Revert r4301

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h

@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2013 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_PAYLOAD_REGISTRY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PAYLOAD_REGISTRY_H_
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+// This strategy deals with the audio/video-specific aspects
+// of payload handling.
+class RTPPayloadStrategy {
+ public:
+  virtual ~RTPPayloadStrategy() {};
+
+  virtual bool CodecsMustBeUnique() const = 0;
+
+  virtual bool PayloadIsCompatible(
+      const ModuleRTPUtility::Payload& payload,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate) const = 0;
+
+  virtual void UpdatePayloadRate(
+      ModuleRTPUtility::Payload* payload,
+      const uint32_t rate) const = 0;
+
+  virtual ModuleRTPUtility::Payload* CreatePayloadType(
+      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+      const int8_t payloadType,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate) const = 0;
+
+  static RTPPayloadStrategy* CreateStrategy(const bool handling_audio);
+
+ protected:
+   RTPPayloadStrategy() {};
+};
+
+class RTPPayloadRegistry {
+ public:
+  // The registry takes ownership of the strategy.
+  RTPPayloadRegistry(const int32_t id,
+                     RTPPayloadStrategy* rtp_payload_strategy);
+  ~RTPPayloadRegistry();
+
+  int32_t RegisterReceivePayload(
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const int8_t payload_type,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate,
+      bool* created_new_payload_type);
+
+  int32_t DeRegisterReceivePayload(
+      const int8_t payload_type);
+
+  int32_t ReceivePayloadType(
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate,
+      int8_t* payload_type) const;
+
+  int32_t PayloadTypeToPayload(
+    const uint8_t payload_type,
+    ModuleRTPUtility::Payload*& payload) const;
+
+  void ResetLastReceivedPayloadTypes() {
+    last_received_payload_type_ = -1;
+    last_received_media_payload_type_ = -1;
+  }
+
+  // Returns true if the new media payload type has not changed.
+  bool ReportMediaPayloadType(uint8_t media_payload_type);
+
+  int8_t red_payload_type() const { return red_payload_type_; }
+  int8_t last_received_payload_type() const {
+    return last_received_payload_type_;
+  }
+  void set_last_received_payload_type(int8_t last_received_payload_type) {
+    last_received_payload_type_ = last_received_payload_type;
+  }
+
+  int8_t last_received_media_payload_type() const {
+    return last_received_media_payload_type_;
+  };
+
+ private:
+  // Prunes the payload type map of the specific payload type, if it exists.
+  void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
+      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+      const size_t payload_name_length,
+      const uint32_t frequency,
+      const uint8_t channels,
+      const uint32_t rate);
+
+  ModuleRTPUtility::PayloadTypeMap payload_type_map_;
+  int32_t id_;
+  scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
+  int8_t  red_payload_type_;
+  int8_t  last_received_payload_type_;
+  int8_t  last_received_media_payload_type_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PAYLOAD_REGISTRY_H_