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Delete old TODOs.

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Bug: webrtc:10198
Change-Id: I7ea6ddedd97db17a9fc8caf6434cf72f6cd0d6ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268761
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Auto-Submit: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37544}
This commit is contained in:
Niels Möller 2022-07-18 13:14:42 +02:00 committed by WebRTC LUCI CQ
parent e4bda7d008
commit d78789eee2
3 changed files with 0 additions and 12 deletions
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2
audio/channel_receive.cc
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@ -1099,8 +1099,6 @@ int64_t ChannelReceive::GetRTT() const {
return associated_send_channel_->GetRTT();
}
// TODO(nisse): This method computes RTT based on sender reports, even though
// a receive stream is not supposed to do that.
for (const ReportBlockData& data : report_blocks) {
if (data.report_block().sender_ssrc == remote_ssrc_) {
return data.last_rtt_ms();

7
modules/rtp_rtcp/source/rtcp_receiver.cc
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@ -669,13 +669,6 @@ void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
// If no SR has been received yet, the field is set to zero.
// Receiver rtp_rtcp module is not expected to calculate rtt using
// Sender Reports even if it accidentally can.
// TODO(nisse): Use this way to determine the RTT only when `receiver_only_`
// is false. However, that currently breaks the tests of the
// googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
// delete all dependencies on RTT measurements for audio receive streams, or
// ensure that audio receive streams that need RTT and stats that depend on it
// are configured with an associated audio send stream.
if (send_time_ntp != 0) {
uint32_t delay_ntp = report_block.delay_since_last_sr();
// Local NTP time.

3
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
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@ -274,9 +274,6 @@ RTCPSender::FeedbackState ModuleRtpRtcpImpl2::GetFeedbackState() {
return state;
}
// TODO(nisse): This method shouldn't be called for a receive-only
// stream. Delete rtp_sender_ check as soon as all applications are
// updated.
int32_t ModuleRtpRtcpImpl2::SetSendingStatus(const bool sending) {
if (rtcp_sender_.Sending() != sending) {
// Sends RTCP BYE when going from true to false