeout issue

ASTERISK_28143 attempted to fix an issue where calls with no audio would never
timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its
calculations. Unfortunately these frames seem to show up at irregular time
intervals. This resulted in app_amd returning prematurely most of the time.

* Removed AST_FRAME_NULL from the calculations
* Added a check to see how much time has actually passed since app_amd began

ASTERISK-28608

Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42
---
M apps/app_amd.c
1 file changed, 23 insertions(+), 1 deletion(-)

Approvals:
  Kevin Harwell: Looks good to me, but someone else must approve; Approved for Submit
  Joshua Colp: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved



diff --git a/apps/app_amd.c b/apps/app_amd.c
index 1a0af3f..39d0b79 100644
--- a/apps/app_amd.c
+++ b/apps/app_amd.c
@@ -162,8 +162,10 @@
 static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 {
 	int res = 0;
+	int audioFrameCount = 0;
 	struct ast_frame *f = NULL;
 	struct ast_dsp *silenceDetector = NULL;
+	struct timeval amd_tvstart;
 	int dspsilence = 0, framelength = 0;
 	RAII_VAR(struct ast_format *, readFormat, NULL, ao2_cleanup);
 	int inInitialSilence = 1;
@@ -275,6 +277,9 @@
 	/* Set silence threshold to specified value */
 	ast_dsp_set_threshold(silenceDetector, silenceThreshold);
 
+	/* Set our start time so we can tie the loop to real world time and not RTP updates */
+	amd_tvstart = ast_tvnow();
+
 	/* Now we go into a loop waiting for frames from the channel */
 	while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
 		int ms = 0;
@@ -293,7 +298,24 @@
 			break;
 		}
 
-		if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_NULL || f->frametype == AST_FRAME_CNG) {
+		/* Check to make sure we haven't gone over our real-world timeout in case frames get stalled for whatever reason */
+		if ( (ast_tvdiff_ms(ast_tvnow(), amd_tvstart)) > totalAnalysisTime ) {
+			ast_frfree(f);
+			strcpy(amdStatus , "NOTSURE");
+			if ( audioFrameCount == 0 ) {
+				ast_verb(3, "AMD: Channel [%s]. No audio data received in [%d] seconds.\n", ast_channel_name(chan), totalAnalysisTime);
+				sprintf(amdCause , "NOAUDIODATA-%d", iTotalTime);
+				break;
+			}
+			ast_verb(3, "AMD: Channel [%s]. Timeout...\n", ast_channel_name(chan));
+			sprintf(amdCause , "TOOLONG-%d", iTotalTime);
+			break;
+		}
+
+		if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_CNG) {
+			/* keep track of the number of audio frames we get */
+			audioFrameCount++;
+
 			/* Figure out how long the frame is in milliseconds */
 			if (f->frametype == AST_FRAME_VOICE) {
 				framelength = (ast_codec_samples_count(f) / DEFAULT_SAMPLES_PER_MS);

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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42
Gerrit-Change-Number: 13252
Gerrit-PatchSet: 1
Gerrit-Owner: Michael Cargile <mikec at vicidial.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-MessageType: merged
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