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We already cherry-picked this when we vendored 7fff587a09. Upstream commit: https://webrtc.googlesource.com/src/+/e94c7da1df402ab0193fe5bf010646c7eb08b629 Revert "Return audio stats regarless if we have a codec." This reverts commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf. Reason for revert: breaks downstream test Original change's description: > Return audio stats regarless if we have a codec. > > Bug: b/331602608 > Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743 > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#42964} Bug: b/331602608 Change-Id: Ia87ef3b3066e1373654e1f0d96726217e7ed4117 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361761 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#42965}
50 lines
2.1 KiB
Diff
50 lines
2.1 KiB
Diff
From: Dan Minor <dminor@mozilla.com>
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Date: Tue, 27 Mar 2018 15:43:00 -0400
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Subject: Bug 1376873 - Disable Mid support in RtpDemuxer; r=mjf
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The only use of Mid in the current webrtc.org code is in the unit tests.
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RtpStreamReceiverController only allows adding sinks using SSRCs. Because
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of this, we'll end up dropping packets in the RtpDemuxer with the current
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code as none of our Mids will be recognized.
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Tip of webrtc.org fully supports using Mids, so we'll be able to enable this
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code again after the next update.
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Differential Revision: https://phabricator.services.mozilla.com/D7442
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Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b3ba8452e77105c72f6ddbc49cbe5a53dbea1507
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---
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call/rtp_demuxer.cc | 7 ++++++-
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1 file changed, 6 insertions(+), 1 deletion(-)
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diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc
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index 5dce864e7a..90a156f588 100644
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--- a/call/rtp_demuxer.cc
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+++ b/call/rtp_demuxer.cc
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@@ -282,13 +282,17 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
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// RSID and RRID are routed to the same sinks. If an RSID is specified on a
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// repair packet, it should be ignored and the RRID should be used.
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std::string packet_mid, packet_rsid;
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- bool has_mid = use_mid_ && packet.GetExtension<RtpMid>(&packet_mid);
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+ //bool has_mid = use_mid_ && packet.GetExtension<RtpMid>(&packet_mid);
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bool has_rsid = packet.GetExtension<RepairedRtpStreamId>(&packet_rsid);
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if (!has_rsid) {
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has_rsid = packet.GetExtension<RtpStreamId>(&packet_rsid);
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}
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uint32_t ssrc = packet.Ssrc();
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+ // Mid support is half-baked in branch 64. RtpStreamReceiverController only
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+ // supports adding sinks by ssrc, so our mids will never show up in
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+ // known_mids_, causing us to drop packets here.
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+#if 0
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// The BUNDLE spec says to drop any packets with unknown MIDs, even if the
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// SSRC is known/latched.
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if (has_mid && known_mids_.find(packet_mid) == known_mids_.end()) {
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@@ -362,6 +366,7 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
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}
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}
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+#endif
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// We trust signaled SSRC more than payload type which is likely to conflict
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// between streams.
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const auto ssrc_sink_it = sink_by_ssrc_.find(ssrc);
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