Commits · 69840241b7ef689da37159badc650e5d6eeb3055 · savoirfairelinux / jami-daemon

Sep 12, 2012
- * #15493: fix FSF address for daemon/src/audio · 69840241
  Tristan Matthews authored 12 years ago
  
  69840241
- * #15503: fix copyright years for daemon/src and daemon/src/audio/* · a775bea9
  Tristan Matthews authored 12 years ago
  
  a775bea9
- * #15502: audio: removed unused echosuppress/echocancel code · e95839bc
  Tristan Matthews authored 12 years ago
  
  e95839bc
Sep 05, 2012
- * #15266: dtmf: Use correct event codes for DTMF over RTP · f7246562
  Tristan Matthews authored 12 years ago
  
  f7246562
Aug 22, 2012
- #14848: Remove explicite reference to audioCodec->getFrameSize · 908e1fb3
  Alexandre Savard authored 12 years ago
  
  move transportRate to audioRtpSession instead of audioSymmetricRtpSession
  908e1fb3
Aug 21, 2012
- #14818: Get MainBuffer as a reference instead of a pointer · fd1583b8
  Alexandre Savard authored 12 years ago
  
  fd1583b8
Aug 16, 2012
- #14615: Remove dynamic casts from audio_rtp_factory · a252e8c4
  Alexandre Savard authored 12 years ago
  
  a252e8c4
Aug 15, 2012
- #14615: Make RTP sessions to return empty vector if not a secured session · 270493e1
  Alexandre Savard authored 12 years ago
  
  270493e1
- #14615: Get sip account directly in audio rtp factory · 6fb4cc6b
  Alexandre Savard authored 12 years ago
  
  6fb4cc6b
- [ #14666] rtp: make transport rate dynamic · 64fbdd66
  Emmanuel Lepage Vallee authored 12 years ago
  
  64fbdd66
- [#14602] codec/rtp: Change delay from 20 ms to 10 ms (G729 and Opus) · 8beace9e
  Emmanuel Lepage Vallee authored 12 years ago
  
  8beace9e
Aug 08, 2012
- * #14455: audiortprecord: don't use audio processing objects after they've been destroyed · b5cbeb4d
  Tristan Matthews authored 12 years ago
  
  b5cbeb4d
Aug 07, 2012
- * #14455: audiortp: don't destroy AudioCodec unless holding audio codec mutex · 31c50af3
  Tristan Matthews authored 12 years ago
  
  31c50af3
- * #14455: audiortp: don't destroy noise suppress if not holding the audioProcessMutex · 1b45e68e
  Tristan Matthews authored 12 years ago
  
  1b45e68e
- * #9832: audiortp: remove verbose logging from non-main thread · 336d20a1
  Tristan Matthews authored 12 years ago
  
  336d20a1
Jul 27, 2012
- * #11783: audiortp: fix loud audio crack on callstart/going off hold · 9250f1cc
  Tristan Matthews authored 12 years ago
  
  9250f1cc
Jul 26, 2012
- #13892: Make ZRTP and optional feature · 320e5ae2
  Alexandre Savard authored 12 years ago
  
  320e5ae2
- #13882: Make speexdsp and noise suppression and optional dependency · 6a372a4c
  Alexandre Savard authored 12 years ago
  
  6a372a4c
Jul 19, 2012
- #13678: Fix Zrtp thread may hang when hanging up a call · dfda5df8
  Alexandre Savard authored 12 years ago
  
  dfda5df8
- #13652: fix controlable sip log level for NULL pointer · eac7e9a3
  Alexandre Savard authored 12 years ago
  
  eac7e9a3
Jun 22, 2012
- #12896: Fix echo suppressor memory alocation problem · 64b5de44
  Alexandre Savard authored 12 years ago
  
  64b5de44
Jun 04, 2012
- * #11964: mainbuffer/ringbuffer: cleanup API · 61bf3908
  Tristan Matthews authored 12 years ago
  
  61bf3908
May 31, 2012
- * #11840: audiortp: remove some global symbols/variables · 796d60a8
  Tristan Matthews authored 12 years ago
  
  796d60a8
May 29, 2012

* #10797: send DTMF over RTP as per RFC2833 · f6ced13e

Tristan Matthews authored 12 years ago

They must be sent repeatedly and eventually with an end bit set.
Fixes regression from commit 9a8bc388

f6ced13e

May 18, 2012
- * #11269: merged master into video · 9963c8ff
  Tristan Matthews authored 12 years ago
  
  9963c8ff
May 04, 2012
- #10335: Add a noise suppressor for incoming rtp streams · 75b772b1
  Alexandre Savard authored 12 years ago
  
  75b772b1
May 03, 2012
- #10230: Get back default mainbuffer sampling rate to 8kHz, no need of decoding noise suppressor · c46eeb8a
  Alexandre Savard authored 12 years ago
  
  c46eeb8a
- #10230: Use a different samplerate converter for rtp encoding and decoding · 1bd22d30
  Alexandre Savard authored 12 years ago
  
  1bd22d30
- * #9903: create DynamicPayloadFormat on stack, initialize earlier · 2a9992d7
  Tristan Matthews authored 12 years ago
  
  2a9992d7
May 02, 2012
- * #10213: increase resample buffer size, and check output size when resampling · 8a9f8f03
  Tristan Matthews authored 12 years ago
  
  This avoids a buffer overrun crash when upsampling.
  8a9f8f03
May 01, 2012
- * #10096: srtp: use vectors to simplify key/salt manipulation · d8c8aab1
  Tristan Matthews authored 12 years ago
  
  d8c8aab1
- #10096: add case for non-srtp calls · ee8273da
  Alexandre Savard authored 12 years ago
  
  ee8273da
- #10096: use c++ arrays to store keys in srtp sesssion · b1da3cdb
  Alexandre Savard authored 12 years ago
  
  b1da3cdb
Apr 30, 2012
- #10096: Fix onhold/offhold srtp · 25c754de
  Alexandre Savard authored 12 years ago
  
  25c754de
- #10096: Reinit crypto context when required on INVITE request · 60cea81d
  Alexandre Savard authored 12 years ago
  
  60cea81d
- #100096: Set in/out queue crypto context at initialization, not when starting the thread · 0b470ca6
  Alexandre Savard authored 12 years ago
  
  0b470ca6
- #10096: Update srtp key generation when holding/unholding a call · fe92b2e1
  Alexandre Savard authored 12 years ago
  
  fe92b2e1
Apr 28, 2012
- * #9832: audio_rtp_record_handler: cleanup logging · 3953a176
  Tristan Matthews authored 12 years ago
  
  3953a176
- * #9832: zrtp: cleanup logging · 697dff98
  Tristan Matthews authored 12 years ago
  
  697dff98
- * #9832: AudioZRTPSession: cleanup logging · 6126efbc
  Tristan Matthews authored 12 years ago
  
  6126efbc