Split iSAC encoder/decoder: Test more cases (and make sure they work)

webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.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.
  */
 
@@ skipping 75 matching lines @@
   static inline int16_t SetDecSampRate(instance_type* inst,
                                        uint16_t sample_rate_hz) {
     DCHECK_EQ(sample_rate_hz, kFixSampleRate);
     return 0;
   }
   static inline int16_t SetEncSampRate(instance_type* inst,
                                        uint16_t sample_rate_hz) {
     DCHECK_EQ(sample_rate_hz, kFixSampleRate);
     return 0;
   }
+  static inline void SetEncSampRateInDecoder(instance_type* inst,
+                                             uint16_t sample_rate_hz) {
+    DCHECK_EQ(sample_rate_hz, kFixSampleRate);
+  }
+  static inline void SetInitialBweBottleneck(
+      instance_type* inst,
+      int bottleneck_bits_per_second) {
+    WebRtcIsacfix_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
+  }
   static inline int16_t UpdateBwEstimate(instance_type* inst,
                                          const uint8_t* encoded,
                                          int32_t packet_size,
                                          uint16_t rtp_seq_number,
                                          uint32_t send_ts,
                                          uint32_t arr_ts) {
     return WebRtcIsacfix_UpdateBwEstimate(inst, encoded, packet_size,
                                           rtp_seq_number, send_ts, arr_ts);
   }
   static inline int16_t SetMaxPayloadSize(instance_type* inst,
@@ skipping 42 matching lines @@
   int ErrorCode() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
   int PacketDurationRedundant(const uint8_t* encoded,
                               size_t encoded_len) const override;
   bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
   size_t Channels() const override;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_