NetEq: Ask AudioDecoder for sample rate instead of passing it as an argument

webrtc/modules/audio_coding/neteq/neteq_impl.h

 /*
  *  Copyright (c) 2012 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 110 matching lines @@
 
   int GetAudio(AudioFrame* audio_frame, bool* muted) override;
 
   int RegisterPayloadType(NetEqDecoder codec,
                           const std::string& codec_name,
                           uint8_t rtp_payload_type) override;
 
   int RegisterExternalDecoder(AudioDecoder* decoder,
                               NetEqDecoder codec,
                               const std::string& codec_name,
-                              uint8_t rtp_payload_type,
-                              int sample_rate_hz) override;
+                              uint8_t rtp_payload_type) override;
 
   // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
   // -1 on failure.
   int RemovePayloadType(uint8_t rtp_payload_type) override;
 
   bool SetMinimumDelay(int delay_ms) override;
 
   bool SetMaximumDelay(int delay_ms) override;
 
   int LeastRequiredDelayMs() const override;
@@ skipping 269 matching lines @@
       AudioFrame::kVadPassive;
   std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
       GUARDED_BY(crit_sect_);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_