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      1 /*
      2  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 
     11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
     12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
     13 
     14 #include <algorithm>
     15 #include <vector>
     16 
     17 #include "webrtc/base/array_view.h"
     18 #include "webrtc/typedefs.h"
     19 
     20 namespace webrtc {
     21 
     22 // This is the interface class for encoders in AudioCoding module. Each codec
     23 // type must have an implementation of this class.
     24 class AudioEncoder {
     25  public:
     26   struct EncodedInfoLeaf {
     27     size_t encoded_bytes = 0;
     28     uint32_t encoded_timestamp = 0;
     29     int payload_type = 0;
     30     bool send_even_if_empty = false;
     31     bool speech = true;
     32   };
     33 
     34   // This is the main struct for auxiliary encoding information. Each encoded
     35   // packet should be accompanied by one EncodedInfo struct, containing the
     36   // total number of |encoded_bytes|, the |encoded_timestamp| and the
     37   // |payload_type|. If the packet contains redundant encodings, the |redundant|
     38   // vector will be populated with EncodedInfoLeaf structs. Each struct in the
     39   // vector represents one encoding; the order of structs in the vector is the
     40   // same as the order in which the actual payloads are written to the byte
     41   // stream. When EncoderInfoLeaf structs are present in the vector, the main
     42   // struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
     43   // vector.
     44   struct EncodedInfo : public EncodedInfoLeaf {
     45     EncodedInfo();
     46     ~EncodedInfo();
     47 
     48     std::vector<EncodedInfoLeaf> redundant;
     49   };
     50 
     51   virtual ~AudioEncoder() = default;
     52 
     53   // Returns the maximum number of bytes that can be produced by the encoder
     54   // at each Encode() call. The caller can use the return value to determine
     55   // the size of the buffer that needs to be allocated. This value is allowed
     56   // to depend on encoder parameters like bitrate, frame size etc., so if
     57   // any of these change, the caller of Encode() is responsible for checking
     58   // that the buffer is large enough by calling MaxEncodedBytes() again.
     59   virtual size_t MaxEncodedBytes() const = 0;
     60 
     61   // Returns the input sample rate in Hz and the number of input channels.
     62   // These are constants set at instantiation time.
     63   virtual int SampleRateHz() const = 0;
     64   virtual size_t NumChannels() const = 0;
     65 
     66   // Returns the rate at which the RTP timestamps are updated. The default
     67   // implementation returns SampleRateHz().
     68   virtual int RtpTimestampRateHz() const;
     69 
     70   // Returns the number of 10 ms frames the encoder will put in the next
     71   // packet. This value may only change when Encode() outputs a packet; i.e.,
     72   // the encoder may vary the number of 10 ms frames from packet to packet, but
     73   // it must decide the length of the next packet no later than when outputting
     74   // the preceding packet.
     75   virtual size_t Num10MsFramesInNextPacket() const = 0;
     76 
     77   // Returns the maximum value that can be returned by
     78   // Num10MsFramesInNextPacket().
     79   virtual size_t Max10MsFramesInAPacket() const = 0;
     80 
     81   // Returns the current target bitrate in bits/s. The value -1 means that the
     82   // codec adapts the target automatically, and a current target cannot be
     83   // provided.
     84   virtual int GetTargetBitrate() const = 0;
     85 
     86   // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
     87   // NumChannels() samples). Multi-channel audio must be sample-interleaved.
     88   // The encoder produces zero or more bytes of output in |encoded| and
     89   // returns additional encoding information.
     90   // The caller is responsible for making sure that |max_encoded_bytes| is
     91   // not smaller than the number of bytes actually produced by the encoder.
     92   // Encode() checks some preconditions, calls EncodeInternal() which does the
     93   // actual work, and then checks some postconditions.
     94   EncodedInfo Encode(uint32_t rtp_timestamp,
     95                      rtc::ArrayView<const int16_t> audio,
     96                      size_t max_encoded_bytes,
     97                      uint8_t* encoded);
     98 
     99   virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
    100                                      rtc::ArrayView<const int16_t> audio,
    101                                      size_t max_encoded_bytes,
    102                                      uint8_t* encoded) = 0;
    103 
    104   // Resets the encoder to its starting state, discarding any input that has
    105   // been fed to the encoder but not yet emitted in a packet.
    106   virtual void Reset() = 0;
    107 
    108   // Enables or disables codec-internal FEC (forward error correction). Returns
    109   // true if the codec was able to comply. The default implementation returns
    110   // true when asked to disable FEC and false when asked to enable it (meaning
    111   // that FEC isn't supported).
    112   virtual bool SetFec(bool enable);
    113 
    114   // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
    115   // able to comply. The default implementation returns true when asked to
    116   // disable DTX and false when asked to enable it (meaning that DTX isn't
    117   // supported).
    118   virtual bool SetDtx(bool enable);
    119 
    120   // Sets the application mode. Returns true if the codec was able to comply.
    121   // The default implementation just returns false.
    122   enum class Application { kSpeech, kAudio };
    123   virtual bool SetApplication(Application application);
    124 
    125   // Tells the encoder about the highest sample rate the decoder is expected to
    126   // use when decoding the bitstream. The encoder would typically use this
    127   // information to adjust the quality of the encoding. The default
    128   // implementation does nothing.
    129   virtual void SetMaxPlaybackRate(int frequency_hz);
    130 
    131   // Tells the encoder what the projected packet loss rate is. The rate is in
    132   // the range [0.0, 1.0]. The encoder would typically use this information to
    133   // adjust channel coding efforts, such as FEC. The default implementation
    134   // does nothing.
    135   virtual void SetProjectedPacketLossRate(double fraction);
    136 
    137   // Tells the encoder what average bitrate we'd like it to produce. The
    138   // encoder is free to adjust or disregard the given bitrate (the default
    139   // implementation does the latter).
    140   virtual void SetTargetBitrate(int target_bps);
    141 };
    142 }  // namespace webrtc
    143 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
    144