Rent-A-Codec: Reference count the shared iSAC bandwidth estimation state

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 
 namespace webrtc {
 
 template <typename T>
 typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
     const CodecInst& codec_inst,
-    LockedIsacBandwidthInfo* bwinfo) {
+    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
   typename AudioEncoderIsacT<T>::Config config;
   config.bwinfo = bwinfo;
   config.payload_type = codec_inst.pltype;
   config.sample_rate_hz = codec_inst.plfreq;
   config.frame_size_ms =
       rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
   config.adaptive_mode = (codec_inst.rate == -1);
   if (codec_inst.rate != -1)
     config.bit_rate = codec_inst.rate;
   return config;
@@ skipping 27 matching lines @@
       return false;
   }
 }
 
 template <typename T>
 AudioEncoderIsacT<T>::AudioEncoderIsacT(const Config& config) {
   RecreateEncoderInstance(config);
 }
 
 template <typename T>
-AudioEncoderIsacT<T>::AudioEncoderIsacT(const CodecInst& codec_inst,
-                                        LockedIsacBandwidthInfo* bwinfo)
+AudioEncoderIsacT<T>::AudioEncoderIsacT(
+    const CodecInst& codec_inst,
+    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
     : AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
 
 template <typename T>
 AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
   RTC_CHECK_EQ(0, T::Free(isac_state_));
 }
 
 template <typename T>
 size_t AudioEncoderIsacT<T>::MaxEncodedBytes() const {
   return kSufficientEncodeBufferSizeBytes;
@@ skipping 101 matching lines @@
   // we get an encoding that isn't bit-for-bit identical with what a combined
   // encoder+decoder object produces.
   RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
 
   config_ = config;
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_