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kenner at gnat.com
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 Posted: Tue Jul 07, 2015 1:02 pm Post subject: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0? |
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I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {
id = AST_FORMAT_SLINEAR16, fattr = {format_attr = {
0 <repeats 64 times>}, rtp_marker_bit = 0 '\000'}}}, datalen = 0,
samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64,
src = 0x51623b0 "func_jitterbuffer interpolation", data = {ptr = 0x0,
uint32 = 0, pad = "\000\000\000\000\000\000\000"}, delivery = {
tv_sec = 1436290187, tv_usec = 304285}, frame_list = {next = 0x0},
flags = 0, ts = 0, len = 0, seqno = 0}
so datalen is 0 and samples nonzero. ast_frame_adjust_volume, however,
iterates over samples, not datalen. Is that correct?
What does it mean to have a packet with a zero datalen anyway?
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jcolp at digium.com
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| Posted: Wed Jul 08, 2015 9:52 am Post subject: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0? |
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Richard Kenner wrote:
| Quote: | I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f $6 = {frametype = AST_FRAME_VOICE, subclass = {integer
= 100021, format = { id = AST_FORMAT_SLINEAR16, fattr = {format_attr
= { 0<repeats 64 times>}, rtp_marker_bit = 0 '\000'}}}, datalen = 0,
samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64, src
= 0x51623b0 "func_jitterbuffer interpolation", data = {ptr = 0x0,
uint32 = 0, pad = "\000\000\000\000\000\000\000"}, delivery = {
tv_sec = 1436290187, tv_usec = 304285}, frame_list = {next = 0x0},
flags = 0, ts = 0, len = 0, seqno = 0}
so datalen is 0 and samples nonzero. ast_frame_adjust_volume,
however, iterates over samples, not datalen. Is that correct?
What does it mean to have a packet with a zero datalen anyway?
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This is an interpolated frame from func_jitterbuffer. It's part of
packet loss concealment. What scenario exposed this?
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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kenner at gnat.com
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| Posted: Wed Jul 08, 2015 10:57 am Post subject: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0? |
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| Quote: | This is an interpolated frame from func_jitterbuffer. It's part of
packet loss concealment. What scenario exposed this?
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We were testing for clipping by doing Set(VOLUME(RX)=100) but we were
connecting to a ConfBridge that had a jitterbuffer. This occurred when
the phone (SIP) hung up.
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_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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