Merged revisions 46154 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) | 2 lines

add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46155 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_h323.c

@@ -3122,6 +3122,8 @@ static char *convertcap(int cap)
 		return "ULAW";
 	case AST_FORMAT_ALAW:
 		return "ALAW";
+	case AST_FORMAT_G722:
+		return "G.722";
 	case AST_FORMAT_ADPCM:
 		return "G.728";
 	case AST_FORMAT_G729A:

channels/chan_iax2.c

@@ -183,7 +183,8 @@ int (*iax2_regfunk)(const char *username, int onoff) = NULL;
 #define IAX_CAPABILITY_MEDBANDWIDTH 	(IAX_CAPABILITY_FULLBANDWIDTH & 	\
 					 ~AST_FORMAT_SLINEAR &			\
 					 ~AST_FORMAT_ULAW &			\
-					 ~AST_FORMAT_ALAW) 
+					 ~AST_FORMAT_ALAW &			\
+					 ~AST_FORMAT_G722) 
 /* A modem */
 #define IAX_CAPABILITY_LOWBANDWIDTH	(IAX_CAPABILITY_MEDBANDWIDTH & 		\
 					 ~AST_FORMAT_G726 &			\

formats/format_pcm.c

@@ -443,6 +443,18 @@ static const struct ast_format pcm_f = {
 	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
 };
 
+static const struct ast_format g722_f = {
+	.name = "g722",
+	.exts = "g722",
+	.format = AST_FORMAT_G722,
+	.write = pcm_write,
+	.seek = pcm_seek,
+	.trunc = pcm_trunc,
+	.tell = pcm_tell,
+	.read = pcm_read,
+	.buf_size = (BUF_SIZE * 2) + AST_FRIENDLY_OFFSET,
+};
+
 static const struct ast_format au_f = {
 	.name = "au",
 	.exts = "au",
@@ -467,14 +479,18 @@ static int load_module(void)
 	for (index = 0; index < (sizeof(alaw_silence) / sizeof(alaw_silence[0])); index++)
 		alaw_silence[index] = AST_LIN2A(0);
 
-	return ast_format_register(&pcm_f) || ast_format_register(&alaw_f)
-		|| ast_format_register(&au_f);
+	return ast_format_register(&pcm_f)
+		|| ast_format_register(&alaw_f)
+		|| ast_format_register(&au_f)
+		|| ast_format_register(&g722_f);
 }
 
 static int unload_module(void)
 {
-	return ast_format_unregister(pcm_f.name) || ast_format_unregister(alaw_f.name)
-		|| ast_format_unregister(au_f.name);
+	return ast_format_unregister(pcm_f.name)
+		|| ast_format_unregister(alaw_f.name)
+		|| ast_format_unregister(au_f.name)
+		|| ast_format_unregister(g722_f.name);
 }	
 
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Raw/Sun uLaw/ALaw 8khz Audio support (PCM,PCMA,AU)");
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Raw/Sun uLaw/ALaw 8KHz Audio support (PCM,PCMA,AU) and G.722 16Khz Audio Support");

include/asterisk/frame.h

@@ -240,6 +240,8 @@ extern struct ast_frame ast_null_frame;
 #define AST_FORMAT_ILBC		(1 << 10)
 /*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
 #define AST_FORMAT_G726		(1 << 11)
+/*! G.722 */
+#define AST_FORMAT_G722		(1 << 12)
 /*! Maximum audio format */
 #define AST_FORMAT_MAX_AUDIO	(1 << 15)
 /*! Maximum audio mask */

main/frame.c

@@ -106,14 +106,15 @@ static struct ast_format_list AST_FORMAT_LIST[] = {					/*!< Bit number: comment
 	{ 1, AST_FORMAT_GSM, "gsm" , "GSM", 33, 20, 300, 20, 20 },		/*!<  2: codec_gsm.c */
 	{ 1, AST_FORMAT_ULAW, "ulaw", "G.711 u-law", 80, 10, 150, 10, 20 },	/*!<  3: codec_ulaw.c */
 	{ 1, AST_FORMAT_ALAW, "alaw", "G.711 A-law", 80, 10, 150, 10, 20 },	/*!<  4: codec_alaw.c */
-	{ 1, AST_FORMAT_G726, "g726", "G.726 RFC3551", 40, 10, 300, 10, 20 },/*!<  5: codec_g726.c */
+	{ 1, AST_FORMAT_G726, "g726", "G.726 RFC3551", 40, 10, 300, 10, 20 },	/*!<  5: codec_g726.c */
 	{ 1, AST_FORMAT_ADPCM, "adpcm" , "ADPCM", 40, 10, 300, 10, 20 },	/*!<  6: codec_adpcm.c */
 	{ 1, AST_FORMAT_SLINEAR, "slin", "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE },	/*!< 7 */
-	{ 1, AST_FORMAT_LPC10, "lpc10", "LPC10", 7, 20, 20, 20, 20 },	/*!<  8: codec_lpc10.c */ 
+	{ 1, AST_FORMAT_LPC10, "lpc10", "LPC10", 7, 20, 20, 20, 20 },		/*!<  8: codec_lpc10.c */ 
 	{ 1, AST_FORMAT_G729A, "g729", "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 },	/*!<  9: Binary commercial distribution */
-	{ 1, AST_FORMAT_SPEEX, "speex", "SpeeX", 10, 10, 60, 10, 20 },	/*!< 10: codec_speex.c */
+	{ 1, AST_FORMAT_SPEEX, "speex", "SpeeX", 10, 10, 60, 10, 20 },		/*!< 10: codec_speex.c */
 	{ 1, AST_FORMAT_ILBC, "ilbc", "iLBC", 50, 30, 30, 30, 30 },		/*!< 11: codec_ilbc.c */ /* inc=30ms - workaround */
 	{ 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2", 40, 10, 300, 10, 20 },	/*!<  12: codec_g726.c */
+	{ 1, AST_FORMAT_G722, "g722", "G722"},					/*!< 13 */
 	{ 0, 0, "nothing", "undefined" },
 	{ 0, 0, "nothing", "undefined" },
 	{ 0, 0, "nothing", "undefined" },
@@ -1356,6 +1357,7 @@ int ast_codec_get_samples(struct ast_frame *f)
 		break;
 	case AST_FORMAT_ULAW:
 	case AST_FORMAT_ALAW:
+	case AST_FORMAT_G722:
 		samples = f->datalen;
 		break;
 	case AST_FORMAT_ADPCM:

main/rtp.c

@@ -1330,6 +1330,7 @@ static struct {
 	{{1, AST_FORMAT_G729A}, "audio", "G729"},
 	{{1, AST_FORMAT_SPEEX}, "audio", "speex"},
 	{{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
+	{{1, AST_FORMAT_G722}, "audio", "G722"},
 	{{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32"},
 	{{0, AST_RTP_DTMF}, "audio", "telephone-event"},
 	{{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
@@ -1356,6 +1357,7 @@ static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
 	[6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
 	[7] = {1, AST_FORMAT_LPC10},
 	[8] = {1, AST_FORMAT_ALAW},
+	[9] = {1, AST_FORMAT_G722},
 	[10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
 	[11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
 	[13] = {0, AST_RTP_CN},

main/translate.c

@@ -476,7 +476,7 @@ static void rebuild_matrix(int samples)
 
 static int show_translation(int fd, int argc, char *argv[])
 {
-#define SHOW_TRANS 12
+#define SHOW_TRANS 13
 	int x, y, z;
 	int curlen = 0, longest = 0;