Send audio and video codecs to RTPPayloadRegistry

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

 /*
  *  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_impl.h"
 
 #include <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
 #include "webrtc/base/logging.h"
+#include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 
 namespace webrtc {
 
 using RtpUtility::Payload;
 
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
     Clock* clock,
@@ skipping 44 matching lines @@
 
   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
 }
 
 RtpReceiverImpl::~RtpReceiverImpl() {
   for (int i = 0; i < num_csrcs_; ++i) {
     cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
   }
 }
 
-int32_t RtpReceiverImpl::RegisterReceivePayload(
-    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-    const int8_t payload_type,
-    const uint32_t frequency,
-    const size_t channels,
-    const uint32_t rate) {
+int32_t RtpReceiverImpl::RegisterReceivePayload(const CodecInst& audio_codec) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
 
   // TODO(phoglund): Try to streamline handling of the RED codec and some other
   // cases which makes it necessary to keep track of whether we created a
   // payload or not.
   bool created_new_payload = false;
   int32_t result = rtp_payload_registry_->RegisterReceivePayload(
-      payload_name, payload_type, frequency, channels, rate,
-      &created_new_payload);
+      audio_codec, &created_new_payload);
   if (created_new_payload) {
-    if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type,
-                                                     frequency) != 0) {
-      LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/"
-                    << static_cast<int>(payload_type);
+    if (rtp_media_receiver_->OnNewPayloadTypeCreated(audio_codec) != 0) {
+      LOG(LS_ERROR) << "Failed to register payload: " << audio_codec.plname
+                    << "/" << static_cast<int>(audio_codec.pltype);
       return -1;
     }
   }
   return result;
 }
 
+// TODO(magjed): Remove once external code is updated.
+int32_t RtpReceiverImpl::RegisterReceivePayload(
+    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+    const int8_t payload_type,
+    const uint32_t frequency,
+    const size_t channels,
+    const uint32_t rate) {
+  CodecInst codec;
+  codec.pltype = payload_type;
+  strncpy(codec.plname, payload_name, RTP_PAYLOAD_NAME_SIZE);
+  codec.plfreq = frequency;
+  codec.channels = channels;
+  codec.rate = rate;
+  return RegisterReceivePayload(codec);
+}
+
 int32_t RtpReceiverImpl::DeRegisterReceivePayload(
     const int8_t payload_type) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
 }
 
 uint32_t RtpReceiverImpl::SSRC() const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   return ssrc_;
 }
@@ skipping 337 matching lines @@
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
     }
   }
 }
 
 }  // namespace webrtc