WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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 <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/test/mock_audio_mixer.h"
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
@@ skipping 83 matching lines @@
               .Times(1);
           EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory())
               .WillOnce(ReturnRef(decoder_factory_));
           testing::Expectation expect_set =
               EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
               .Times(1)
               .After(expect_set);
           EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
+          EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_))
+              .WillRepeatedly(
+                  Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
+                    EXPECT_THAT(codecs, testing::IsEmpty());
+                  }));
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 300;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
@@ skipping 222 matching lines @@
 
   EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0));
   EXPECT_CALL(helper.voice_engine(), StopPlayout(_));
   EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream))
       .WillOnce(Return(true));
 
   recv_stream.Start();
 }
 }  // namespace test
 }  // namespace webrtc