Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 234fd7fce50207a614d756dce1b6972d5a11c652..1023bf8bb20ce05af91433c78decd931c3bcc55a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -16,7 +16,6 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {
@@ -120,31 +119,6 @@
 
 RTPPayloadRegistry::~RTPPayloadRegistry() = default;
 
-void RTPPayloadRegistry::SetAudioReceivePayloads(
-    std::map<int, SdpAudioFormat> codecs) {
-  rtc::CritScope cs(&crit_sect_);
-
-#if RTC_DCHECK_IS_ON
-  RTC_DCHECK(!used_for_video_);
-  used_for_audio_ = true;
-#endif
-
-  payload_type_map_.clear();
-  for (const auto& kv : codecs) {
-    const int& rtp_payload_type = kv.first;
-    const SdpAudioFormat& audio_format = kv.second;
-    const CodecInst ci = SdpToCodecInst(rtp_payload_type, audio_format);
-    RTC_DCHECK(IsPayloadTypeValid(rtp_payload_type));
-    payload_type_map_.insert(
-        std::make_pair(rtp_payload_type, CreatePayloadType(ci)));
-  }
-
-  // Clear the value of last received payload type since it might mean
-  // something else now.
-  last_received_payload_type_ = -1;
-  last_received_media_payload_type_ = -1;
-}
-
 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec,
                                                    bool* created_new_payload) {
   rtc::CritScope cs(&crit_sect_);