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Merged revisions 46154 via svnmerge from

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

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2006-10-25 00:32:23 +00:00
parent 9178e4223e
commit 88efcea05e
7 changed files with 35 additions and 10 deletions
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2
main/rtp.c
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@@ -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},