Send audio and video codecs to RTPPayloadRegistry

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.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_audio.h"
 
 #include <assert.h>  // assert
 #include <math.h>   // pow()
 #include <string.h>  // memcpy()
 
+#include "webrtc/common_types.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
     RtpData* data_callback) {
   return new RTPReceiverAudio(data_callback);
 }
 
 RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
@@ skipping 69 matching lines @@
 // -   G729      frame         N/A             8,000        10         20
 // -   G729D     frame         N/A             8,000        10         20
 // -   G729E     frame         N/A             8,000        10         20
 // -   GSM       frame         N/A             8,000        20         20
 // -   GSM-EFR   frame         N/A             8,000        20         20
 // -   LPC       frame         N/A             8,000        20         20
 // -   MPA       frame         N/A              var.      var.
 // -
 // -   G7221     frame         N/A
 int32_t RTPReceiverAudio::OnNewPayloadTypeCreated(
-    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-    int8_t payload_type,
-    uint32_t frequency) {
+    const CodecInst& audio_codec) {
   rtc::CritScope lock(&crit_sect_);
 
-  if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) {
-    telephone_event_payload_type_ = payload_type;
+  if (RtpUtility::StringCompare(audio_codec.plname, "telephone-event", 15)) {
+    telephone_event_payload_type_ = audio_codec.pltype;
   }
-  if (RtpUtility::StringCompare(payload_name, "cn", 2)) {
+  if (RtpUtility::StringCompare(audio_codec.plname, "cn", 2)) {
     // We support comfort noise at four different frequencies.
-    if (frequency == 8000) {
-      cng_nb_payload_type_ = payload_type;
-    } else if (frequency == 16000) {
-      cng_wb_payload_type_ = payload_type;
-    } else if (frequency == 32000) {
-      cng_swb_payload_type_ = payload_type;
-    } else if (frequency == 48000) {
-      cng_fb_payload_type_ = payload_type;
+    if (audio_codec.plfreq == 8000) {
+      cng_nb_payload_type_ = audio_codec.pltype;
+    } else if (audio_codec.plfreq == 16000) {
+      cng_wb_payload_type_ = audio_codec.pltype;
+    } else if (audio_codec.plfreq == 32000) {
+      cng_swb_payload_type_ = audio_codec.pltype;
+    } else if (audio_codec.plfreq == 48000) {
+      cng_fb_payload_type_ = audio_codec.pltype;
     } else {
       assert(false);
       return -1;
     }
   }
   return 0;
 }
 
 int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                                          const PayloadUnion& specific_payload,
@@ skipping 165 matching lines @@
     // only one frame in the RED strip the one byte to help NetEq
     return data_callback_->OnReceivedPayloadData(
         payload_data + 1, payload_length - 1, rtp_header);
   }
 
   rtp_header->type.Audio.channel = audio_specific.channels;
   return data_callback_->OnReceivedPayloadData(
       payload_data, payload_length, rtp_header);
 }
 }  // namespace webrtc