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}

audio/audio_send_stream.cc

diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index a72292d..942551b 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -21,6 +21,7 @@
 #include "api/call/transport.h"
 #include "api/crypto/frame_encryptor_interface.h"
 #include "api/function_view.h"
+#include "api/media_transport_config.h"
 #include "audio/audio_state.h"
 #include "audio/channel_send.h"
 #include "audio/conversion.h"
@@ -104,7 +105,7 @@
                       voe::CreateChannelSend(clock,
                                              task_queue_factory,
                                              module_process_thread,
-                                             config.media_transport,
+                                             config.media_transport_config,
                                              /*overhead_observer=*/this,
                                              config.send_transport,
                                              rtcp_rtt_stats,
@@ -127,8 +128,7 @@
     std::unique_ptr<voe::ChannelSendInterface> channel_send)
     : clock_(clock),
       worker_queue_(rtp_transport->GetWorkerQueue()),
-      config_(Config(/*send_transport=*/nullptr,
-                     /*media_transport=*/nullptr)),
+      config_(Config(/*send_transport=*/nullptr, MediaTransportConfig())),
       audio_state_(audio_state),
       channel_send_(std::move(channel_send)),
       event_log_(event_log),
@@ -151,15 +151,15 @@
   // time being, we can have either. When media transport is injected, there
   // should be no rtp_transport, and below check should be strengthened to XOR
   // (either rtp_transport or media_transport but not both).
-  RTC_DCHECK(rtp_transport || config.media_transport);
-  if (config.media_transport) {
+  RTC_DCHECK(rtp_transport || config.media_transport_config.media_transport);
+  if (config.media_transport_config.media_transport) {
     // TODO(sukhanov): Currently media transport audio overhead is considered
     // constant, we will not get overhead_observer calls when using
     // media_transport. In the future when we introduce RTP media transport we
     // should make audio overhead interface consistent and work for both RTP and
     // non-RTP implementations.
     audio_overhead_per_packet_bytes_ =
-        config.media_transport->GetAudioPacketOverhead();
+        config.media_transport_config.media_transport->GetAudioPacketOverhead();
   }
   rtp_rtcp_module_ = channel_send_->GetRtpRtcp();
   RTC_DCHECK(rtp_rtcp_module_);