addition of G.719 pass-through support

(closes issue #16293)
Reported by: malcolmd
Patches:
      g719.passthrough.patch.7 uploaded by malcolmd (license 924)
      format_g719.c uploaded by malcolmd (license 924)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -67,6 +67,7 @@ SIP Changes
    to each other
  * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
    Charge messages to snom phones.
+ * Added support for G.719 media streams.
 
 IAX2 Changes
 -----------
@@ -498,6 +499,8 @@ Miscellaneous
    significant increase in performance (about 3X) for installations using this switchtype.
  * Distributed devicestate now supports the use of the XMPP protocol, in addition to
    AIS.  For more information, please see doc/distributed_devstate-XMPP.txt
+ * The addition of G.719 pass-through support.
+
 
 CLI Changes
 -----------

channels/chan_iax2.c

@@ -316,6 +316,7 @@ static int (*iax2_regfunk)(const char *username, int onoff) = NULL;
                      ~AST_FORMAT_SLINEAR16 &    \
                      ~AST_FORMAT_SIREN7 &       \
                      ~AST_FORMAT_SIREN14 &      \
+                     ~AST_FORMAT_G719 &         \
                      ~AST_FORMAT_ULAW &         \
                      ~AST_FORMAT_ALAW &         \
                      ~AST_FORMAT_G722)

channels/chan_sip.c

@@ -8520,6 +8520,15 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
 					}
 				}
 				break;
+			case AST_FORMAT_G719:
+				if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
+					if (bit_rate != 64000) {
+						ast_log(LOG_WARNING, "Got G.719 offer at %d bps, but only 64000 bps supported; ignoring.\n", bit_rate);
+						ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
+					} else {
+						found = TRUE;
+					}
+				}
 			}
 		}
 	}
@@ -9771,6 +9780,10 @@ static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec,
 		/* Indicate that we only expect 48Kbps */
 		ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
 		break;
+	case AST_FORMAT_G719:
+		/* Indicate that we only expect 64Kbps */
+		ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
+		break;
 	}
 
 	if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))

formats/format_g719.c

@@ -0,0 +1,143 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2010, Anthony Minessale and Digium, Inc.
+ * Anthony Minessale (anthmct@yahoo.com)
+ * Kevin P. Fleming <kpfleming@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief ITU G.719 , 64kbps bitrate only
+ * \arg File name extensions: g719
+ * \ingroup formats
+ */
+ 
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/mod_format.h"
+#include "asterisk/module.h"
+#include "asterisk/endian.h"
+
+#define BUF_SIZE	160		/* 20 milliseconds == 160 bytes, 960 samples */
+#define SAMPLES_TO_BYTES(x)	((typeof(x)) x / ((float) 960 / 160))
+#define BYTES_TO_SAMPLES(x)	((typeof(x)) x * ((float) 960 / 160))
+
+static struct ast_frame *g719read(struct ast_filestream *s, int *whennext)
+{
+	int res;
+	/* Send a frame from the file to the appropriate channel */
+
+	s->fr.frametype = AST_FRAME_VOICE;
+	s->fr.subclass.codec = AST_FORMAT_G719;
+	s->fr.mallocd = 0;
+	AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
+	if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) != s->fr.datalen) {
+		if (res)
+			ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
+		return NULL;
+	}
+	*whennext = s->fr.samples = BYTES_TO_SAMPLES(res);
+	return &s->fr;
+}
+
+static int g719write(struct ast_filestream *fs, struct ast_frame *f)
+{
+	int res;
+
+	if (f->frametype != AST_FRAME_VOICE) {
+		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
+		return -1;
+	}
+	if (f->subclass.codec != AST_FORMAT_G719) {
+		ast_log(LOG_WARNING, "Asked to write non-G.719 frame (%s)!\n", ast_getformatname(f->subclass.codec));
+		return -1;
+	}
+	if ((res = fwrite(f->data.ptr, 1, f->datalen, fs->f)) != f->datalen) {
+		ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
+		return -1;
+	}
+	return 0;
+}
+
+static int g719seek(struct ast_filestream *fs, off_t sample_offset, int whence)
+{
+	off_t offset = 0, min = 0, cur, max;
+
+	sample_offset = SAMPLES_TO_BYTES(sample_offset);
+
+	cur = ftello(fs->f);
+
+	fseeko(fs->f, 0, SEEK_END);
+
+	max = ftello(fs->f);
+
+	if (whence == SEEK_SET)
+		offset = sample_offset;
+	else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
+		offset = sample_offset + cur;
+	else if (whence == SEEK_END)
+		offset = max - sample_offset;
+
+	if (whence != SEEK_FORCECUR)
+		offset = (offset > max) ? max : offset;
+
+	/* always protect against seeking past begining. */
+	offset = (offset < min) ? min : offset;
+
+	return fseeko(fs->f, offset, SEEK_SET);
+}
+
+static int g719trunc(struct ast_filestream *fs)
+{
+	return ftruncate(fileno(fs->f), ftello(fs->f));
+}
+
+static off_t g719tell(struct ast_filestream *fs)
+{
+	return BYTES_TO_SAMPLES(ftello(fs->f));
+}
+
+static const struct ast_format g719_f = {
+	.name = "g719",
+	.exts = "g719",
+	.format = AST_FORMAT_G719,
+	.write = g719write,
+	.seek = g719seek,
+	.trunc = g719trunc,
+	.tell = g719tell,
+	.read = g719read,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+};
+
+static int load_module(void)
+{
+	if (ast_format_register(&g719_f))
+		return AST_MODULE_LOAD_DECLINE;
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	return ast_format_unregister(g719_f.name);
+}	
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER,"ITU G.719",
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = 10,
+);
+

include/asterisk/frame.h

@@ -294,6 +294,8 @@ extern struct ast_frame ast_null_frame;
 /*! Maximum text mask */
 #define AST_FORMAT_MAX_TEXT   (1ULL << 28)
 #define AST_FORMAT_TEXT_MASK  (((1ULL << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))
+/*! G.719 (64 kbps assumed) */
+#define AST_FORMAT_G719	      (1ULL << 32)
 /*! Raw mu-law data (G.711) */
 #define AST_FORMAT_TESTLAW       (1ULL << 47)
 /*! Reserved bit - do not use */
@@ -746,6 +748,8 @@ static force_inline int ast_format_rate(format_t format)
 		return 16000;
 	case AST_FORMAT_SIREN14:
 		return 32000;
+	case AST_FORMAT_G719:
+		return 48000;
 	default:
 		return 8000;
 	}

main/channel.c

@@ -792,6 +792,7 @@ format_t ast_best_codec(format_t fmts)
 		AST_FORMAT_ULAW,
 		/*! Unless of course, you're a silly European, so then prefer ALAW */
 		AST_FORMAT_ALAW,
+		AST_FORMAT_G719,
 		AST_FORMAT_SIREN14,
 		AST_FORMAT_SIREN7,
 		AST_FORMAT_TESTLAW,

main/frame.c

@@ -120,6 +120,7 @@ static const struct ast_format_list AST_FORMAT_LIST[] = {
 	{ AST_FORMAT_SIREN7, "siren7", 16000, "ITU G.722.1 (Siren7, licensed from Polycom)", 80, 20, 80, 20, 20 },			/*!< Binary commercial distribution */
 	{ AST_FORMAT_SIREN14, "siren14", 32000, "ITU G.722.1 Annex C, (Siren14, licensed from Polycom)", 120, 20, 80, 20, 20 },	/*!< Binary commercial distribution */
 	{ AST_FORMAT_TESTLAW, "testlaw", 8000, "G.711 test-law", 80, 10, 150, 10, 20 },                                 /*!< codec_ulaw.c */
+	{ AST_FORMAT_G719, "g719", 48000, "ITU G.719", 160, 20, 80, 20, 20 },
 };
 
 struct ast_frame ast_null_frame = { AST_FRAME_NULL, };
@@ -1491,6 +1492,10 @@ int ast_codec_get_samples(struct ast_frame *f)
 		/* 32,000 samples per second at 48kbps is 6,000 bytes per second */
 		samples = (int) f->datalen * ((float) 32000 / 6000);
 		break;
+	case AST_FORMAT_G719:
+		/* 48,000 samples per second at 64kbps is 8,000 bytes per second */
+		samples = (int) f->datalen * ((float) 48000 / 8000);
+		break;
 	default:
 		ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), f->subclass.codec));
 	}
@@ -1538,6 +1543,10 @@ int ast_codec_get_len(format_t format, int samples)
 		/* 32,000 samples per second at 48kbps is 6,000 bytes per second */
 		len = (int) samples / ((float) 32000 / 6000);
 		break;
+	case AST_FORMAT_G719:
+		/* 48,000 samples per second at 64kbps is 8,000 bytes per second */
+		len = (int) samples / ((float) 48000 / 8000);
+		break;
 	default:
 		ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
 	}

main/rtp_engine.c

@@ -122,6 +122,7 @@ static const struct ast_rtp_mime_type {
 	{{1, AST_FORMAT_T140}, "text", "T140", 1000},
 	{{1, AST_FORMAT_SIREN7}, "audio", "G7221", 16000},
 	{{1, AST_FORMAT_SIREN14}, "audio", "G7221", 32000},
+	{{1, AST_FORMAT_G719}, "audio", "G719", 48000},
 };
 
 /*!
@@ -169,6 +170,7 @@ static const struct ast_rtp_payload_type static_RTP_PT[AST_RTP_MAX_PT] = {
 	[111] = {1, AST_FORMAT_G726},
 	[112] = {1, AST_FORMAT_G726_AAL2},
 	[115] = {1, AST_FORMAT_SIREN14},
+	[116] = {1, AST_FORMAT_G719},
 	[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
 };
 

res/res_rtp_asterisk.c

@@ -1231,6 +1231,7 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
 		case AST_FORMAT_G723_1:
 		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;