Media engine and channel support for per-channel dscp values, specified by RtpParameter
- Similar to network priority
- Still requires MediaConfig.enable_dscp = true (i.e. googDscp == true to peerconnection)
- Needs followups 1) Specify value in chrome renderer js idl 2) disable audio bwe when value differs from video 3)remove googDscp guard
Bug: webrtc:5008
Change-Id: Ibdcbb1183f0ca2ae85e3bced6d0aedbccae3ced4
Reviewed-on: https://webrtc-review.googlesource.com/c/93560
Commit-Queue: Tim Haloun <thaloun@chromium.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25280}diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h
index 9c013cd..be2fe0b 100644
--- a/media/engine/webrtcvoiceengine.h
+++ b/media/engine/webrtcvoiceengine.h
@@ -220,6 +220,7 @@
if (DscpEnabled()) {
rtc_options.dscp = PreferredDscp();
}
+ rtc_options.dscp = PreferredDscp();
rtc_options.info_signaled_after_sent.included_in_feedback =
options.included_in_feedback;
rtc_options.info_signaled_after_sent.included_in_allocation =
@@ -233,6 +234,7 @@
if (DscpEnabled()) {
rtc_options.dscp = PreferredDscp();
}
+
return VoiceMediaChannel::SendRtcp(&packet, rtc_options);
}
@@ -264,6 +266,7 @@
std::vector<AudioCodec> recv_codecs_;
int max_send_bitrate_bps_ = 0;
+ rtc::DiffServCodePoint preferred_dscp_ = rtc::DSCP_DEFAULT;
AudioOptions options_;
absl::optional<int> dtmf_payload_type_;
int dtmf_payload_freq_ = -1;
