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/audio/channel_receive.h b/audio/channel_receive.h
index 1f78874..d29f624 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -22,6 +22,7 @@
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
+#include "api/media_transport_config.h"
#include "api/media_transport_interface.h"
#include "api/rtp_receiver_interface.h"
#include "call/rtp_packet_sink_interface.h"
@@ -143,7 +144,7 @@
Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
- MediaTransportInterface* media_transport,
+ const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t remote_ssrc,