[Pkg-voip-commits] [twinkle] 39/73: Imported Debian patch 1:1.4-2

[Peter Colberg]

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pc-guest pushed a commit to branch master
in repository twinkle.

commit 870eebd020fc2500c2dd566e450f433dbc8e39f6
Author: Mark Purcell <msp at debian.org>
Date:   Tue Jan 27 17:30:42 2009 +1100

    Imported Debian patch 1:1.4-2
---
 debian/changelog |  9 +++++++++
 debian/control   | 22 ++++++++++++++++++++--
 2 files changed, 29 insertions(+), 2 deletions(-)

diff --git a/debian/changelog b/debian/changelog
index f1ef81c..7631130 100644
--- a/debian/changelog
+++ b/debian/changelog
@@ -1,3 +1,12 @@
+twinkle (1:1.4-2) experimental; urgency=low
+
+  * Fix "twinkle_1:1.4-1(sparc/experimental): FTBFS: error:
+    readline/readline.h: No such file or directory" 
+    - Add Build-Depends: libreadline-dev & libspeexdsp-dev (Closes: #513135)
+  * Update Description: to match upstream
+
+ -- Mark Purcell <msp at debian.org>  Tue, 27 Jan 2009 17:30:42 +1100
+
 twinkle (1:1.4-1) experimental; urgency=low
 
   * New upstream release
diff --git a/debian/control b/debian/control
index ba57bc4..8a9b1c5 100644
--- a/debian/control
+++ b/debian/control
@@ -8,7 +8,8 @@ Build-Depends: debhelper (>= 4.0.0), cdbs, help2man,
  kdelibs4-dev, libqt3-mt-dev (>= 3:3.3.0), qt3-apps-dev,
  libcommoncpp2-dev (>= 1.4.2), libccrtp-dev (>= 1.7.0), libzrtpcpp-dev (>= 1.3.0),
  libasound2-dev [!kfreebsd-i386 !kfreebsd-amd64 !hurd-i386],
- libgsm1-dev, libsndfile1-dev, libspeex-dev,
+ libgsm1-dev, libsndfile1-dev, libspeex-dev, libspeexdsp-dev,
+ libreadline-dev,
  libboost-regex-dev, libxml2-dev, libxpm-dev, libmagic-dev
 Standards-Version: 3.8.0
 Homepage: http://www.twinklephone.com/
@@ -68,6 +69,17 @@ Description: Voice over Internet Protocol (VoIP) SIP Phone
   User definable number conversion rules
   Simple address book
   Support for UDP and TCP (new) as transport for SIP
+  Presence
+  Instant messaging
+  Simple file transfer with instant message
+  Instant message composition indication
+  Command line interface (CLI)
+ .
+ VoIP security
+  Secure voice communication by ZRTP/SRTP
+  MD5 digest authentication support for all SIP requests
+  AKAv1-MD5 digest authentication support for all SIP requests (new)
+  Identity hiding
  .
  Audio codecs
   G.711 A-law (64 kbps payload, 8 kHz sampling rate)
@@ -78,7 +90,13 @@ Description: Voice over Internet Protocol (VoIP) SIP Phone
   Speex ultra wide band (36 kbps payload, 32 kHz sampling rate)
   G.726 (16, 24, 32 or 40 kbps payload, 8 kHz sampling rate)
  .
- KDE goodies
+ For all codecs the following preprocessing options are available to improve quality at the far end of a call. 
+  Automatic gain control (AGC) (new)
+  Noise reduction (new)
+  Voice activity detection (VAD) (new)
+  Acoustic echo control (AEC) [experimental] (new)
+ .
+ KDE3 goodies
   Balloon pop-up from system tray when a call comes in
   Interface to KAddressBook: dial phone numbers from your address book
   Display names from KAddressBook on incoming calls

-- 
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