Move AudioDecoder and related stuff to the api/ directory

webrtc/modules/audio_coding/codecs/audio_decoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_decoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
deleted file mode 100644
index afa5115d5a1740036c566ded3c9c9019b78bb233..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- *  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 <memory>
-#include <utility>
-
-#include "webrtc/base/array_view.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/sanitizer.h"
-#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
-
-namespace webrtc {
-
-AudioDecoder::ParseResult::ParseResult() = default;
-AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
-AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
-                                       int priority,
-                                       std::unique_ptr<EncodedAudioFrame> frame)
-    : timestamp(timestamp), priority(priority), frame(std::move(frame)) {
-  RTC_DCHECK_GE(priority, 0);
-}
-
-AudioDecoder::ParseResult::~ParseResult() = default;
-
-AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=(
-    ParseResult&& b) = default;
-
-std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
-    rtc::Buffer&& payload,
-    uint32_t timestamp) {
-  std::vector<ParseResult> results;
-  std::unique_ptr<EncodedAudioFrame> frame(
-      new LegacyEncodedAudioFrame(this, std::move(payload)));
-  results.emplace_back(timestamp, 0, std::move(frame));
-  return results;
-}
-
-int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
-                         int sample_rate_hz, size_t max_decoded_bytes,
-                         int16_t* decoded, SpeechType* speech_type) {
-  TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
-  rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
-  int duration = PacketDuration(encoded, encoded_len);
-  if (duration >= 0 &&
-      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
-    return -1;
-  }
-  return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
-                        speech_type);
-}
-
-int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
-                                  int sample_rate_hz, size_t max_decoded_bytes,
-                                  int16_t* decoded, SpeechType* speech_type) {
-  TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
-  rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
-  int duration = PacketDurationRedundant(encoded, encoded_len);
-  if (duration >= 0 &&
-      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
-    return -1;
-  }
-  return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
-                                 speech_type);
-}
-
-int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
-                                          size_t encoded_len,
-                                          int sample_rate_hz, int16_t* decoded,
-                                          SpeechType* speech_type) {
-  return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
-                        speech_type);
-}
-
-bool AudioDecoder::HasDecodePlc() const { return false; }
-
-size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
-  return 0;
-}
-
-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) const {
-  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;
-}
-
-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