Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"

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.
  */
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #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 {
 
 namespace {
 
 bool PayloadIsCompatible(const RtpUtility::Payload& payload,
                          const CodecInst& audio_codec) {
   if (!payload.audio)
     return false;
@@ skipping 83 matching lines @@
 
 RTPPayloadRegistry::RTPPayloadRegistry()
     : incoming_payload_type_(-1),
       last_received_payload_type_(-1),
       last_received_media_payload_type_(-1),
       rtx_(false),
       ssrc_rtx_(0) {}
 
 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_);
 
 #if RTC_DCHECK_IS_ON
   RTC_DCHECK(!used_for_video_);
   used_for_audio_ = true;
 #endif
 
   *created_new_payload = false;
@@ skipping 264 matching lines @@
     const char* payload_name) const {
   rtc::CritScope cs(&crit_sect_);
   for (const auto& it : payload_type_map_) {
     if (_stricmp(it.second.name, payload_name) == 0)
       return it.first;
   }
   return -1;
 }
 
 }  // namespace webrtc