Asterisk media architecture conversion - no more format bitfields

This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 48 insertions, 52 deletions

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 785b4040d..f91cdb117 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -130,8 +130,8 @@ struct ast_rtp {
 	unsigned int cycles;            /*!< Shifted count of sequence number cycles */
 	double rxjitter;                /*!< Interarrival jitter at the moment */
 	double rxtransit;               /*!< Relative transit time for previous packet */
-	format_t lasttxformat;
-	format_t lastrxformat;
+	struct ast_format lasttxformat;
+	struct ast_format lastrxformat;
 
 	int rtptimeout;			/*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
 	int rtpholdtimeout;		/*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
@@ -368,9 +368,9 @@ static int rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size,
 	return __rtp_sendto(instance, buf, size, flags, sa, 0);
 }
 
-static int rtp_get_rate(format_t subclass)
+static int rtp_get_rate(struct ast_format *format)
 {
-	return (subclass == AST_FORMAT_G722) ? 8000 : ast_format_rate(subclass);
+	return (format->id == AST_FORMAT_G722) ? 8000 : ast_format_rate(format);
 }
 
 static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
@@ -560,7 +560,7 @@ static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
 	}
 
 	/* Grab the payload that they expect the RFC2833 packet to be received in */
-	payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 0, AST_RTP_DTMF);
+	payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF);
 
 	rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
 	rtp->send_duration = 160;
@@ -675,7 +675,7 @@ static int ast_rtp_dtmf_end_with_duration(struct ast_rtp_instance *instance, cha
 
 	rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
 
-	if (duration > 0 && (measured_samples = duration * rtp_get_rate(rtp->f.subclass.codec) / 1000) > rtp->send_duration) {
+	if (duration > 0 && (measured_samples = duration * rtp_get_rate(&rtp->f.subclass.format) / 1000) > rtp->send_duration) {
 		ast_debug(2, "Adjusting final end duration from %u to %u\n", rtp->send_duration, measured_samples);
 		rtp->send_duration = measured_samples;
 	}
@@ -1037,9 +1037,9 @@ static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame
 	int pred, mark = 0;
 	unsigned int ms = calc_txstamp(rtp, &frame->delivery);
 	struct ast_sockaddr remote_address = { {0,} };
-	int rate = rtp_get_rate(frame->subclass.codec) / 1000;
+	int rate = rtp_get_rate(&frame->subclass.format) / 1000;
 
-	if (frame->subclass.codec == AST_FORMAT_G722) {
+	if (frame->subclass.format.id == AST_FORMAT_G722) {
 		frame->samples /= 2;
 	}
 
@@ -1063,7 +1063,7 @@ static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame
 			}
 		}
 	} else if (frame->frametype == AST_FRAME_VIDEO) {
-		mark = frame->subclass.codec & 0x1;
+		mark = ast_format_get_video_mark(&frame->subclass.format);
 		pred = rtp->lastovidtimestamp + frame->samples;
 		/* Re-calculate last TS */
 		rtp->lastts = rtp->lastts + ms * 90;
@@ -1195,7 +1195,8 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
 {
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 	struct ast_sockaddr remote_address = { {0,} };
-	format_t codec, subclass;
+	struct ast_format subclass;
+	int codec;
 
 	ast_rtp_instance_get_remote_address(instance, &remote_address);
 
@@ -1225,19 +1226,17 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
 	}
 
 	/* Grab the subclass and look up the payload we are going to use */
-	subclass = frame->subclass.codec;
-	if (frame->frametype == AST_FRAME_VIDEO) {
-		subclass &= ~0x1LL;
-	}
-	if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, subclass)) < 0) {
-		ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(frame->subclass.codec));
+	ast_format_copy(&subclass, &frame->subclass.format);
+	if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, &subclass, 0)) < 0) {
+		ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(&frame->subclass.format));
 		return -1;
 	}
 
 	/* Oh dear, if the format changed we will have to set up a new smoother */
-	if (rtp->lasttxformat != subclass) {
-		ast_debug(1, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
+	if (ast_format_cmp(&rtp->lasttxformat, &subclass) == AST_FORMAT_CMP_NOT_EQUAL) {
+		ast_debug(1, "Ooh, format changed from %s to %s\n", ast_getformatname(&rtp->lasttxformat), ast_getformatname(&subclass));
 		rtp->lasttxformat = subclass;
+		ast_format_copy(&rtp->lasttxformat, &subclass);
 		if (rtp->smoother) {
 			ast_smoother_free(rtp->smoother);
 			rtp->smoother = NULL;
@@ -1246,9 +1245,9 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
 
 	/* 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);
+		struct ast_format_list fmt = ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance)->pref, &subclass);
 
-		switch (subclass) {
+		switch (subclass.id) {
 		case AST_FORMAT_SPEEX:
 		case AST_FORMAT_SPEEX16:
 		case AST_FORMAT_G723_1:
@@ -1261,13 +1260,13 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
 		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));
+					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));
+				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));
 			}
 		}
 	}
@@ -1315,7 +1314,7 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
 	double d;
 	double dtv;
 	double prog;
-	int rate = rtp_get_rate(rtp->f.subclass.codec);
+	int rate = rtp_get_rate(&rtp->f.subclass.format);
 
 	double normdev_rxjitter_current;
 	if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
@@ -1478,7 +1477,7 @@ static void process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned cha
 			if ((rtp->lastevent != seqno) && rtp->resp) {
 				rtp->dtmf_duration = new_duration;
 				f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
-				f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.codec)), ast_tv(0, 0));
+				f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(&f->subclass.format)), ast_tv(0, 0));
 				rtp->resp = 0;
 				rtp->dtmf_duration = rtp->dtmf_timeout = 0;
 				AST_LIST_INSERT_TAIL(frames, f, frame_list);
@@ -1489,7 +1488,7 @@ static void process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned cha
 			if (rtp->resp && rtp->resp != resp) {
 				/* Another digit already began. End it */
 				f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
-				f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.codec)), ast_tv(0, 0));
+				f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(&f->subclass.format)), ast_tv(0, 0));
 				rtp->resp = 0;
 				rtp->dtmf_duration = rtp->dtmf_timeout = 0;
 				AST_LIST_INSERT_TAIL(frames, f, frame_list);
@@ -1586,10 +1585,10 @@ static struct ast_frame *process_dtmf_cisco(struct ast_rtp_instance *instance, u
 		}
 	} else if ((rtp->resp == resp) && !power) {
 		f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE));
-		f->samples = rtp->dtmfsamples * (rtp->lastrxformat ? (rtp_get_rate(rtp->lastrxformat) / 1000) : 8);
+		f->samples = rtp->dtmfsamples * (rtp->lastrxformat.id ? (rtp_get_rate(&rtp->lastrxformat) / 1000) : 8);
 		rtp->resp = 0;
 	} else if (rtp->resp == resp)
-		rtp->dtmfsamples += 20 * (rtp->lastrxformat ? (rtp_get_rate(rtp->lastrxformat) / 1000) : 8);
+		rtp->dtmfsamples += 20 * (rtp->lastrxformat.id ? (rtp_get_rate(&rtp->lastrxformat) / 1000) : 8);
 
 	rtp->dtmf_timeout = 0;
 
@@ -1604,7 +1603,7 @@ static struct ast_frame *process_cn_rfc3389(struct ast_rtp_instance *instance, u
 	   totally help us out becuase we don't have an engine to keep it going and we are not
 	   guaranteed to have it every 20ms or anything */
 	if (rtpdebug)
-		ast_debug(0, "- RTP 3389 Comfort noise event: Level %" PRId64 " (len = %d)\n", rtp->lastrxformat, len);
+		ast_debug(0, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", (int) rtp->lastrxformat.id, len);
 
 	if (ast_test_flag(rtp, FLAG_3389_WARNING)) {
 		struct ast_sockaddr remote_address = { {0,} };
@@ -1924,10 +1923,11 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int
 	payload_type = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(instance), payload);
 
 	/* Otherwise adjust bridged payload to match */
-	bridged_payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance1), payload_type.asterisk_format, payload_type.code);
+	bridged_payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance1), payload_type.asterisk_format, &payload_type.format, payload_type.rtp_code);
 
 	/* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
-	if (!(ast_rtp_instance_get_codecs(instance1)->payloads[bridged_payload].code)) {
+	if (!(ast_rtp_instance_get_codecs(instance1)->payloads[bridged_payload].rtp_code) &&
+	!(ast_rtp_instance_get_codecs(instance1)->payloads[bridged_payload].asterisk_format)) {
 		return -1;
 	}
 
@@ -2176,15 +2176,15 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 	/* If the payload is not actually an Asterisk one but a special one pass it off to the respective handler */
 	if (!payload.asterisk_format) {
 		struct ast_frame *f = NULL;
-		if (payload.code == AST_RTP_DTMF) {
+		if (payload.rtp_code == AST_RTP_DTMF) {
 			/* process_dtmf_rfc2833 may need to return multiple frames. We do this
 			 * by passing the pointer to the frame list to it so that the method
 			 * can append frames to the list as needed.
 			 */
 			process_dtmf_rfc2833(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark, &frames);
-		} else if (payload.code == AST_RTP_CISCO_DTMF) {
+		} else if (payload.rtp_code == AST_RTP_CISCO_DTMF) {
 			f = process_dtmf_cisco(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
-		} else if (payload.code == AST_RTP_CN) {
+		} else if (payload.rtp_code == AST_RTP_CN) {
 			f = process_cn_rfc3389(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
 		} else {
 			ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n",
@@ -2204,8 +2204,9 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		return &ast_null_frame;
 	}
 
-	rtp->lastrxformat = rtp->f.subclass.codec = payload.code;
-	rtp->f.frametype = (rtp->f.subclass.codec & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass.codec & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
+	ast_format_copy(&rtp->lastrxformat, &payload.format);
+	ast_format_copy(&rtp->f.subclass.format, &payload.format);
+	rtp->f.frametype = (AST_FORMAT_GET_TYPE(rtp->f.subclass.format.id) == AST_FORMAT_TYPE_AUDIO) ? AST_FRAME_VOICE : (AST_FORMAT_GET_TYPE(rtp->f.subclass.format.id) == AST_FORMAT_TYPE_VIDEO) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
 
 	rtp->rxseqno = seqno;
 
@@ -2215,7 +2216,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		if (rtp->resp) {
 			struct ast_frame *f;
 			f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0);
-			f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.codec)), ast_tv(0, 0));
+			f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(&f->subclass.format)), ast_tv(0, 0));
 			rtp->resp = 0;
 			rtp->dtmf_timeout = rtp->dtmf_duration = 0;
 			AST_LIST_INSERT_TAIL(&frames, f, frame_list);
@@ -2232,7 +2233,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 	rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
 	rtp->f.seqno = seqno;
 
-	if (rtp->f.subclass.codec == AST_FORMAT_T140 && (int)seqno - (prev_seqno+1) > 0 && (int)seqno - (prev_seqno+1) < 10) {
+	if (rtp->f.subclass.format.id == AST_FORMAT_T140 && (int)seqno - (prev_seqno+1) > 0 && (int)seqno - (prev_seqno+1) < 10) {
 		unsigned char *data = rtp->f.data.ptr;
 
 		memmove(rtp->f.data.ptr+3, rtp->f.data.ptr, rtp->f.datalen);
@@ -2242,7 +2243,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		*data = 0xBD;
 	}
 
-	if (rtp->f.subclass.codec == AST_FORMAT_T140RED) {
+	if (rtp->f.subclass.format.id == AST_FORMAT_T140RED) {
 		unsigned char *data = rtp->f.data.ptr;
 		unsigned char *header_end;
 		int num_generations;
@@ -2251,7 +2252,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
 		int x;
 
-		rtp->f.subclass.codec = AST_FORMAT_T140;
+		ast_format_set(&rtp->f.subclass.format, AST_FORMAT_T140, 0);
 		header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
 		if (header_end == NULL) {
 			return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
@@ -2289,17 +2290,17 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		}
 	}
 
-	if (rtp->f.subclass.codec & AST_FORMAT_AUDIO_MASK) {
+	if (AST_FORMAT_GET_TYPE(rtp->f.subclass.format.id) == AST_FORMAT_TYPE_AUDIO) {
 		rtp->f.samples = ast_codec_get_samples(&rtp->f);
-		if ((rtp->f.subclass.codec == AST_FORMAT_SLINEAR) || (rtp->f.subclass.codec == AST_FORMAT_SLINEAR16)) {
+		if ((rtp->f.subclass.format.id == AST_FORMAT_SLINEAR) || (rtp->f.subclass.format.id == AST_FORMAT_SLINEAR16)) {
 			ast_frame_byteswap_be(&rtp->f);
 		}
 		calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
 		/* 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 / (rtp_get_rate(rtp->f.subclass.codec) / 1000);
-		rtp->f.len = rtp->f.samples / ((ast_format_rate(rtp->f.subclass.codec) / 1000));
-	} else if (rtp->f.subclass.codec & AST_FORMAT_VIDEO_MASK) {
+		rtp->f.ts = timestamp / (rtp_get_rate(&rtp->f.subclass.format) / 1000);
+		rtp->f.len = rtp->f.samples / ((ast_format_rate(&rtp->f.subclass.format) / 1000));
+	} else if (AST_FORMAT_GET_TYPE(rtp->f.subclass.format.id) == AST_FORMAT_TYPE_VIDEO) {
 		/* Video -- samples is # of samples vs. 90000 */
 		if (!rtp->lastividtimestamp)
 			rtp->lastividtimestamp = timestamp;
@@ -2307,13 +2308,8 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		rtp->lastividtimestamp = timestamp;
 		rtp->f.delivery.tv_sec = 0;
 		rtp->f.delivery.tv_usec = 0;
-		/* Pass the RTP marker bit as bit 0 in the subclass field.
-		 * This is ok because subclass is actually a bitmask, and
-		 * the low bits represent audio formats, that are not
-		 * involved here since we deal with video.
-		 */
-		if (mark)
-			rtp->f.subclass.codec |= 0x1;
+		/* Pass the RTP marker bit as bit */
+		ast_format_set_video_mark(&rtp->f.subclass.format);
 	} else {
 		/* TEXT -- samples is # of samples vs. 1000 */
 		if (!rtp->lastitexttimestamp)
@@ -2440,7 +2436,7 @@ static int rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int
 	}
 
 	rtp->red->t140.frametype = AST_FRAME_TEXT;
-	rtp->red->t140.subclass.codec = AST_FORMAT_T140RED;
+	ast_format_set(&rtp->red->t140.subclass.format, AST_FORMAT_T140RED, 0);
 	rtp->red->t140.data.ptr = &rtp->red->buf_data;
 
 	rtp->red->t140.ts = 0;