Replaced the _audio parameter with a strategy.

The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.

In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.

BUG=
TEST=vie/voe_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1001006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc

@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
+
+#include <cstdlib>
+
+namespace webrtc {
+
+RTPReceiverStrategy::RTPReceiverStrategy() {
+  memset(&last_payload_, 0, sizeof(last_payload_));
+}
+
+void RTPReceiverStrategy::GetLastMediaSpecificPayload(
+  ModuleRTPUtility::PayloadUnion* payload) const {
+  memcpy(payload, &last_payload_, sizeof(*payload));
+}
+
+void RTPReceiverStrategy::SetLastMediaSpecificPayload(
+  const ModuleRTPUtility::PayloadUnion& payload) {
+  memcpy(&last_payload_, &payload, sizeof(last_payload_));
+}
+
+}  // namespace webrtc