Move AudioDecoderIsac* to its own files

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

Index: webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
new file mode 100644
index 0000000000000000000000000000000000000000..714fe2d22a452f35725c638b83d29dc2ea391f23
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2015 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_DECODER_ISAC_T_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
+
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+
+namespace webrtc {
+
+template <typename T>
+class AudioDecoderIsacT final : public AudioDecoder {
+ public:
+  AudioDecoderIsacT();
+  explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
+  ~AudioDecoderIsacT() override;
+
+  bool HasDecodePlc() const override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+  void Reset() override;
+  int IncomingPacket(const uint8_t* payload,
+                     size_t payload_len,
+                     uint16_t rtp_sequence_number,
+                     uint32_t rtp_timestamp,
+                     uint32_t arrival_timestamp) override;
+  int ErrorCode() override;
+  size_t Channels() const override;
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  typename T::instance_type* isac_state_;
+  LockedIsacBandwidthInfo* bwinfo_;
+  int decoder_sample_rate_hz_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_