Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 106 insertions, 0 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
new file mode 100644
index 0000000000..37c3e6e49a
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -0,0 +1,106 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class TelephoneEventHandler;
+
+// This strategy deals with media-specific RTP packet processing.
+// This class is not thread-safe and must be protected by its caller.
+class RTPReceiverStrategy {
+ public:
+  static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
+  static RTPReceiverStrategy* CreateAudioStrategy(
+      RtpData* data_callback,
+      RtpAudioFeedback* incoming_messages_callback);
+
+  virtual ~RTPReceiverStrategy() {}
+
+  // Parses the RTP packet and calls the data callback with the payload data.
+  // Implementations are encouraged to use the provided packet buffer and RTP
+  // header as arguments to the callback; implementations are also allowed to
+  // make changes in the data as necessary. The specific_payload argument
+  // provides audio or video-specific data. The is_first_packet argument is true
+  // if this packet is either the first packet ever or the first in its frame.
+  virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
+                                 const PayloadUnion& specific_payload,
+                                 bool is_red,
+                                 const uint8_t* payload,
+                                 size_t payload_length,
+                                 int64_t timestamp_ms,
+                                 bool is_first_packet) = 0;
+
+  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
+
+  // Retrieves the last known applicable frequency.
+  virtual int GetPayloadTypeFrequency() const = 0;
+
+  // Computes the current dead-or-alive state.
+  virtual RTPAliveType ProcessDeadOrAlive(
+      uint16_t last_payload_length) const = 0;
+
+  // Returns true if we should report CSRC changes for this payload type.
+  // TODO(phoglund): should move out of here along with other payload stuff.
+  virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
+
+  // Notifies the strategy that we have created a new non-RED payload type in
+  // the payload registry.
+  virtual int32_t OnNewPayloadTypeCreated(
+      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+      int8_t payloadType,
+      uint32_t frequency) = 0;
+
+  // Invokes the OnInitializeDecoder callback in a media-specific way.
+  virtual int32_t InvokeOnInitializeDecoder(
+      RtpFeedback* callback,
+      int8_t payload_type,
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const PayloadUnion& specific_payload) const = 0;
+
+  // Checks if the payload type has changed, and returns whether we should
+  // reset statistics and/or discard this packet.
+  virtual void CheckPayloadChanged(int8_t payload_type,
+                                   PayloadUnion* specific_payload,
+                                   bool* should_discard_changes);
+
+  virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
+
+  // Stores / retrieves the last media specific payload for later reference.
+  void GetLastMediaSpecificPayload(PayloadUnion* payload) const;
+  void SetLastMediaSpecificPayload(const PayloadUnion& payload);
+
+ protected:
+  // The data callback is where we should send received payload data.
+  // See ParseRtpPacket. This class does not claim ownership of the callback.
+  // Implementations must NOT hold any critical sections while calling the
+  // callback.
+  //
+  // Note: Implementations may call the callback for other reasons than calls
+  // to ParseRtpPacket, for instance if the implementation somehow recovers a
+  // packet.
+  RTPReceiverStrategy(RtpData* data_callback);
+
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  PayloadUnion last_payload_;
+  RtpData* data_callback_;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_