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 1023bf8bb20ce05af91433c78decd931c3bcc55a..234fd7fce50207a614d756dce1b6972d5a11c652 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -16,6 +16,7 @@
 #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 {
@@ -119,6 +120,31 @@ RTPPayloadRegistry::RTPPayloadRegistry()
 
 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_);