From d78789eee235db622adb2a891afd83f477a85dad Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Niels=20M=C3=B6ller?= Date: Mon, 18 Jul 2022 13:14:42 +0200 Subject: [PATCH] Delete old TODOs. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:10198 Change-Id: I7ea6ddedd97db17a9fc8caf6434cf72f6cd0d6ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268761 Reviewed-by: Danil Chapovalov Commit-Queue: Jakob Ivarsson‎ Auto-Submit: Niels Moller Commit-Queue: Niels Moller Reviewed-by: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#37544} --- audio/channel_receive.cc | 2 -- modules/rtp_rtcp/source/rtcp_receiver.cc | 7 ------- modules/rtp_rtcp/source/rtp_rtcp_impl2.cc | 3 --- 3 files changed, 12 deletions(-) diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index 6954447256..7a4ceb289b 100644 --- a/audio/channel_receive.cc +++ b/audio/channel_receive.cc @@ -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(); diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc index 3665b60c8e..0a24481762 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -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. diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc index 6c199af407..b1729c0c3f 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc @@ -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