apply extended codec support

include/asterisk/astobj2.h

@@ -17,6 +17,8 @@
 #ifndef _ASTERISK_ASTOBJ2_H
 #define _ASTERISK_ASTOBJ2_H
 
+#define REF_DEBUG 1
+
 #include "asterisk/compat.h"
 #include "asterisk/lock.h"
 

include/asterisk/format.h

@@ -219,6 +219,10 @@ struct ast_format_attr_interface {
 	 * \retval -1 failure, Value was either not found, or not allowed to be accessed.
 	 */
 	int (* const format_attr_get_val)(const struct ast_format_attr *format_attr, int key, void *val);
+
+	int (* const format_samples)(const struct ast_frame *f);
+	int (* const format_rate)(const struct ast_frame *f);
+	int (* const allowSmoother)(void);
 
 	/*
 	 * \brief Parse SDP attribute information, interpret it, and store it in ast_format_attr structure.
@@ -471,4 +475,11 @@ int ast_format_is_slinear(const struct ast_format *format);
  * \brief Get the best slinear format id for a given sample rate
  */
 enum ast_format_id ast_format_slin_by_rate(unsigned int rate);
+
+int ast_format_custom_add(struct ast_format_list *newFormat);
+int ast_format_custom_get_samples(const struct ast_format *format, const struct ast_frame *f);
+int ast_format_custom_get_rate(const struct ast_format *format);
+int ast_format_custom_requestFormatId(enum ast_format_type);
+int ast_format_custom_register(struct ast_format_attr_interface *newFormat, enum ast_format_type formatType);
+int ast_format_allowSmoother(const struct ast_format *format);
 #endif /* _AST_FORMAT_H */

include/asterisk/rtp_engine.h

@@ -2108,6 +2108,9 @@ void ast_rtp_dtls_cfg_copy(const struct ast_rtp_dtls_cfg *src_cfg, struct ast_rt
  */
 void ast_rtp_dtls_cfg_free(struct ast_rtp_dtls_cfg *dtls_cfg);
 
+void ast_add_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate);
+void ast_add_static_payload(int map, const struct ast_format *format, int rtp_code);
+
 #if defined(__cplusplus) || defined(c_plusplus)
 }
 #endif

main/channel.c

@@ -925,6 +925,8 @@ struct ast_format *ast_best_codec(struct ast_format_cap *cap, struct ast_format
 		AST_FORMAT_G729A,
 		/*! Down to G.723.1 which is proprietary but at least designed for voice */
 		AST_FORMAT_G723_1,
+
+		//TODO how to handle custom formats
 	};
 	char buf[512];
 	int x;

main/format.c

@@ -172,7 +172,7 @@ static int format_set_helper(struct ast_format *format, va_list ap)
 	struct interface_ao2_wrapper *wrapper;
 
 	if (!(wrapper = find_interface(format))) {
-		ast_log(LOG_WARNING, "Could not find format interface to set.\n");
+		ast_log(LOG_WARNING, "Could not find format interface for %d to set.\n", format->id);
 		return -1;
 	}
 
@@ -729,6 +729,8 @@ const char *ast_codec2str(struct ast_format *format)
 
 int ast_format_rate(const struct ast_format *format)
 {
+	int rate;
+
 	switch (format->id) {
 	case AST_FORMAT_SLINEAR12:
 		return 12000;
@@ -783,7 +785,8 @@ int ast_format_rate(const struct ast_format *format)
 		}
 	}
 	default:
-		return 8000;
+		rate = 16000;//ast_format_custom_get_rate(format);
+		return (rate > 0) ? rate : 8000;
 	}
 }
 
@@ -1431,3 +1434,90 @@ int ast_format_attr_unreg_interface(const struct ast_format_attr_interface *inte
 	f_list = ast_format_list_destroy(f_list);
 	return 0;
 }
+
+
+int ast_format_custom_add(struct ast_format_list *newFormat){
+	int res = 0;
+
+	ast_log(LOG_NOTICE, "Add new Format %s\n", newFormat->name);
+	res = format_list_add_custom(newFormat);
+	build_format_list_array();
+	return res;
+}
+
+int ast_format_custom_get_samples(const struct ast_format *format, const struct ast_frame *f){
+	struct interface_ao2_wrapper *wrapper;
+	int samples = 0;
+
+	if ((wrapper = find_interface(format))) {
+		if (wrapper->interface && wrapper->interface->format_samples) {
+			samples = wrapper->interface->format_samples(f);
+		}
+		ao2_ref(wrapper, -1);
+	}
+
+	return samples;
+}
+
+int ast_format_custom_get_rate(const struct ast_format *format){
+	struct interface_ao2_wrapper *wrapper;
+	int rate = -1;
+
+	if ((wrapper = find_interface(format))) {
+		if (wrapper->interface && wrapper->interface->format_rate) {
+			rate = wrapper->interface->format_rate(&format->fattr);
+		}
+		ao2_ref(wrapper, -1);
+	}
+
+	return rate;
+}
+int ast_format_allowSmoother(const struct ast_format *format){
+	struct interface_ao2_wrapper *wrapper;
+	int result = 1;
+
+	if ((wrapper = find_interface(format))) {
+		if (wrapper->interface && wrapper->interface->allowSmoother) {
+			result = wrapper->interface->allowSmoother();
+		}
+		ao2_ref(wrapper, -1);
+	}
+
+	return result;
+}
+
+
+int ast_format_custom_register(struct ast_format_attr_interface *newFormatInterface, enum ast_format_type formatType){
+
+	int x = 1;
+	int i;
+	size_t f_len;
+	const struct ast_format_list *f_list;
+
+	ao2_wrlock(interfaces);
+
+	/* if this format does not have a format id, get some new */
+	ast_log(LOG_NOTICE, "register new codec with id: %d\n", newFormatInterface->id);
+// 	if(newFormatInterface->id < 1){
+
+		ast_log(LOG_NOTICE, "search for new id\n");
+
+		f_list = ast_format_list_get(&f_len);
+		for (x = 0; x < f_len; x++) {
+			if (AST_FORMAT_GET_TYPE(f_list[x].format.id)  == formatType) {
+				i = f_list[x].format.id;
+			}
+		}
+		f_list = ast_format_list_destroy(f_list);
+
+
+		newFormatInterface->id = ++i;
+		ast_log(LOG_NOTICE, "found new id %d\n", newFormatInterface->id);
+
+// 	}
+	ao2_unlock(interfaces);
+
+	ast_format_attr_reg_interface(newFormatInterface);
+
+	return 0;
+}

main/frame.c

@@ -1084,7 +1084,10 @@ int ast_codec_get_samples(struct ast_frame *f)
 		samples = ast_format_rate(&f->subclass.format) / 50;
 		break;
 	default:
-		ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
+		samples = ast_format_custom_get_samples(&f->subclass.format, f);
+		if(!samples){
+			ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
+		}
 	}
 	return samples;
 }

main/rtp_engine.c

@@ -594,6 +594,7 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
 		 * rates are not compared */
 		if (sample_rate && t->sample_rate &&
 		    (sample_rate != t->sample_rate)) {
+		  ast_log(LOG_WARNING, "sample rate does not match for %s\n", mimesubtype);
 			continue;
 		}
 
@@ -623,7 +624,7 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs,
 		break;
 	}
 	ast_rwlock_unlock(&mime_types_lock);
-
+ast_log(LOG_WARNING, "found: %d\n", found);
 	return (found ? 0 : -2);
 }
 
@@ -2333,3 +2334,10 @@ int ast_rtp_engine_init()
 
 	return 0;
 }
+
+void ast_add_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate){
+	return set_next_mime_type(format, rtp_code, type, subtype, sample_rate);
+}
+void ast_add_static_payload(int map, const struct ast_format *format, int rtp_code){
+	return add_static_payload(map, format, rtp_code);
+}

res/res_format_attr_h264.c

@@ -270,6 +270,10 @@ static void h264_format_attr_sdp_generate(const struct ast_format_attr *format_a
 			if (!added) {
 				ast_str_append(str, 0, "a=fmtp:%d profile-level-id=%X%X%X", payload, format_attr->format_attr[H264_ATTR_KEY_PROFILE_IDC],
 					       format_attr->format_attr[H264_ATTR_KEY_PROFILE_IOP], format_attr->format_attr[H264_ATTR_KEY_LEVEL]);
+
+
+
+				ast_str_append(str, 0, "\r\na=orient:portrait\r\na=rtcp-fb:* nack pli");
 				added = 1;
 			} else {
 				ast_str_append(str, 0, ";profile-level-id=%X%X%X", format_attr->format_attr[H264_ATTR_KEY_PROFILE_IDC],

res/res_rtp_asterisk.c

@@ -2607,6 +2607,7 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
 	struct ast_sockaddr remote_address = { {0,} };
 	struct ast_format subclass;
 	int codec;
+	int allowSmoother = 1;
 
 	ast_rtp_instance_get_remote_address(instance, &remote_address);
 
@@ -2652,12 +2653,11 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
 			rtp->smoother = NULL;
 		}
 	}
-
-	/* If no smoother is present see if we have to set one up */
-	if (!rtp->smoother) {
-		struct ast_format_list fmt = ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance)->pref, &subclass);
-
-		switch (subclass.id) {
+
+	allowSmoother = ast_format_allowSmoother(&subclass);
+
+	/* disable smoother for frame based formats */
+	switch (subclass.id) {
 		case AST_FORMAT_SPEEX:
 		case AST_FORMAT_SPEEX16:
 		case AST_FORMAT_SPEEX32:
@@ -2667,21 +2667,29 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
 		case AST_FORMAT_SIREN7:
 		case AST_FORMAT_SIREN14:
 		case AST_FORMAT_G719:
-			/* these are all frame-based codecs and cannot be safely run through
-			   a smoother */
-			break;
+
+			allowSmoother = 0;
+		break;
 		default:
-			if (fmt.inc_ms) {
-				if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
-					ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %d len: %d\n", ast_getformatname(&subclass), fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
-					return -1;
-				}
-				if (fmt.flags) {
-					ast_smoother_set_flags(rtp->smoother, fmt.flags);
-				}
-				ast_debug(1, "Created smoother: format: %s ms: %d len: %d\n", ast_getformatname(&subclass), fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
+		  break;
+	}
+
+
+	/* If no smoother is present see if we have to set one up */
+	if (!rtp->smoother && allowSmoother) {
+		struct ast_format_list fmt = ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance)->pref, &subclass);
+
+		if (fmt.inc_ms) {
+			if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
+				ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %d len: %d\n", ast_getformatname(&subclass), fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
+				return -1;
+			}
+			if (fmt.flags) {
+				ast_smoother_set_flags(rtp->smoother, fmt.flags);
 			}
+			ast_debug(1, "Created smoother: format: %s ms: %d len: %d\n", ast_getformatname(&subclass), fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
 		}
+
 	}
 
 	/* Feed audio frames into the actual function that will create a frame and send it */