Add header extension filtering for WebRtcVoiceEngine/MediaChannel.

talk/media/webrtc/webrtcvideoengine2.cc

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 225 matching lines @@
   if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) {
     LOG(LS_ERROR)
         << "RTX SSRCs exist, but don't cover all SSRCs (unsupported): "
         << sp.ToString();
     return false;
   }
 
   return true;
 }
 
-static std::string RtpExtensionsToString(
-    const std::vector<RtpHeaderExtension>& extensions) {
-  std::stringstream out;
-  out << '{';
-  for (size_t i = 0; i < extensions.size(); ++i) {
-    out << "{" << extensions[i].uri << ": " << extensions[i].id << "}";
-    if (i != extensions.size() - 1) {
-      out << ", ";
-    }
-  }
-  out << '}';
-  return out.str();
-}
-
 inline const webrtc::RtpExtension* FindHeaderExtension(
     const std::vector<webrtc::RtpExtension>& extensions,
     const std::string& name) {
   for (const auto& kv : extensions) {
     if (kv.name == name) {
       return &kv;
     }
   }
   return NULL;
 }
@@ skipping 93 matching lines @@
                                    VideoCodec* matching_codec) {
   for (size_t i = 0; i < codecs.size(); ++i) {
     if (requested_codec.Matches(codecs[i])) {
       *matching_codec = codecs[i];
       return true;
     }
   }
   return false;
 }
 
-static bool ValidateRtpHeaderExtensionIds(
-    const std::vector<RtpHeaderExtension>& extensions) {
-  std::set<int> extensions_used;
-  for (size_t i = 0; i < extensions.size(); ++i) {
-    if (extensions[i].id <= 0 || extensions[i].id >= 15 ||
-        !extensions_used.insert(extensions[i].id).second) {
-      LOG(LS_ERROR) << "RTP extensions are with incorrect or duplicate ids.";
-      return false;
-    }
-  }
-  return true;
-}
-
-static bool CompareRtpHeaderExtensionIds(
-    const webrtc::RtpExtension& extension1,
-    const webrtc::RtpExtension& extension2) {
-  // Sorting on ID is sufficient, more than one extension per ID is unsupported.
-  return extension1.id > extension2.id;
-}
-
-static std::vector<webrtc::RtpExtension> FilterRtpExtensions(
-    const std::vector<RtpHeaderExtension>& extensions) {
-  std::vector<webrtc::RtpExtension> webrtc_extensions;
-  for (size_t i = 0; i < extensions.size(); ++i) {
-    // Unsupported extensions will be ignored.
-    if (webrtc::RtpExtension::IsSupportedForVideo(extensions[i].uri)) {
-      webrtc_extensions.push_back(webrtc::RtpExtension(
-          extensions[i].uri, extensions[i].id));
-    } else {
-      LOG(LS_WARNING) << "Unsupported RTP extension: " << extensions[i].uri;
-    }
-  }
-
-  // Sort filtered headers to make sure that they can later be compared
-  // regardless of in which order they were entered.
-  std::sort(webrtc_extensions.begin(), webrtc_extensions.end(),
-            CompareRtpHeaderExtensionIds);
-  return webrtc_extensions;
-}
-
-static bool RtpExtensionsHaveChanged(
-    const std::vector<webrtc::RtpExtension>& before,
-    const std::vector<webrtc::RtpExtension>& after) {
-  if (before.size() != after.size())
-    return true;
-  for (size_t i = 0; i < before.size(); ++i) {
-    if (before[i].id != after[i].id)
-      return true;
-    if (before[i].name != after[i].name)
-      return true;
-  }
-  return false;
-}
-
 std::vector<webrtc::VideoStream>
 WebRtcVideoChannel2::WebRtcVideoSendStream::CreateSimulcastVideoStreams(
     const VideoCodec& codec,
     const VideoOptions& options,
     int max_bitrate_bps,
     size_t num_streams) {
   int max_qp = kDefaultQpMax;
   codec.GetParam(kCodecParamMaxQuantization, &max_qp);
 
   return GetSimulcastConfig(
@@ skipping 1063 matching lines @@
     return false;
   }
 
   send_streams_[ssrc]->MuteStream(mute);
   return true;
 }
 
 bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
     const std::vector<RtpHeaderExtension>& extensions) {
   TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions");
-  LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
-               << RtpExtensionsToString(extensions);

[stefan-webrtc, 2015/12/01 15:36:57]

There might be a point to keep this since all API methods of this class seem to
start with a LOG?

[the sun, 2015/12/01 16:34:45]

I've added log statements in Set[Recv|Send]Parameters instead.

-  if (!ValidateRtpHeaderExtensionIds(extensions))
+  if (!ValidateRtpExtensions(extensions)) {
     return false;
-
-  std::vector<webrtc::RtpExtension> filtered_extensions =
-      FilterRtpExtensions(extensions);
-  if (!RtpExtensionsHaveChanged(recv_rtp_extensions_, filtered_extensions)) {
+  }
+  std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
+      extensions, webrtc::RtpExtension::IsSupportedForVideo, false);
+  if (recv_rtp_extensions_ == filtered_extensions) {
     LOG(LS_INFO) << "Ignoring call to SetRecvRtpHeaderExtensions because "
                     "header extensions haven't changed.";
     return true;
   }
-
-  recv_rtp_extensions_ = filtered_extensions;
+  recv_rtp_extensions_.swap(filtered_extensions);
 
   rtc::CritScope stream_lock(&stream_crit_);
   for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
            receive_streams_.begin();
        it != receive_streams_.end(); ++it) {
     it->second->SetRtpExtensions(recv_rtp_extensions_);
   }
   return true;
 }
 
 bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
     const std::vector<RtpHeaderExtension>& extensions) {
   TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendRtpHeaderExtensions");
-  LOG(LS_INFO) << "SetSendRtpHeaderExtensions: "
-               << RtpExtensionsToString(extensions);
-  if (!ValidateRtpHeaderExtensionIds(extensions))
+  if (!ValidateRtpExtensions(extensions)) {
     return false;
-
-  std::vector<webrtc::RtpExtension> filtered_extensions =
-      FilterRtpExtensions(FilterRedundantRtpExtensions(
-          extensions, kBweExtensionPriorities, kBweExtensionPrioritiesLength));
-  if (!RtpExtensionsHaveChanged(send_rtp_extensions_, filtered_extensions)) {
-    LOG(LS_INFO) << "Ignoring call to SetSendRtpHeaderExtensions because "
+  }
+  std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
+      extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
+  if (send_rtp_extensions_ == filtered_extensions) {
+    LOG(LS_INFO) << "Ignoring call to SetRecvRtpHeaderExtensions because "
                     "header extensions haven't changed.";
     return true;
   }
-
-  send_rtp_extensions_ = filtered_extensions;
+  send_rtp_extensions_.swap(filtered_extensions);
 
   const webrtc::RtpExtension* cvo_extension = FindHeaderExtension(
       send_rtp_extensions_, kRtpVideoRotationHeaderExtension);
 
   rtc::CritScope stream_lock(&stream_crit_);
   for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
            send_streams_.begin();
        it != send_streams_.end(); ++it) {
     it->second->SetRtpExtensions(send_rtp_extensions_);
     it->second->SetApplyRotation(!cvo_extension);
@@ skipping 1194 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO