Replace BundleFilter with RtpDemuxer in RtpTransport.
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
type-based demuxing. RtpTransport will support MID-based demuxing later.
Each BaseChannel has its own RTP demuxing criteria and when connecting
to the RtpTransport, BaseChannel will register itself as a demuxer sink.
The inheritance model is changed. New inheritance chain:
DtlsSrtpTransport->SrtpTransport->RtpTranpsort
The JsepTransport2 is renamed to JsepTransport.
NOTE:
When RTCP packets are received, Call::DeliverRtcp will be called for
multiple times (webrtc:9035) which is an existing issue. With this CL,
it will become more of a problem and should be fixed.
Bug: webrtc:8587
Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62
Reviewed-on: https://webrtc-review.googlesource.com/65786
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22867}diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index b59de48..b9b2f84 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -30,8 +30,6 @@
defines = []
sources = [
"audiomonitor.h",
- "bundlefilter.cc",
- "bundlefilter.h",
"channel.cc",
"channel.h",
"channelmanager.cc",
@@ -42,8 +40,8 @@
"dtlssrtptransport.h",
"externalhmac.cc",
"externalhmac.h",
- "jseptransport2.cc",
- "jseptransport2.h",
+ "jseptransport.cc",
+ "jseptransport.h",
"jseptransportcontroller.cc",
"jseptransportcontroller.h",
"mediasession.cc",
@@ -76,10 +74,13 @@
"../api:optional",
"../api:ortc_api",
"../api:video_frame_api",
+ "../call:rtp_interfaces",
+ "../call:rtp_receiver",
"../common_video:common_video",
"../media:rtc_data",
"../media:rtc_h264_profile_id",
"../media:rtc_media_base",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:rtc_base",
@@ -270,12 +271,11 @@
testonly = true
sources = [
- "bundlefilter_unittest.cc",
"channel_unittest.cc",
"channelmanager_unittest.cc",
"currentspeakermonitor_unittest.cc",
"dtlssrtptransport_unittest.cc",
- "jseptransport2_unittest.cc",
+ "jseptransport_unittest.cc",
"jseptransportcontroller_unittest.cc",
"mediasession_unittest.cc",
"rtcpmuxfilter_unittest.cc",
@@ -308,9 +308,11 @@
"../api:array_view",
"../api:fakemetricsobserver",
"../api:libjingle_peerconnection_api",
+ "../call:rtp_interfaces",
"../logging:rtc_event_log_api",
"../media:rtc_media_base",
"../media:rtc_media_tests_utils",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../rtc_base:checks",