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authorKevin P. Fleming <kpfleming@digium.com>2006-10-25 00:32:23 +0000
committerKevin P. Fleming <kpfleming@digium.com>2006-10-25 00:32:23 +0000
commit88efcea05e9e81aadf916d549dbcceeadf0387f3 (patch)
tree94ac43ea2c20c5708d382f8ed7e32b81ce82ccfd /main
parent9178e4223eae460fc85dbc9fe515d3cf1490f9f3 (diff)
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
Diffstat (limited to 'main')
-rw-r--r--main/frame.c8
-rw-r--r--main/rtp.c2
-rw-r--r--main/translate.c2
3 files changed, 8 insertions, 4 deletions
diff --git a/main/frame.c b/main/frame.c
index f797f55fa..1b18a611d 100644
--- a/main/frame.c
+++ b/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:
diff --git a/main/rtp.c b/main/rtp.c
index 589584dee..c34f56b11 100644
--- a/main/rtp.c
+++ b/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},
diff --git a/main/translate.c b/main/translate.c
index 7845a24f1..d130a38de 100644
--- a/main/translate.c
+++ b/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;