admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Ready to support of absolute capture timestamp header extension.

Browse Source 
This does not add it in default SDP offer.

Bug: webrtc:10739
Change-Id: I4e73f4497989fc34f3676927921a4dabb5926096
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169729
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30707}
This commit is contained in:
Minyue Li 2020-03-05 11:33:13 +01:00 committed by Commit Bot
parent 36f4fa7d4c
commit 74dadc1e8e
3 changed files with 14 additions and 4 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
api/rtp_parameters.cc
View File

@ -159,8 +159,7 @@ constexpr int RtpExtension::kOneByteHeaderExtensionMaxValueSize;
bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
return uri == webrtc::RtpExtension::kAudioLevelUri ||
uri == webrtc::RtpExtension::kAbsSendTimeUri ||
// TODO(bugs.webrtc.org/10739): Uncomment once the audio impl is ready.
// uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri ||
uri == webrtc::RtpExtension::kMidUri ||
@ -171,8 +170,7 @@ bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
uri == webrtc::RtpExtension::kAbsSendTimeUri ||
// TODO(bugs.webrtc.org/10739): Uncomment once the video impl is ready.
// uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri ||
uri == webrtc::RtpExtension::kVideoRotationUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri ||

11
audio/audio_send_stream.cc
View File

@ -212,6 +212,8 @@ AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds(
ids.rid = extension.id;
} else if (extension.uri == RtpExtension::kRepairedRidUri) {
ids.repaired_rid = extension.id;
} else if (extension.uri == RtpExtension::kAbsoluteCaptureTimeUri) {
ids.abs_capture_time = extension.id;
}
}
return ids;
@ -323,6 +325,15 @@ void AudioSendStream::ConfigureStream(
rtp_rtcp_module_->SetRid(new_config.rtp.rid);
}
if (first_time || new_ids.abs_capture_time != old_ids.abs_capture_time) {
rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(
kRtpExtensionAbsoluteCaptureTime);
if (new_ids.abs_capture_time) {
rtp_rtcp_module_->RegisterRtpHeaderExtension(
AbsoluteCaptureTimeExtension::kUri, new_ids.abs_capture_time);
}
}
if (!ReconfigureSendCodec(new_config)) {
RTC_LOG(LS_ERROR) << "Failed to set up send codec state.";
}

1
audio/audio_send_stream.h
View File

@ -190,6 +190,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
struct ExtensionIds {
int audio_level = 0;
int abs_send_time = 0;
int abs_capture_time = 0;
int transport_sequence_number = 0;
int mid = 0;
int rid = 0;