TT#31405 handle non-media pseudo-codecs (telephone-event)

Change-Id: I10a0156d500f49941d7d3d558e54e3c198638c4a

daemon/codec.c

@@ -140,7 +140,7 @@ void codec_handlers_update(struct call_media *receiver, struct call_media *sink)
 	for (GList *l = sink->codecs_prefs_send.head; l; l = l->next) {
 		struct rtp_payload_type *pt = l->data;
 		__ensure_codec_def(pt);
-		if (!pt->codec_def) // not supported, next
+		if (!pt->codec_def || pt->codec_def->avcodec_id == -1) // not supported, next
 			continue;
 		ilog(LOG_DEBUG, "Default sink codec is " STR_FORMAT, STR_FMT(&pt->encoding));
 		pref_dest_codec = pt;
@@ -175,9 +175,11 @@ void codec_handlers_update(struct call_media *receiver, struct call_media *sink)
 				continue;
 			}
 
-			ilog(LOG_DEBUG, "Accepting offered codec " STR_FORMAT " due to transcoding",
-					STR_FMT(&pt->encoding));
-			MEDIA_SET(receiver, TRANSCODE);
+			if (pt->codec_def->avcodec_id != -1) {
+				ilog(LOG_DEBUG, "Accepting offered codec " STR_FORMAT " due to transcoding",
+						STR_FMT(&pt->encoding));
+				MEDIA_SET(receiver, TRANSCODE);
+			}
 
 			// we need a new pt entry
 			pt = __rtp_payload_type_copy(pt);
@@ -225,6 +227,12 @@ void codec_handlers_update(struct call_media *receiver, struct call_media *sink)
 		// check our own support for this codec
 		__ensure_codec_def(pt);
 
+		if (!pt->codec_def || pt->codec_def->avcodec_id == -1) {
+			// not supported, or not a real audio codec
+			__make_passthrough(handler);
+			goto next;
+		}
+
 		// if the sink's codec preferences are unknown (empty), or there are
 		// no supported codecs to transcode to, then we have nothing
 		// to do. most likely this is an initial offer without a received answer.

lib/codeclib.c

@@ -22,7 +22,15 @@
 
 
 
-#define CODEC_DEF_FULL(ref, id, mult, name, clockrate, channels) { \
+#define CODEC_DEF_FULL(ref, codec_id, mult, name, clockrate, channels) { \
+	.rtpname = #ref, \
+	.avcodec_id = codec_id, \
+	.clockrate_mult = mult, \
+	.avcodec_name = #name, \
+	.default_clockrate = clockrate, \
+	.default_channels = channels, \
+}
+#define CODEC_DEF_AVC(ref, id, mult, name, clockrate, channels) { \
 	.rtpname = #ref, \
 	.avcodec_id = AV_CODEC_ID_ ## id, \
 	.clockrate_mult = mult, \
@@ -30,9 +38,9 @@
 	.default_clockrate = clockrate, \
 	.default_channels = channels, \
 }
-#define CODEC_DEF_MULT_NAME(ref, id, mult, name) CODEC_DEF_FULL(ref, id, mult, name, -1, -1)
+#define CODEC_DEF_MULT_NAME(ref, id, mult, name) CODEC_DEF_AVC(ref, id, mult, name, -1, -1)
 #define CODEC_DEF_MULT_NAME_ENC(ref, id, mult, name, clockrate, channels) \
-	CODEC_DEF_FULL(ref, id, mult, name, clockrate, channels)
+	CODEC_DEF_AVC(ref, id, mult, name, clockrate, channels)
 #define CODEC_DEF_MULT(ref, id, mult) CODEC_DEF_MULT_NAME(ref, id, mult, NULL)
 #define CODEC_DEF_MULT_ENC(ref, id, mult, clockrate, channels) \
 	CODEC_DEF_MULT_NAME_ENC(ref, id, mult, NULL, clockrate, channels)
@@ -41,6 +49,7 @@
 	CODEC_DEF_MULT_NAME_ENC(ref, id, 1, name, clockrate, channels)
 #define CODEC_DEF(ref, id) CODEC_DEF_MULT(ref, id, 1)
 #define CODEC_DEF_ENC(ref, id, clockrate, channels) CODEC_DEF_MULT_ENC(ref, id, 1, clockrate, channels)
+#define CODEC_DEF_STUB(ref) CODEC_DEF_FULL(ref, -1, 1, ref, -1, -1)
 
 static const struct codec_def_s codecs[] = {
 	CODEC_DEF(PCMA, PCM_ALAW),
@@ -65,6 +74,7 @@ static const struct codec_def_s codecs[] = {
 #endif
 	CODEC_DEF(AMR, AMR_NB),
 	CODEC_DEF(AMR-WB, AMR_WB),
+	CODEC_DEF_STUB(telephone-event),
 };
 
 
@@ -83,6 +93,9 @@ const codec_def_t *codec_find(const str *name) {
 decoder_t *decoder_new_fmt(const codec_def_t *def, int clockrate, int channels, const format_t *resample_fmt) {
 	const char *err = NULL;
 
+	if (def->avcodec_id == -1)
+		return NULL;
+
 	clockrate *= def->clockrate_mult;
 
 	decoder_t *ret = g_slice_alloc0(sizeof(*ret));

recording-daemon/decoder.c

@@ -46,6 +46,9 @@ decoder_t *decoder_new(const char *payload_str) {
 		ilog(LOG_WARN, "No decoder for payload %s", payload_str);
 		return NULL;
 	}
+	if (def->avcodec_id == -1) // not a real audio codec
+		return NULL;
+
 	clockrate *= def->clockrate_mult;
 
 	if (!resample_audio)