Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(2047)

Unified Diff: webrtc/api/call/flexfec_receive_stream.h

Issue 2542413002: Generalize FlexfecReceiveStream::Config. (CL1) (Closed)
Patch Set: Rebase. Created 4 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « no previous file | webrtc/call/call.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/api/call/flexfec_receive_stream.h
diff --git a/webrtc/api/call/flexfec_receive_stream.h b/webrtc/api/call/flexfec_receive_stream.h
index 5918f7730fa9b081ecbe09ccf9392972177e19e4..64b1d5903fc78290d04318e7e3c8ce9501ed2af3 100644
--- a/webrtc/api/call/flexfec_receive_stream.h
+++ b/webrtc/api/call/flexfec_receive_stream.h
@@ -12,7 +12,9 @@
#define WEBRTC_API_CALL_FLEXFEC_RECEIVE_STREAM_H_
#include <string>
+#include <vector>
+#include "webrtc/api/call/transport.h"
#include "webrtc/config.h"
namespace webrtc {
@@ -32,10 +34,39 @@ class FlexfecReceiveStream {
int flexfec_bitrate_bps;
};
- // TODO(brandtr): When we add multistream protection, and thus add a
- // FlexfecSendStream class, remove FlexfecConfig from config.h and add
- // the appropriate configs here and in FlexfecSendStream.
- using Config = FlexfecConfig;
+ struct Config {
+ std::string ToString() const;
+
+ // Payload type for FlexFEC.
+ int payload_type = -1;
+
+ // SSRC for FlexFEC stream to be received.
+ uint32_t remote_ssrc = 0;
+
+ // Vector containing a single element, corresponding to the SSRC of the
+ // media stream being protected by this FlexFEC stream. The vector MUST have
+ // size 1.
+ //
+ // TODO(brandtr): Update comment above when we support multistream
+ // protection.
+ std::vector<uint32_t> protected_media_ssrcs;
+
+ // SSRC for RTCP reports to be sent.
+ uint32_t local_ssrc = 0;
+
+ // What RTCP mode to use in the reports.
+ RtcpMode rtcp_mode = RtcpMode::kCompound;
+
+ // Transport for outgoing RTCP packets.
+ Transport* rtcp_send_transport = nullptr;
+
+ // |transport_cc| is true whenever the send-side BWE RTCP feedback message
+ // has been negotiated. This is a prerequisite for enabling send-side BWE.
+ bool transport_cc = false;
+
+ // RTP header extensions that have been negotiated for this track.
+ std::vector<RtpExtension> extensions;
+ };
// Starts stream activity.
// When a stream is active, it can receive and process packets.
« no previous file with comments | « no previous file | webrtc/call/call.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698