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Remove really broke MP3 stuff in favor of G.726 in the near future

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2003-11-04 02:40:09 +00:00
parent 7ec16a6af7
commit 1e95c3a4ac
58 changed files with 27 additions and 15549 deletions
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4
rtp.c
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@@ -508,7 +508,7 @@ static struct {
{{1, AST_FORMAT_GSM}, "audio", "GSM"},
{{1, AST_FORMAT_ULAW}, "audio", "PCMU"},
{{1, AST_FORMAT_ALAW}, "audio", "PCMA"},
{{1, AST_FORMAT_MP3}, "audio", "MPA"},
{{1, AST_FORMAT_G726}, "audio", "G726-32"},
{{1, AST_FORMAT_ADPCM}, "audio", "DVI4"},
{{1, AST_FORMAT_SLINEAR}, "audio", "L16"},
{{1, AST_FORMAT_LPC10}, "audio", "LPC"},
@@ -529,6 +529,7 @@ static struct {
table for transmission */
static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
[0] = {1, AST_FORMAT_ULAW},
[2] = {1, AST_FORMAT_G726}, // Technically this is G.721, but if Cisco can do it, so can we...
[3] = {1, AST_FORMAT_GSM},
[4] = {1, AST_FORMAT_G723_1},
[5] = {1, AST_FORMAT_ADPCM}, // 8 kHz
@@ -538,7 +539,6 @@ static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
[10] = {1, AST_FORMAT_SLINEAR}, // 2 channels
[11] = {1, AST_FORMAT_SLINEAR}, // 1 channel
[13] = {0, AST_RTP_CN},
[14] = {1, AST_FORMAT_MP3},
[16] = {1, AST_FORMAT_ADPCM}, // 11.025 kHz
[17] = {1, AST_FORMAT_ADPCM}, // 22.050 kHz
[18] = {1, AST_FORMAT_G729A},