commit 4f08faae82ccb0a252e83aec2baa9598c9f1716a
author Anton Sukhanov <sukhanov@webrtc.org> Tue May 21 18:12:57 2019
committer Commit Bot <commit-bot@chromium.org> Tue May 21 18:58:33 2019
tree 6ac64aa22fd70c96572a25c4f6093b7282320e06
parent 4880e157074cdba8d39b2aabb26879df5216da58

Introduce MediaTransportConfig

Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.

TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.


Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}

pc/channel_manager.h

diff --git a/pc/channel_manager.h b/pc/channel_manager.h
index a749b7f..34f9013 100644
--- a/pc/channel_manager.h
+++ b/pc/channel_manager.h
@@ -12,13 +12,14 @@
 #define PC_CHANNEL_MANAGER_H_
 
 #include <stdint.h>
+
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "api/audio_options.h"
 #include "api/crypto/crypto_options.h"
-#include "api/media_transport_interface.h"
+#include "api/media_transport_config.h"
 #include "call/call.h"
 #include "media/base/codec.h"
 #include "media/base/media_channel.h"
@@ -95,7 +96,7 @@
       webrtc::Call* call,
       const cricket::MediaConfig& media_config,
       webrtc::RtpTransportInternal* rtp_transport,
-      webrtc::MediaTransportInterface* media_transport,
+      const webrtc::MediaTransportConfig& media_transport_config,
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       bool srtp_required,
@@ -112,7 +113,7 @@
       webrtc::Call* call,
       const cricket::MediaConfig& media_config,
       webrtc::RtpTransportInternal* rtp_transport,
-      webrtc::MediaTransportInterface* media_transport,
+      const webrtc::MediaTransportConfig& media_transport_config,
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       bool srtp_required,