Delete MediaTransportInterface and DatagramTransportInterface
Bug: webrtc:9719
Change-Id: Ic9936a78ab42f4a9bb4cc3265f0a2cf36946558f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31536}
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 7f9009a..07be28c 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -26,7 +26,6 @@
#include "api/media_stream_interface.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
-#include "api/transport/media/media_transport_config.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/video_content_type.h"
#include "api/video/video_sink_interface.h"
@@ -195,15 +194,8 @@
virtual cricket::MediaType media_type() const = 0;
- // Sets the abstract interface class for sending RTP/RTCP data and
- // interface for media transport (experimental). If media transport is
- // provided, it should be used instead of RTP/RTCP.
- // 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,
- const webrtc::MediaTransportConfig& media_transport_config)
+ // Sets the abstract interface class for sending RTP/RTCP data.
+ virtual void SetInterface(NetworkInterface* iface)
RTC_LOCKS_EXCLUDED(network_interface_crit_);
// Called when a RTP packet is received.
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
@@ -270,10 +262,6 @@
return SetOptionLocked(type, opt, option);
}
- const webrtc::MediaTransportConfig& media_transport_config() const {
- return media_transport_config_;
- }
-
// Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
// Set to true if it's allowed to mix one- and two-byte RTP header extensions
// in the same stream. The setter and getter must only be called from
@@ -361,7 +349,6 @@
nullptr;
rtc::DiffServCodePoint preferred_dscp_
RTC_GUARDED_BY(network_interface_crit_) = rtc::DSCP_DEFAULT;
- webrtc::MediaTransportConfig media_transport_config_;
bool extmap_allow_mixed_ = false;
};