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gitosis at osmocom.org gitosis at osmocom.orgThis is an automated email from the git hooks/post-receive script. It was generated because a ref change was pushed to the repository containing the project "Osmocom BTS-side code (Abis, scheduling, ...)". The branch, master has been updated via 1eaa3d72eaa10bb854b2adff48d3f7f3e06dcb8c (commit) via 668f8df3be7f644056d16ea4a7fd7929429da0b4 (commit) via cc4a08bdc753492c4367e4786a573a6cdf84b56e (commit) from 862807504b5ecf4fe79c435286487a1a576d09eb (commit) Those revisions listed above that are new to this repository have not appeared on any other notification email; so we list those revisions in full, below. - Log ----------------------------------------------------------------- http://cgit.osmocom.org/osmo-bts/commit/?id=1eaa3d72eaa10bb854b2adff48d3f7f3e06dcb8c commit 1eaa3d72eaa10bb854b2adff48d3f7f3e06dcb8c Author: Holger Hans Peter Freyther <holger at moiji-mobile.com> Date: Mon Sep 21 11:28:03 2015 +0200 audio/rsl: Honor the speech mode and don't send anything Spotted by Ciaby while debugging an audio issue. Do not send anything to port==0 to the BSC/NITB. Look at the upper bits of the speech_mode to determine if sending is allowed. 0x1 means recv_only and all other modes allow us to send. Manually verified with a single phone call with LCR bridge mode to send a CRCX early but a MDCX sendrecv later. The audio starts to flow after the MDCX message. Virtual Addr space didn't increase over 10 calls. The l1p_msg is freed by the caller. The code might not re-set speech_mode from one call to another but if it is ever != 0 it can be expected that the BSC will always set it. This is because we do not (and don't want to) allocate the lchan dynamically on every usage. Fixes: SYS#2111 http://cgit.osmocom.org/osmo-bts/commit/?id=668f8df3be7f644056d16ea4a7fd7929429da0b4 commit 668f8df3be7f644056d16ea4a7fd7929429da0b4 Author: Holger Hans Peter Freyther <holger at moiji-mobile.com> Date: Mon Sep 21 11:18:06 2015 +0200 audio/rsl: Include statistics for one call Use the new libosmo-abis API to query the session for the statistics and then send it as a TLV element to the BSC. This can be used to do post processing about the call. E.g to figure out if no audio arrived at all. http://cgit.osmocom.org/osmo-bts/commit/?id=cc4a08bdc753492c4367e4786a573a6cdf84b56e commit cc4a08bdc753492c4367e4786a573a6cdf84b56e Author: Holger Hans Peter Freyther <holger at moiji-mobile.com> Date: Mon Sep 21 09:52:36 2015 +0200 audio/rsl: Include the connection identifier in the DLCX ind I have traces that include the connection identifier in the DLCX indication. ----------------------------------------------------------------------- Summary of changes: src/common/rsl.c | 41 ++++++++++++++++++++++++++++++++++++++--- src/osmo-bts-sysmo/tch.c | 8 ++++++++ 2 files changed, 46 insertions(+), 3 deletions(-) hooks/post-receive -- Osmocom BTS-side code (Abis, scheduling, ...)