acked out 4 changesets (bug 1632489) for android crashes at test_peerConnection_maxFsConstraint.html. CLOSED TREE

Backed out changeset 22ef19e55421 (bug 1632489)
Backed out changeset 8e0260c41854 (bug 1632489)
Backed out changeset cc52615f7682 (bug 1632489)
Backed out changeset 45ad6b848a8f (bug 1632489)

dom/media/tests/mochitest/sdpUtils.js

@@ -238,10 +238,6 @@ var sdputils = {
       .replace(
         /a=extmap:[^\s]* urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id.*\r\n/g,
         ""
-      )
-      .replace(
-        /a=extmap:[^\s]* urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id.*\r\n/g,
-        ""
       );
   },
 
@@ -269,16 +265,6 @@ var sdputils = {
       "a=extmap:" +
       extmap_id[1] +
       "/recvonly urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\n";
-    var extmap_id = offer_sdp.match(
-      "a=extmap:([0-9+])/sendonly urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
-    );
-    ok(extmap_id != null, "Offer contains RID RTP header extension");
-    new_answer_sdp =
-      new_answer_sdp +
-      "a=extmap:" +
-      extmap_id[1] +
-      "/recvonly urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\n";
-
     return new_answer_sdp;
   },
 

media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp

@@ -406,8 +406,6 @@ std::vector<SdpExtmapAttributeList::Extmap> JsepSessionImpl::GetRtpExtensions(
         // TODO: Would it be worth checking that the direction is sane?
         AddVideoRtpExtension(webrtc::RtpExtension::kRtpStreamIdUri,
                              SdpDirectionAttribute::kSendonly);
-        AddVideoRtpExtension(webrtc::RtpExtension::kRepairedRtpStreamIdUri,
-                             SdpDirectionAttribute::kSendonly);
       }
       break;
     default:;

media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_packet.h

@@ -118,14 +118,6 @@ class RtpPacket {
   // Returns empty arrayview on failure.
   rtc::ArrayView<uint8_t> AllocateRawExtension(int id, size_t length);
 
-  // Find an extension |type|.
-  // Returns view of the raw extension or empty view on failure.
-  rtc::ArrayView<const uint8_t> FindExtension(ExtensionType type) const;
-
-  // Find or allocate an extension |type|. Returns view of size |length|
-  // to write raw extension to or an empty view on failure.
-  rtc::ArrayView<uint8_t> AllocateExtension(ExtensionType type, size_t length);
-
   // Reserve size_bytes for payload. Returns nullptr on failure.
   uint8_t* SetPayloadSize(size_t size_bytes);
   // Same as SetPayloadSize but doesn't guarantee to keep current payload.
@@ -143,6 +135,14 @@ class RtpPacket {
   // but does not touch packet own buffer, leaving packet in invalid state.
   bool ParseBuffer(const uint8_t* buffer, size_t size);
 
+  // Find an extension |type|.
+  // Returns view of the raw extension or empty view on failure.
+  rtc::ArrayView<const uint8_t> FindExtension(ExtensionType type) const;
+
+  // Find or allocate an extension |type|. Returns view of size |length|
+  // to write raw extension to or an empty view on failure.
+  rtc::ArrayView<uint8_t> AllocateExtension(ExtensionType type, size_t length);
+
   uint8_t* WriteAt(size_t offset);
   void WriteAt(size_t offset, uint8_t byte);
 

media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc

@@ -1193,63 +1193,14 @@ bool RTPSender::SetFecParameters(const FecProtectionParams& delta_params,
   return true;
 }
 
-static void CopyHeaderAndExtensionsToRtxPacket(const RtpPacketToSend& packet,
-                                               RtpPacketToSend* rtx_packet) {
-  // Set the relevant fixed packet headers. The following are not set:
-  // * Payload type - it is replaced in rtx packets.
-  // * Sequence number - RTX has a separate sequence numbering.
-  // * SSRC - RTX stream has its own SSRC.
-  rtx_packet->SetMarker(packet.Marker());
-  rtx_packet->SetTimestamp(packet.Timestamp());
-
-  // Set the variable fields in the packet header:
-  // * CSRCs - must be set before header extensions.
-  // * Header extensions - replace Rid header with RepairedRid header.
-  const std::vector<uint32_t> csrcs = packet.Csrcs();
-  rtx_packet->SetCsrcs(csrcs);
-  for (int extension_num = kRtpExtensionNone + 1;
-       extension_num < kRtpExtensionNumberOfExtensions; ++extension_num) {
-    auto extension = static_cast<RTPExtensionType>(extension_num);
-
-    // Stream ID header extensions (MID, RSID) are sent per-SSRC. Since RTX
-    // operates on a different SSRC, the presence and values of these header
-    // extensions should be determined separately and not blindly copied.
-    if (extension == kRtpExtensionMid ||
-        extension == kRtpExtensionRtpStreamId) {
-      continue;
-    }
-
-    rtc::ArrayView<const uint8_t> source = packet.FindExtension(extension);
-
-    // Empty extensions should be supported, so not checking |source.empty()|.
-    // TODO: But this does not work in Mozilla's version of libwebrtc. Remove
-    // this check with the next update from tip of libwebrtc.
-    if (source.empty()) {
-      continue;
-    }
-
-    rtc::ArrayView<uint8_t> destination =
-        rtx_packet->AllocateExtension(extension, source.size());
-
-    // Could happen if any:
-    // 1. Extension has 0 length.
-    // 2. Extension is not registered in destination.
-    // 3. Allocating extension in destination failed.
-    if (destination.empty() || source.size() != destination.size()) {
-      continue;
-    }
-
-    std::memcpy(destination.begin(), source.begin(), destination.size());
-  }
-}
-
 std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
     const RtpPacketToSend& packet) {
   // TODO(danilchap): Create rtx packet with extra capacity for SRTP
   // when transport interface would be updated to take buffer class.
   std::unique_ptr<RtpPacketToSend> rtx_packet(new RtpPacketToSend(
-      &rtp_header_extension_map_, max_packet_size_));
+      &rtp_header_extension_map_, packet.size() + kRtxHeaderSize));
   // Add original RTP header.
+  rtx_packet->CopyHeaderFrom(packet);
   {
     rtc::CritScope lock(&send_critsect_);
     if (!sending_media_)
@@ -1268,18 +1219,8 @@ std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
 
     // Replace SSRC.
     rtx_packet->SetSsrc(*ssrc_rtx_);
-
-    CopyHeaderAndExtensionsToRtxPacket(packet, rtx_packet.get());
-
-    // Copy rtp-stream-id from packet to repaired-rtp-stream-id
-    if (rtp_header_extension_map_.IsRegistered(kRtpExtensionRtpStreamId) &&
-        rtp_header_extension_map_.IsRegistered(kRtpExtensionRepairedRtpStreamId)) {
-      std::string rid;
-      if (packet.GetExtension<RtpStreamId>(&rid)) {
-        rtx_packet->SetExtension<RepairedRtpStreamId>(rid);
-      }
-    }
   }
+
   uint8_t* rtx_payload =
       rtx_packet->AllocatePayload(packet.payload_size() + kRtxHeaderSize);
   RTC_DCHECK(rtx_payload);

modules/libpref/init/all.js

@@ -427,14 +427,13 @@ pref("media.videocontrols.picture-in-picture.video-toggle.min-video-secs", 45);
   pref("media.navigator.video.use_remb", true);
   #ifdef EARLY_BETA_OR_EARLIER
     pref("media.navigator.video.use_transport_cc", true);
-    pref("media.peerconnection.video.use_rtx", true);
   #else
     pref("media.navigator.video.use_transport_cc", false);
-    pref("media.peerconnection.video.use_rtx", false);
   #endif
   pref("media.navigator.video.use_tmmbr", false);
   pref("media.navigator.audio.use_fec", true);
   pref("media.navigator.video.red_ulpfec_enabled", false);
+  pref("media.peerconnection.video.use_rtx", false);
 
   #ifdef NIGHTLY_BUILD
     pref("media.peerconnection.sdp.parser", "sipcc");