Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 34 insertions, 7 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 06e23ad07..4d9408f38 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -8885,11 +8885,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 	}
 
 	ast_debug(4, "We have an owner, now see if we need to change this call\n");
-
-	if (!(ast_format_cap_has_joint(p->owner->nativeformats, p->jointcaps)) && ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO)) {
+	if (ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO)) {
 		if (debug) {
 			char s1[SIPBUFSIZE], s2[SIPBUFSIZE];
-			ast_debug(1, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n",
+			ast_debug(1, "Setting native formats after processing SDP. peer joint formats %s, old nativeformats %s\n",
 				ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcaps),
 				ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats));
 		}
@@ -9109,13 +9108,12 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
 				ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
 		}
 	} else if (sscanf(a, "fmtp: %30u %63s", &codec, fmtp_string) == 2) {
-		struct ast_rtp_payload_type payload;
+		struct ast_format *format;
 
-		payload = ast_rtp_codecs_payload_lookup(newaudiortp, codec);
-		if (payload.format.id && payload.asterisk_format) {
+		if ((format = ast_rtp_codecs_get_payload_format(newaudiortp, codec))) {
 			unsigned int bit_rate;
 
-			switch ((int) payload.format.id) {
+			switch ((int) format->id) {
 			case AST_FORMAT_SIREN7:
 				if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
 					if (bit_rate != 32000) {
@@ -9145,6 +9143,21 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
 						found = TRUE;
 					}
 				}
+				break;
+			case AST_FORMAT_SILK:
+				{
+					int val = 0;
+					if (sscanf(fmtp_string, "maxaveragebitrate=%30u", &val) == 1) {
+						ast_format_append(format, SILK_ATTR_KEY_MAX_BITRATE, val, AST_FORMAT_ATTR_END);
+					}
+					if (sscanf(fmtp_string, "usedtx=%30u", &val) == 1) {
+						ast_format_append(format, SILK_ATTR_KEY_DTX, val ? 1 : 0, AST_FORMAT_ATTR_END);
+					}
+					if (sscanf(fmtp_string, "useinbandfec=%30u", &val) == 1) {
+						ast_format_append(format, SILK_ATTR_KEY_FEC, val ? 1 : 0, AST_FORMAT_ATTR_END);
+					}
+					break;
+				}
 			}
 		}
 	}
@@ -10505,6 +10518,20 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
 		/* Indicate that we only expect 64Kbps */
 		ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
 		break;
+	case AST_FORMAT_SILK:
+		{
+			int val = 0;
+			if (!ast_format_get_value(format, SILK_ATTR_KEY_MAX_BITRATE, &val) && val > 5000 && val < 40000) {
+				ast_str_append(a_buf, 0, "a=fmtp:%d maxaveragebitrate=%u\r\n", rtp_code, val);
+			}
+			if (!ast_format_get_value(format, SILK_ATTR_KEY_DTX, &val)) {
+				ast_str_append(a_buf, 0, "a=fmtp:%d usedtx=%u\r\n", rtp_code, val ? 1 : 0);
+			}
+			if (!ast_format_get_value(format, SILK_ATTR_KEY_FEC, &val)) {
+				ast_str_append(a_buf, 0, "a=fmtp:%d useinbandfec=%u\r\n", rtp_code, val ? 1 : 0);
+			}
+			break;
+		}
 	}
 
 	if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))