From 88efcea05e9e81aadf916d549dbcceeadf0387f3 Mon Sep 17 00:00:00 2001 From: "Kevin P. Fleming" Date: Wed, 25 Oct 2006 00:32:23 +0000 Subject: 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 --- main/frame.c | 8 +++++--- main/rtp.c | 2 ++ main/translate.c | 2 +- 3 files changed, 8 insertions(+), 4 deletions(-) (limited to 'main') 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; -- cgit v1.2.3