Fix AudioEncoderOpus::RecreateEncoderInstance() referring to old config_

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

 /*
  *  Copyright (c) 2014 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 430 matching lines @@
       Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
 }
 
 // If the given config is OK, recreate the Opus encoder instance with those
 // settings, save the config, and return true. Otherwise, do nothing and return
 // false.
 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();
   input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());

[elad.alon_webrtc.org, 2017/03/13 14:50:32]

These functions read from |config_|.

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

Indeed. It should be harmless, since reserve() only impacts performance, but it
certainly isn't nice.

Another acceptable solution (which I probably would have picked) is to simply
remove the reserve() call.

   RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, config.num_channels,
                                            config.application));
   RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config.GetBitrateBps()));
   if (config.fec_enabled) {
     RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
   } else {
     RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
   }
   RTC_CHECK_EQ(
       0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
   // Use the default complexity if the start bitrate is within the hysteresis
   // window.
   complexity_ = config.GetNewComplexity().value_or(config.complexity);
   RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
   if (config.dtx_enabled) {
     RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
   } else {
     RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
   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;
   return true;
 }
 
 void AudioEncoderOpus::SetFrameLength(int frame_length_ms) {
   next_frame_length_ms_ = frame_length_ms;
 }
 
@@ skipping 73 matching lines @@
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc