Move AudioDecoder and related stuff to the api/ directory

webrtc/modules/audio_coding/codecs/audio_decoder.h

 /*
  *  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.
  */
 
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_decoder.h instead!
+// TODO(kwiberg): Remove it.
+
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
 
-#include <memory>
-#include <vector>
+#include "webrtc/api/audio_codecs/audio_decoder.h"
 
-#include "webrtc/base/array_view.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/optional.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// This is the interface class for decoders in NetEQ. Each codec type will have
-// and implementation of this class.
-class AudioDecoder {
- public:
-  enum SpeechType {
-    kSpeech = 1,
-    kComfortNoise = 2
-  };
-
-  // Used by PacketDuration below. Save the value -1 for errors.
-  enum { kNotImplemented = -2 };
-
-  AudioDecoder() = default;
-  virtual ~AudioDecoder() = default;
-
-  class EncodedAudioFrame {
-   public:
-    struct DecodeResult {
-      size_t num_decoded_samples;
-      SpeechType speech_type;
-    };
-
-    virtual ~EncodedAudioFrame() = default;
-
-    // Returns the duration in samples-per-channel of this audio frame.
-    // If no duration can be ascertained, returns zero.
-    virtual size_t Duration() const = 0;
-
-    // Decodes this frame of audio and writes the result in |decoded|.
-    // |decoded| must be large enough to store as many samples as indicated by a
-    // call to Duration() . On success, returns an rtc::Optional containing the
-    // total number of samples across all channels, as well as whether the
-    // decoder produced comfort noise or speech. On failure, returns an empty
-    // rtc::Optional. Decode may be called at most once per frame object.
-    virtual rtc::Optional<DecodeResult> Decode(
-        rtc::ArrayView<int16_t> decoded) const = 0;
-  };
-
-  struct ParseResult {
-    ParseResult();
-    ParseResult(uint32_t timestamp,
-                int priority,
-                std::unique_ptr<EncodedAudioFrame> frame);
-    ParseResult(ParseResult&& b);
-    ~ParseResult();
-
-    ParseResult& operator=(ParseResult&& b);
-
-    // The timestamp of the frame is in samples per channel.
-    uint32_t timestamp;
-    // The relative priority of the frame compared to other frames of the same
-    // payload and the same timeframe. A higher value means a lower priority.
-    // The highest priority is zero - negative values are not allowed.
-    int priority;
-    std::unique_ptr<EncodedAudioFrame> frame;
-  };
-
-  // Let the decoder parse this payload and prepare zero or more decodable
-  // frames. Each frame must be between 10 ms and 120 ms long. The caller must
-  // ensure that the AudioDecoder object outlives any frame objects returned by
-  // this call. The decoder is free to swap or move the data from the |payload|
-  // buffer. |timestamp| is the input timestamp, in samples, corresponding to
-  // the start of the payload.
-  virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
-                                                uint32_t timestamp);
-
-  // Decodes |encode_len| bytes from |encoded| and writes the result in
-  // |decoded|. The maximum bytes allowed to be written into |decoded| is
-  // |max_decoded_bytes|. Returns the total number of samples across all
-  // channels. If the decoder produced comfort noise, |speech_type|
-  // is set to kComfortNoise, otherwise it is kSpeech. The desired output
-  // sample rate is provided in |sample_rate_hz|, which must be valid for the
-  // codec at hand.
-  int Decode(const uint8_t* encoded,
-             size_t encoded_len,
-             int sample_rate_hz,
-             size_t max_decoded_bytes,
-             int16_t* decoded,
-             SpeechType* speech_type);
-
-  // Same as Decode(), but interfaces to the decoders redundant decode function.
-  // The default implementation simply calls the regular Decode() method.
-  int DecodeRedundant(const uint8_t* encoded,
-                      size_t encoded_len,
-                      int sample_rate_hz,
-                      size_t max_decoded_bytes,
-                      int16_t* decoded,
-                      SpeechType* speech_type);
-
-  // Indicates if the decoder implements the DecodePlc method.
-  virtual bool HasDecodePlc() const;
-
-  // Calls the packet-loss concealment of the decoder to update the state after
-  // one or several lost packets. The caller has to make sure that the
-  // memory allocated in |decoded| should accommodate |num_frames| frames.
-  virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
-
-  // Resets the decoder state (empty buffers etc.).
-  virtual void Reset() = 0;
-
-  // Notifies the decoder of an incoming packet to NetEQ.
-  virtual int IncomingPacket(const uint8_t* payload,
-                             size_t payload_len,
-                             uint16_t rtp_sequence_number,
-                             uint32_t rtp_timestamp,
-                             uint32_t arrival_timestamp);
-
-  // Returns the last error code from the decoder.
-  virtual int ErrorCode();
-
-  // Returns the duration in samples-per-channel of the payload in |encoded|
-  // which is |encoded_len| bytes long. Returns kNotImplemented if no duration
-  // estimate is available, or -1 in case of an error.
-  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
-
-  // Returns the duration in samples-per-channel of the redandant payload in
-  // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
-  // duration estimate is available, or -1 in case of an error.
-  virtual int PacketDurationRedundant(const uint8_t* encoded,
-                                      size_t encoded_len) const;
-
-  // Detects whether a packet has forward error correction. The packet is
-  // comprised of the samples in |encoded| which is |encoded_len| bytes long.
-  // Returns true if the packet has FEC and false otherwise.
-  virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
-
-  // Returns the actual sample rate of the decoder's output. This value may not
-  // change during the lifetime of the decoder.
-  virtual int SampleRateHz() const = 0;
-
-  // The number of channels in the decoder's output. This value may not change
-  // during the lifetime of the decoder.
-  virtual size_t Channels() const = 0;
-
- protected:
-  static SpeechType ConvertSpeechType(int16_t type);
-
-  virtual int DecodeInternal(const uint8_t* encoded,
-                             size_t encoded_len,
-                             int sample_rate_hz,
-                             int16_t* decoded,
-                             SpeechType* speech_type) = 0;
-
-  virtual int DecodeRedundantInternal(const uint8_t* encoded,
-                                      size_t encoded_len,
-                                      int sample_rate_hz,
-                                      int16_t* decoded,
-                                      SpeechType* speech_type);
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
-};
-
-}  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_