Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

webrtc/audio/audio_send_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.
  */
 
@@ skipping 93 matching lines @@
           EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
               .Times(1);
           EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
               .Times(1);
           EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull()))
               .Times(1);
           EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
               .Times(1);  // Destructor resets the event log
           return channel_proxy_;
         }));
+    SetupMockForSetSendCodecs();
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 200;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   CongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
 
+  void SetupMockForSetSendCodecs() {
+    EXPECT_CALL(voice_engine_, SetVADStatus(_, _, _, _))
+        .WillRepeatedly(Return(0));
+    EXPECT_CALL(voice_engine_, SetFECStatus(_, _)).WillRepeatedly(Return(0));
+    EXPECT_CALL(voice_engine_, SetSendCodec(_, _)).WillRepeatedly(Return(0));
+    EXPECT_CALL(voice_engine_, GetSendCodec(_, _)).WillOnce(Return(-1));

[the sun, 2016/10/17 14:58:22]

Doesn't this mean we'll get an error log line?

[minyue-webrtc, 2016/10/18 09:39:00]

This is only for the first time when GetSendCodec gets called. It means no codec
has been set. I add a comment here. This first call is invoked by
AudioSendStream::SetupSendCodec, and it does not issue error log for -1.

Another thing to add, this behavior has to match the channel id, see new patch
set.

+  }
+
   void SetupMockForSendTelephoneEvent() {
     EXPECT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_,
         SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType))
             .WillOnce(Return(true));
     EXPECT_CALL(*channel_proxy_,
         SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
             .WillOnce(Return(true));
   }
 
@@ skipping 134 matching lines @@
 
   internal::AudioState* internal_audio_state =
       static_cast<internal::AudioState*>(helper.audio_state().get());
   VoiceEngineObserver* voe_observer =
       static_cast<VoiceEngineObserver*>(internal_audio_state);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
   EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
+
+// TODO(minyue): Add tests on logics involved in SetSendCodecs.
+
 }  // namespace test
 }  // namespace webrtc