Move AudioDecoderIsac* to its own files

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

+/*
+ *  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_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT()
+    : AudioDecoderIsacT(nullptr) {}
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo)
+    : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
+  RTC_CHECK_EQ(0, T::Create(&isac_state_));
+  T::DecoderInit(isac_state_);
+  if (bwinfo_) {
+    IsacBandwidthInfo bi;
+    T::GetBandwidthInfo(isac_state_, &bi);
+    bwinfo_->Set(bi);
+  }
+}
+
+template <typename T>
+AudioDecoderIsacT<T>::~AudioDecoderIsacT() {
+  RTC_CHECK_EQ(0, T::Free(isac_state_));
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  // We want to crate the illusion that iSAC supports 48000 Hz decoding, while
+  // in fact it outputs 32000 Hz. This is the iSAC fullband mode.
+  if (sample_rate_hz == 48000)
+    sample_rate_hz = 32000;
+  RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
+      << "Unsupported sample rate " << sample_rate_hz;
+  if (sample_rate_hz != decoder_sample_rate_hz_) {
+    RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
+    decoder_sample_rate_hz_ = sample_rate_hz;
+  }
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+template <typename T>
+bool AudioDecoderIsacT<T>::HasDecodePlc() const {
+  return false;
+}
+
+template <typename T>
+size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return T::DecodePlc(isac_state_, decoded, num_frames);
+}
+
+template <typename T>
+void AudioDecoderIsacT<T>::Reset() {
+  T::DecoderInit(isac_state_);
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
+                                         size_t payload_len,
+                                         uint16_t rtp_sequence_number,
+                                         uint32_t rtp_timestamp,
+                                         uint32_t arrival_timestamp) {
+  int ret = T::UpdateBwEstimate(isac_state_, payload, payload_len,
+                                rtp_sequence_number, rtp_timestamp,
+                                arrival_timestamp);
+  if (bwinfo_) {
+    IsacBandwidthInfo bwinfo;
+    T::GetBandwidthInfo(isac_state_, &bwinfo);
+    bwinfo_->Set(bwinfo);
+  }
+  return ret;
+}
+
+template <typename T>
+int AudioDecoderIsacT<T>::ErrorCode() {
+  return T::GetErrorCode(isac_state_);
+}
+
+template <typename T>
+size_t AudioDecoderIsacT<T>::Channels() const {
+  return 1;
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_