Move the AudioDecoder interface out of NetEq

It belongs with the codecs, next to the AudioEncoder interface.

R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/codecs/audio_decoder.cc

@@ -0,0 +1,72 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+#include <assert.h>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
+                                  size_t encoded_len,
+                                  int16_t* decoded,
+                                  SpeechType* speech_type) {
+  return Decode(encoded, encoded_len, decoded, speech_type);
+}
+
+bool AudioDecoder::HasDecodePlc() const { return false; }
+
+int AudioDecoder::DecodePlc(int num_frames, int16_t* decoded) { return -1; }
+
+int AudioDecoder::IncomingPacket(const uint8_t* payload,
+                                 size_t payload_len,
+                                 uint16_t rtp_sequence_number,
+                                 uint32_t rtp_timestamp,
+                                 uint32_t arrival_timestamp) {
+  return 0;
+}
+
+int AudioDecoder::ErrorCode() { return 0; }
+
+int AudioDecoder::PacketDuration(const uint8_t* encoded, size_t encoded_len) {
+  return kNotImplemented;
+}
+
+int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
+                                          size_t encoded_len) const {
+  return kNotImplemented;
+}
+
+bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
+                                size_t encoded_len) const {
+  return false;
+}
+
+CNG_dec_inst* AudioDecoder::CngDecoderInstance() {
+  FATAL() << "Not a CNG decoder";
+  return NULL;
+}
+
+AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
+  switch (type) {
+    case 0:  // TODO(hlundin): Both iSAC and Opus return 0 for speech.
+    case 1:
+      return kSpeech;
+    case 2:
+      return kComfortNoise;
+    default:
+      assert(false);
+      return kSpeech;
+  }
+}
+
+}  // namespace webrtc