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voice_engine: dont announce rid/rrid header extensions

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which do not make sense for audio due to lack of support for RTX.

BUG=webrtc:13279

Change-Id: Ida42d8912bf993f01e0dc5c6ffbdbf4b84495c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235061
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35309}
This commit is contained in:
Philipp Hancke
2021-10-14 08:34:53 +02:00
committed by WebRTC LUCI CQ
parent 3677bbaa32
commit bd9106d88f
3 changed files with 4 additions and 25 deletions
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18
audio/audio_send_stream.cc
View File

@ -311,24 +311,6 @@ void AudioSendStream::ConfigureStream(
rtp_rtcp_module_->SetMid(new_config.rtp.mid);
}
// RID RTP header extension
if ((first_time || new_ids.rid != old_ids.rid ||
new_ids.repaired_rid != old_ids.repaired_rid ||
new_config.rtp.rid != old_config.rtp.rid)) {
if (new_ids.rid != 0 || new_ids.repaired_rid != 0) {
if (new_config.rtp.rid.empty()) {
rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(RtpStreamId::Uri());
} else if (new_ids.repaired_rid != 0) {
rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
new_ids.repaired_rid);
} else {
rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
new_ids.rid);
}
}
rtp_rtcp_module_->SetRid(new_config.rtp.rid);
}
if (first_time || new_ids.abs_capture_time != old_ids.abs_capture_time) {
absl::string_view uri = AbsoluteCaptureTimeExtension::Uri();
rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(uri);

1
audio/audio_send_stream_unittest.cc
View File

@ -246,7 +246,6 @@ struct ConfigHelper {
.Times(1);
}
EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
EXPECT_CALL(rtp_rtcp_, SetRid(std::string())).Times(1);
}
void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {

10
media/engine/webrtc_voice_engine.cc
View File

@ -662,12 +662,10 @@ WebRtcVoiceEngine::GetRtpHeaderExtensions() const {
RTC_DCHECK(signal_thread_checker_.IsCurrent());
std::vector<webrtc::RtpHeaderExtensionCapability> result;
int id = 1;
for (const auto& uri :
{webrtc::RtpExtension::kAudioLevelUri,
webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kRidUri,
webrtc::RtpExtension::kRepairedRidUri}) {
for (const auto& uri : {webrtc::RtpExtension::kAudioLevelUri,
webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kMidUri}) {
result.emplace_back(uri, id++, webrtc::RtpTransceiverDirection::kSendRecv);
}
return result;