Fix AudioEncoderOpus::RecreateEncoderInstance() referring to old config_

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 75655bb0154a6c6ec86f5d4d90b20bed0ece58ad..bac963f4e8e2de9ff047ac78d889e9b187c6469e 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -448,6 +448,7 @@ size_t AudioEncoderOpus::SufficientOutputBufferSize() const {
 bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
   if (!config.IsOk())
     return false;
+  config_ = config;

[elad.alon_webrtc.org, 2017/03/13 14:50:31]

Saving this for last would make more sense if we had a rollback mechanism for
problems that might arise later in the function. But since we either reach this
or crash the application, we might as well do it first.

[kwiberg-webrtc, 2017/03/14 07:04:35]

Acknowledged.

   if (inst_)
     RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
   input_buffer_.clear();
@@ -474,7 +475,6 @@ bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
   RTC_CHECK_EQ(0,
                WebRtcOpus_SetPacketLossRate(
                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
-  config_ = config;
 
   num_channels_to_encode_ = NumChannels();
   next_frame_length_ms_ = config_.frame_size_ms;