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

Review URL: https://codereview.webrtc.org/1300863002

Cr-Commit-Position: refs/heads/master@{#9723}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 5ebd4a8..e118fb2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -638,16 +638,48 @@
 }
 
 RTCPSender::BuildResult RTCPSender::BuildREMB(RtcpContext* ctx) {
-  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))
+  // sanity
+  if (ctx->position + 20 + 4 * remb_ssrcs_.size() >= IP_PACKET_SIZE)
     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");