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Fix some comment blocks for easier merging

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Jim Gustafson 2024-06-21 13:52:48 -07:00 committed by GitHub
parent 67c77356bc
commit 1357e97179
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2 changed files with 89 additions and 78 deletions
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149
media/engine/webrtc_video_engine.cc
View file

@ -809,7 +809,6 @@ std::vector<VideoCodec> WebRtcVideoEngine::recv_codecs(bool include_rtx) const {
include_rtx, trials_);
}
// RingRTC change to disable unused header extensions
std::vector<webrtc::RtpHeaderExtensionCapability>
WebRtcVideoEngine::GetRtpHeaderExtensions() const {
std::vector<webrtc::RtpHeaderExtensionCapability> result;
@ -821,6 +820,7 @@ WebRtcVideoEngine::GetRtpHeaderExtensions() const {
webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kVideoRotationUri,
webrtc::RtpExtension::kTransportSequenceNumberUri,
// RingRTC change to disable unused header extensions
// webrtc::RtpExtension::kPlayoutDelayUri,
// webrtc::RtpExtension::kVideoContentTypeUri,
// webrtc::RtpExtension::kVideoTimingUri,
@ -831,29 +831,32 @@ WebRtcVideoEngine::GetRtpHeaderExtensions() const {
}) {
result.emplace_back(uri, id++, webrtc::RtpTransceiverDirection::kSendRecv);
}
// for (const auto& uri : {webrtc::RtpExtension::kAbsoluteCaptureTimeUri}) {
// result.emplace_back(uri, id, webrtc::RtpTransceiverDirection::kStopped);
// }
// result.emplace_back(webrtc::RtpExtension::kGenericFrameDescriptorUri00, id,
// IsEnabled(trials_, "WebRTC-GenericDescriptorAdvertised")
// ? webrtc::RtpTransceiverDirection::kSendRecv
// : webrtc::RtpTransceiverDirection::kStopped);
// result.emplace_back(
// webrtc::RtpExtension::kDependencyDescriptorUri, id,
// IsEnabled(trials_, "WebRTC-DependencyDescriptorAdvertised")
// ? webrtc::RtpTransceiverDirection::kSendRecv
// : webrtc::RtpTransceiverDirection::kStopped);
// result.emplace_back(
// webrtc::RtpExtension::kVideoLayersAllocationUri, id,
// IsEnabled(trials_, "WebRTC-VideoLayersAllocationAdvertised")
// ? webrtc::RtpTransceiverDirection::kSendRecv
// : webrtc::RtpTransceiverDirection::kStopped);
//
// // VideoFrameTrackingId is a test-only extension.
// if (IsEnabled(trials_, "WebRTC-VideoFrameTrackingIdAdvertised")) {
// result.emplace_back(webrtc::RtpExtension::kVideoFrameTrackingIdUri, id,
// webrtc::RtpTransceiverDirection::kSendRecv);
// }
// RingRTC change to disable unused header extensions
#if 0
for (const auto& uri : {webrtc::RtpExtension::kAbsoluteCaptureTimeUri}) {
result.emplace_back(uri, id, webrtc::RtpTransceiverDirection::kStopped);
}
result.emplace_back(webrtc::RtpExtension::kGenericFrameDescriptorUri00, id,
IsEnabled(trials_, "WebRTC-GenericDescriptorAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
result.emplace_back(
webrtc::RtpExtension::kDependencyDescriptorUri, id,
IsEnabled(trials_, "WebRTC-DependencyDescriptorAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
result.emplace_back(
webrtc::RtpExtension::kVideoLayersAllocationUri, id,
IsEnabled(trials_, "WebRTC-VideoLayersAllocationAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
// VideoFrameTrackingId is a test-only extension.
if (IsEnabled(trials_, "WebRTC-VideoFrameTrackingIdAdvertised")) {
result.emplace_back(webrtc::RtpExtension::kVideoFrameTrackingIdUri, id,
webrtc::RtpTransceiverDirection::kSendRecv);
}
#endif // RingRTC change to disable unused header extensions
return result;
}
@ -3104,54 +3107,56 @@ bool WebRtcVideoReceiveChannel::MaybeCreateDefaultReceiveStream(
// know what stream it associates with, and we shouldn't ever create an
// implicit channel for these.
// RingRTC change to not process unsignaled SSRCs
// bool is_rtx_payload = false;
// for (auto& codec : recv_codecs_) {
// if (packet.PayloadType() == codec.ulpfec.red_rtx_payload_type ||
// packet.PayloadType() == codec.ulpfec.ulpfec_payload_type) {
// return false;
// }
//
// if (packet.PayloadType() == codec.rtx_payload_type) {
// is_rtx_payload = true;
// break;
// }
// }
//
// if (is_rtx_payload) {
// // As we don't support receiving simulcast there can only be one RTX
// // stream, which will be associated with unsignaled media stream.
// absl::optional<uint32_t> current_default_ssrc = GetUnsignaledSsrc();
// if (current_default_ssrc) {
// FindReceiveStream(*current_default_ssrc)->UpdateRtxSsrc(packet.Ssrc());
// } else {
// // Received unsignaled RTX packet before a media packet. Create a default
// // stream with a "random" SSRC and the RTX SSRC from the packet. The
// // stream will be recreated on the first media packet, unless we are
// // extremely lucky and used the right media SSRC.
// ReCreateDefaultReceiveStream(/*ssrc =*/14795, /*rtx_ssrc=*/packet.Ssrc());
// }
// return true;
// } else {
// // Ignore unknown ssrcs if we recently created an unsignalled receive
// // stream since this shouldn't happen frequently. Getting into a state
// // of creating decoders on every packet eats up processing time (e.g.
// // https://crbug.com/1069603) and this cooldown prevents that.
// if (last_unsignalled_ssrc_creation_time_ms_.has_value()) {
// int64_t now_ms = rtc::TimeMillis();
// if (now_ms - last_unsignalled_ssrc_creation_time_ms_.value() <
// kUnsignaledSsrcCooldownMs) {
// // We've already created an unsignalled ssrc stream within the last
// // 0.5 s, ignore with a warning.
// RTC_LOG(LS_WARNING)
// << "Another unsignalled ssrc packet arrived shortly after the "
// << "creation of an unsignalled ssrc stream. Dropping packet.";
// return false;
// }
// }
// }
// // RTX SSRC not yet known.
// ReCreateDefaultReceiveStream(packet.Ssrc(), absl::nullopt);
// last_unsignalled_ssrc_creation_time_ms_ = rtc::TimeMillis();
#if 0
bool is_rtx_payload = false;
for (auto& codec : recv_codecs_) {
if (packet.PayloadType() == codec.ulpfec.red_rtx_payload_type ||
packet.PayloadType() == codec.ulpfec.ulpfec_payload_type) {
return false;
}
if (packet.PayloadType() == codec.rtx_payload_type) {
is_rtx_payload = true;
break;
}
}
if (is_rtx_payload) {
// As we don't support receiving simulcast there can only be one RTX
// stream, which will be associated with unsignaled media stream.
absl::optional<uint32_t> current_default_ssrc = GetUnsignaledSsrc();
if (current_default_ssrc) {
FindReceiveStream(*current_default_ssrc)->UpdateRtxSsrc(packet.Ssrc());
} else {
// Received unsignaled RTX packet before a media packet. Create a default
// stream with a "random" SSRC and the RTX SSRC from the packet. The
// stream will be recreated on the first media packet, unless we are
// extremely lucky and used the right media SSRC.
ReCreateDefaultReceiveStream(/*ssrc =*/14795, /*rtx_ssrc=*/packet.Ssrc());
}
return true;
} else {
// Ignore unknown ssrcs if we recently created an unsignalled receive
// stream since this shouldn't happen frequently. Getting into a state
// of creating decoders on every packet eats up processing time (e.g.
// https://crbug.com/1069603) and this cooldown prevents that.
if (last_unsignalled_ssrc_creation_time_ms_.has_value()) {
int64_t now_ms = rtc::TimeMillis();
if (now_ms - last_unsignalled_ssrc_creation_time_ms_.value() <
kUnsignaledSsrcCooldownMs) {
// We've already created an unsignalled ssrc stream within the last
// 0.5 s, ignore with a warning.
RTC_LOG(LS_WARNING)
<< "Another unsignalled ssrc packet arrived shortly after the "
<< "creation of an unsignalled ssrc stream. Dropping packet.";
return false;
}
}
}
// RTX SSRC not yet known.
ReCreateDefaultReceiveStream(packet.Ssrc(), absl::nullopt);
last_unsignalled_ssrc_creation_time_ms_ = rtc::TimeMillis();
#endif // RingRTC change to not process unsignaled SSRCs
return false;
}

18
media/engine/webrtc_video_engine_unittest.cc
View file

@ -443,19 +443,23 @@ TEST_F(WebRtcVideoEngineTest, SupportsVideoRotationHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoRotationUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsPlayoutDelayHeaderExtension) {
// RingRTC change to disable unused header extensions
TEST_F(WebRtcVideoEngineTest, DISABLED_SupportsPlayoutDelayHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kPlayoutDelayUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsVideoContentTypeHeaderExtension) {
// RingRTC change to disable unused header extensions
TEST_F(WebRtcVideoEngineTest, DISABLED_SupportsVideoContentTypeHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoContentTypeUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsVideoTimingHeaderExtension) {
// RingRTC change to disable unused header extensions
TEST_F(WebRtcVideoEngineTest, DISABLED_SupportsVideoTimingHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoTimingUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsColorSpaceHeaderExtension) {
// RingRTC change to disable unused header extensions
TEST_F(WebRtcVideoEngineTest, DISABLED_SupportsColorSpaceHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kColorSpaceUri, true);
}
@ -922,7 +926,8 @@ void WebRtcVideoEngineTest::ExpectRtpCapabilitySupport(const char* uri,
}
}
TEST_F(WebRtcVideoEngineTest, SendsFeedbackAfterUnsignaledRtxPacket) {
// RingRTC change to not process unsignaled SSRCs
TEST_F(WebRtcVideoEngineTest, DISABLED_SendsFeedbackAfterUnsignaledRtxPacket) {
// Setup a channel with VP8, RTX and transport sequence number header
// extension. Receive stream is not explicitly configured.
AddSupportedVideoCodecType("VP8");
@ -1014,7 +1019,8 @@ TEST_F(WebRtcVideoEngineTest, TooHighReceiveBufferSizeViaFieldTrial) {
receive_channel->SetInterface(nullptr);
}
TEST_F(WebRtcVideoEngineTest, UpdatesUnsignaledRtxSsrcAndRecoversPayload) {
// RingRTC change to not process unsignaled SSRCs
TEST_F(WebRtcVideoEngineTest, DISABLED_UpdatesUnsignaledRtxSsrcAndRecoversPayload) {
// Setup a channel with VP8, RTX and transport sequence number header
// extension. Receive stream is not explicitly configured.
AddSupportedVideoCodecType("VP8");