Send audio and video codecs to RTPPayloadRegistry

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc

 /*
  *  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.
  */
 
@@ skipping 15 matching lines @@
       ssrc_rtx_(0) {}
 
 RTPPayloadRegistry::~RTPPayloadRegistry() {
   while (!payload_type_map_.empty()) {
     RtpUtility::PayloadTypeMap::iterator it = payload_type_map_.begin();
     delete it->second;
     payload_type_map_.erase(it);
   }
 }
 
+int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec,
+                                                   bool* created_new_payload) {
+  return RegisterReceivePayload(
+      audio_codec.plname, audio_codec.pltype, audio_codec.plfreq,
+      audio_codec.channels, std::max(0, audio_codec.rate),
+      created_new_payload);
+}
+
+int32_t RTPPayloadRegistry::RegisterReceivePayload(
+    const VideoCodec& video_codec,
+    bool* created_new_payload) {
+  return RegisterReceivePayload(video_codec.plName, video_codec.plType,
+                                kVideoPayloadTypeFrequency, 0 /* channels */,
+                                0 /* rate */, created_new_payload);
+}
+
 int32_t RTPPayloadRegistry::RegisterReceivePayload(
     const char* const payload_name,
     const int8_t payload_type,
     const uint32_t frequency,
     const size_t channels,
     const uint32_t rate,
     bool* created_new_payload) {
   assert(payload_type >= 0);
   assert(payload_name);
   *created_new_payload = false;
@@ skipping 112 matching lines @@
         }
       } else if (RtpUtility::StringCompare(payload_name, "red", 3)) {
         delete payload;
         payload_type_map_.erase(iterator);
         break;
       }
     }
   }
 }
 
+int32_t RTPPayloadRegistry::ReceivePayloadType(const CodecInst& audio_codec,
+                                               int8_t* payload_type) const {
+  return ReceivePayloadType(audio_codec.plname, audio_codec.plfreq,
+                            audio_codec.channels, std::max(0, audio_codec.rate),
+                            payload_type);
+}
+
+int32_t RTPPayloadRegistry::ReceivePayloadType(const VideoCodec& video_codec,
+                                               int8_t* payload_type) const {
+  return ReceivePayloadType(video_codec.plName, kVideoPayloadTypeFrequency,
+                            0 /* channels */, 0 /* rate */,
+                            payload_type);
+}
+
 int32_t RTPPayloadRegistry::ReceivePayloadType(const char* const payload_name,
                                                const uint32_t frequency,
                                                const size_t channels,
                                                const uint32_t rate,
                                                int8_t* payload_type) const {
   assert(payload_type);
   size_t payload_name_length = strlen(payload_name);
 
   rtc::CritScope cs(&crit_sect_);
 
@@ skipping 284 matching lines @@
 RTPPayloadStrategy* RTPPayloadStrategy::CreateStrategy(
     const bool handling_audio) {
   if (handling_audio) {
     return new RTPPayloadAudioStrategy();
   } else {
     return new RTPPayloadVideoStrategy();
   }
 }
 
 }  // namespace webrtc