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}
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 4930d41..04edd9c 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -22,7 +22,7 @@
#include "api/audio_options.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
-#include "api/media_transport_interface.h"
+#include "api/media_transport_config.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
@@ -193,8 +193,9 @@
// TODO(sukhanov): Currently media transport can co-exist with RTP/RTCP, but
// in the future we will refactor code to send all frames with media
// transport.
- virtual void SetInterface(NetworkInterface* iface,
- webrtc::MediaTransportInterface* media_transport);
+ virtual void SetInterface(
+ NetworkInterface* iface,
+ const webrtc::MediaTransportConfig& media_transport_config);
// Called when a RTP packet is received.
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) = 0;
@@ -261,8 +262,12 @@
return network_interface_->SetOption(type, opt, option);
}
+ const webrtc::MediaTransportConfig& media_transport_config() const {
+ return media_transport_config_;
+ }
+
webrtc::MediaTransportInterface* media_transport() {
- return media_transport_;
+ return media_transport_config_.media_transport;
}
// Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
@@ -331,7 +336,7 @@
nullptr;
rtc::DiffServCodePoint preferred_dscp_
RTC_GUARDED_BY(network_interface_crit_) = rtc::DSCP_DEFAULT;
- webrtc::MediaTransportInterface* media_transport_ = nullptr;
+ webrtc::MediaTransportConfig media_transport_config_;
bool extmap_allow_mixed_ = false;
};