text: /*
   *  Copyright (c) 2014 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_AUDIO_ENCODER_H_
  #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
  
  #include <algorithm>
  #include <vector>
  
  #include "webrtc/typedefs.h"
  
  namespace webrtc {
  
  // This is the interface class for encoders in AudioCoding module. Each codec
  // type must have an implementation of this class.
  class AudioEncoder {
   public:
    struct EncodedInfoLeaf {
      size_t encoded_bytes = 0;
      uint32_t encoded_timestamp = 0;
      int payload_type = 0;
      bool send_even_if_empty = false;
      bool speech = true;
    };
  
    // This is the main struct for auxiliary encoding information. Each encoded
    // packet should be accompanied by one EncodedInfo struct, containing the
    // total number of |encoded_bytes|, the |encoded_timestamp| and the
    // |payload_type|. If the packet contains redundant encodings, the |redundant|
    // vector will be populated with EncodedInfoLeaf structs. Each struct in the
    // vector represents one encoding; the order of structs in the vector is the
    // same as the order in which the actual payloads are written to the byte
    // stream. When EncoderInfoLeaf structs are present in the vector, the main
    // struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
    // vector.
    struct EncodedInfo : public EncodedInfoLeaf {
      EncodedInfo();
      ~EncodedInfo();
  
      std::vector<EncodedInfoLeaf> redundant;
    };
  
    virtual ~AudioEncoder() = default;
  
    // Returns the maximum number of bytes that can be produced by the encoder
    // at each Encode() call. The caller can use the return value to determine
    // the size of the buffer that needs to be allocated. This value is allowed
    // to depend on encoder parameters like bitrate, frame size etc., so if
    // any of these change, the caller of Encode() is responsible for checking
    // that the buffer is large enough by calling MaxEncodedBytes() again.
    virtual size_t MaxEncodedBytes() const = 0;
  
    // Returns the input sample rate in Hz and the number of input channels.
    // These are constants set at instantiation time.
    virtual int SampleRateHz() const = 0;
    virtual int NumChannels() const = 0;
  
    // Returns the rate at which the RTP timestamps are updated. The default
    // implementation returns SampleRateHz().
    virtual int RtpTimestampRateHz() const;
  
    // Returns the number of 10 ms frames the encoder will put in the next
    // packet. This value may only change when Encode() outputs a packet; i.e.,
    // the encoder may vary the number of 10 ms frames from packet to packet, but
    // it must decide the length of the next packet no later than when outputting
    // the preceding packet.
    virtual size_t Num10MsFramesInNextPacket() const = 0;
  
    // Returns the maximum value that can be returned by
    // Num10MsFramesInNextPacket().
    virtual size_t Max10MsFramesInAPacket() const = 0;
  
    // Returns the current target bitrate in bits/s. The value -1 means that the
    // codec adapts the target automatically, and a current target cannot be
    // provided.
    virtual int GetTargetBitrate() const = 0;
  
    // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
    // NumChannels() samples). Multi-channel audio must be sample-interleaved.
    // The encoder produces zero or more bytes of output in |encoded| and
    // returns additional encoding information.
    // The caller is responsible for making sure that |max_encoded_bytes| is
    // not smaller than the number of bytes actually produced by the encoder.
    // Encode() checks some preconditions, calls EncodeInternal() which does the
    // actual work, and then checks some postconditions.
    EncodedInfo Encode(uint32_t rtp_timestamp,
                       const int16_t* audio,
                       size_t num_samples_per_channel,
                       size_t max_encoded_bytes,
                       uint8_t* encoded);
  
    virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                                       const int16_t* audio,
                                       size_t max_encoded_bytes,
                                       uint8_t* encoded) = 0;
  
    // Resets the encoder to its starting state, discarding any input that has
    // been fed to the encoder but not yet emitted in a packet.
    virtual void Reset() = 0;
  
    // Enables or disables codec-internal FEC (forward error correction). Returns
    // true if the codec was able to comply. The default implementation returns
    // true when asked to disable FEC and false when asked to enable it (meaning
    // that FEC isn't supported).
    virtual bool SetFec(bool enable);
  
    // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
    // able to comply. The default implementation returns true when asked to
    // disable DTX and false when asked to enable it (meaning that DTX isn't
    // supported).
    virtual bool SetDtx(bool enable);
  
    // Sets the application mode. Returns true if the codec was able to comply.
    // The default implementation just returns false.
    enum class Application { kSpeech, kAudio };
    virtual bool SetApplication(Application application);
  
    // Tells the encoder about the highest sample rate the decoder is expected to
    // use when decoding the bitstream. The encoder would typically use this
    // information to adjust the quality of the encoding. The default
    // implementation just returns true.
    virtual void SetMaxPlaybackRate(int frequency_hz);
  
    // Tells the encoder what the projected packet loss rate is. The rate is in
    // the range [0.0, 1.0]. The encoder would typically use this information to
    // adjust channel coding efforts, such as FEC. The default implementation
    // does nothing.
    virtual void SetProjectedPacketLossRate(double fraction);
  
    // Tells the encoder what average bitrate we'd like it to produce. The
    // encoder is free to adjust or disregard the given bitrate (the default
    // implementation does the latter).
    virtual void SetTargetBitrate(int target_bps);
  };
  }  // namespace webrtc
  #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
  