master

prczeroPulseAudio upgrade to v17.0 part2

ea908dbb创建于 2025年1月15日历史提交

文件	最后提交记录	最后更新时间
headerlist.c	Rename functions with "tostring" in the name to one with "to_string" to conform with the convention. component: core <EP-E358F00C1D9A449EAE69225B9D2530F8> BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=88833 Signed-off-by: Deepak Srivastava <srivastava.d@samsung.com>	10 年前
headerlist.h	update FSF addresses to FSF web page FSF addresses used in PA sources are no longer valid and rpmlint generates numerous warnings during packaging because of this. This patch changes all FSF addresses to FSF web page according to the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html Done automatically by sed-ing through sources.	11 年前
meson.build	rtp: Add a GStreamer-based RTP implementation This adds a GStreamer-based RTP implementation to replace our own. The original implementation is retained for cases where it is not possible to include GStreamer as a dependency. The idea with this is to be able to start supporting more advanced RTP features such as RTCP, non-PCM audio, and potentially synchronised playback. Signed-off-by: Arun Raghavan <arun@arunraghavan.net>	6 年前
module-rtp-recv.c	PulseAudio upgrade to v17.0 part2 Signed-off-by: Robin <chenhongjian3@h-partners.com>	1 年前
module-rtp-send.c	PulseAudio upgrade to v17.0 part2 Signed-off-by: Robin <chenhongjian3@h-partners.com>	1 年前
rfc2327.txt	add an RTP sender module git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@712 fefdeb5f-60dc-0310-8127-8f9354f1896f	19 年前
rfc2974.txt	add an RTP sender module git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@712 fefdeb5f-60dc-0310-8127-8f9354f1896f	19 年前
rfc3550.txt	add an RTP sender module git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@712 fefdeb5f-60dc-0310-8127-8f9354f1896f	19 年前
rfc3551.txt	add an RTP sender module git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@712 fefdeb5f-60dc-0310-8127-8f9354f1896f	19 年前
rtp-common.c	rtp: Enable support for OPUS Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/510>	4 年前
rtp-gstreamer.c	rtp: Initialize adapter to NULL for early pipeline error If gstreamer pipeline immediately returns error, adapter pointer would not be initialized and pa_rtp_recv may crash calling gst_object_unref() on it. Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/615>	4 年前
rtp-native.c	PulseAudio upgrade to v17.0 part2 Signed-off-by: Robin <chenhongjian3@h-partners.com>	1 年前
rtp.h	rtp: Enable support for OPUS Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/510>	4 年前
rtsp_client.c	raop: Add autoreconnect feature This patch adds the autoreconnect feature to the raop module. This is mainly to be used in a server context, but can be used also in a desktop usage context. With autoreconnect feature, the raop module behaves like this: - At initialisation or in case of the RTSP TCP connection lost, it tries to reconnect every 5 seconds - In case of any fatal error, it tries to reconnect every 5 seconds - In UDP mode, if no timing packets received anymore for a long time, RTSP connection is closed, then it tries to reconnect.. - After reconnection, once RTSP session has been established again, playing is resumed automatically. - When the connection is not established yet (or loss), the sink behaves like a null sink. In the source code I called it "autonull", even if autonull is set to autoreconnect param value, it could be split into two different params.	6 年前
rtsp_client.h	raop: Add autoreconnect feature This patch adds the autoreconnect feature to the raop module. This is mainly to be used in a server context, but can be used also in a desktop usage context. With autoreconnect feature, the raop module behaves like this: - At initialisation or in case of the RTSP TCP connection lost, it tries to reconnect every 5 seconds - In case of any fatal error, it tries to reconnect every 5 seconds - In UDP mode, if no timing packets received anymore for a long time, RTSP connection is closed, then it tries to reconnect.. - After reconnection, once RTSP session has been established again, playing is resumed automatically. - When the connection is not established yet (or loss), the sink behaves like a null sink. In the source code I called it "autonull", even if autonull is set to autoreconnect param value, it could be split into two different params.	6 年前
sap.c	PulseAudio upgrade to v17.0 part2 Signed-off-by: Robin <chenhongjian3@h-partners.com>	1 年前
sap.h	update FSF addresses to FSF web page FSF addresses used in PA sources are no longer valid and rpmlint generates numerous warnings during packaging because of this. This patch changes all FSF addresses to FSF web page according to the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html Done automatically by sed-ing through sources.	11 年前
sdp.c	PulseAudio upgrade to v17.0 part2 Signed-off-by: Robin <chenhongjian3@h-partners.com>	1 年前
sdp.h	PulseAudio upgrade to v17.0 part2 Signed-off-by: Robin <chenhongjian3@h-partners.com>	1 年前