Reland of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1300863002/ )
Reason for revert:
This wasn't the cause of the breakage. Re-reverting.
https://code.google.com/p/webrtc/issues/detail?id=4923
Original issue's description:
> Revert of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1290573004/ )
>
> Reason for revert:
> A few bots started failing rtc_unittests after this was commited. Ex https://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5048
>
> Original issue's description:
> > Use RtcpPacket to send REMB in RtcpSender
> >
> > BUG=webrtc:2450
> > R=asapersson@webrtc.org
> >
> > Committed: https://chromium.googlesource.com/external/webrtc/+/35ab4baa20a730de71b390008900a16563cbbe8e
>
> TBR=asapersson@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:2450
>
> Committed: https://crrev.com/141c5951f4beda868797c2746002a4b1b267ab2a
> Cr-Commit-Position: refs/heads/master@{#9723}
TBR=asapersson@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2450
Review URL: https://codereview.webrtc.org/1309723002
Cr-Commit-Position: refs/heads/master@{#9754}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index e118fb2..5ebd4a8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -638,48 +638,16 @@
}
RTCPSender::BuildResult RTCPSender::BuildREMB(RtcpContext* ctx) {
- // sanity
- if (ctx->position + 20 + 4 * remb_ssrcs_.size() >= IP_PACKET_SIZE)
+ rtcp::Remb remb;
+ remb.From(ssrc_);
+ for (uint32_t ssrc : remb_ssrcs_)
+ remb.AppliesTo(ssrc);
+ remb.WithBitrateBps(remb_bitrate_);
+
+ PacketBuiltCallback callback(ctx);
+ if (!callback.BuildPacket(remb))
return BuildResult::kTruncated;
- // add application layer feedback
- uint8_t FMT = 15;
- *ctx->AllocateData(1) = 0x80 + FMT;
- *ctx->AllocateData(1) = 206;
-
- *ctx->AllocateData(1) = 0;
- *ctx->AllocateData(1) = static_cast<uint8_t>(remb_ssrcs_.size() + 4);
-
- // Add our own SSRC
- ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), ssrc_);
-
- // Remote SSRC must be 0
- ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), 0);
-
- *ctx->AllocateData(1) = 'R';
- *ctx->AllocateData(1) = 'E';
- *ctx->AllocateData(1) = 'M';
- *ctx->AllocateData(1) = 'B';
-
- *ctx->AllocateData(1) = remb_ssrcs_.size();
- // 6 bit Exp
- // 18 bit mantissa
- uint8_t brExp = 0;
- for (uint32_t i = 0; i < 64; i++) {
- if (remb_bitrate_ <= (0x3FFFFu << i)) {
- brExp = i;
- break;
- }
- }
- const uint32_t brMantissa = (remb_bitrate_ >> brExp);
- *ctx->AllocateData(1) =
- static_cast<uint8_t>((brExp << 2) + ((brMantissa >> 16) & 0x03));
- *ctx->AllocateData(1) = static_cast<uint8_t>(brMantissa >> 8);
- *ctx->AllocateData(1) = static_cast<uint8_t>(brMantissa);
-
- for (size_t i = 0; i < remb_ssrcs_.size(); i++)
- ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), remb_ssrcs_[i]);
-
TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"RTCPSender::REMB");