webrtc/modules/audio_coding/codecs/audio_encoder.h

/*
 *  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/base/array_view.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/deprecation.h"
#include "webrtc/base/optional.h"
#include "webrtc/typedefs.h"

namespace webrtc {

class Clock;
class RtcEventLog;

// This is the interface class for encoders in AudioCoding module. Each codec
// type must have an implementation of this class.
class AudioEncoder {
 public:
  // Used for UMA logging of codec usage. The same codecs, with the
  // same values, must be listed in
  // src/tools/metrics/histograms/histograms.xml in chromium to log
  // correct values.
  enum class CodecType {
    kOther = 0,  // Codec not specified, and/or not listed in this enum
    kOpus = 1,
    kIsac = 2,
    kPcmA = 3,
    kPcmU = 4,
    kG722 = 5,
    kIlbc = 6,

    // Number of histogram bins in the UMA logging of codec types. The
    // total number of different codecs that are logged cannot exceed this
    // number.
    kMaxLoggedAudioCodecTypes
  };

  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;
    CodecType encoder_type = CodecType::kOther;
  };

  // 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(const EncodedInfo&);
    EncodedInfo(EncodedInfo&&);
    ~EncodedInfo();
    EncodedInfo& operator=(const EncodedInfo&);
    EncodedInfo& operator=(EncodedInfo&&);

    std::vector<EncodedInfoLeaf> redundant;
  };

  virtual ~AudioEncoder() = default;

  // 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 size_t 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 appends zero or more bytes of output to |encoded| and returns
  // additional encoding information.  Encode() checks some preconditions, calls
  // EncodeImpl() which does the actual work, and then checks some
  // postconditions.
  EncodedInfo Encode(uint32_t rtp_timestamp,
                     rtc::ArrayView<const int16_t> audio,
                     rtc::Buffer* encoded);

  // 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);

  // Returns the status of codec-internal DTX. The default implementation always
  // returns false.
  virtual bool GetDtx() const;

  // 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 does nothing.
  virtual void SetMaxPlaybackRate(int frequency_hz);

  // This is to be deprecated. Please use |OnReceivedTargetAudioBitrate|
  // instead.
  // 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).
  RTC_DEPRECATED virtual void SetTargetBitrate(int target_bps);

  // Causes this encoder to let go of any other encoders it contains, and
  // returns a pointer to an array where they are stored (which is required to
  // live as long as this encoder). Unless the returned array is empty, you may
  // not call any methods on this encoder afterwards, except for the
  // destructor. The default implementation just returns an empty array.
  // NOTE: This method is subject to change. Do not call or override it.
  virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
  ReclaimContainedEncoders();

  // Enables audio network adaptor. Returns true if successful.
  virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
                                         RtcEventLog* event_log,
                                         const Clock* clock);

  // Disables audio network adaptor.
  virtual void DisableAudioNetworkAdaptor();

  // Provides uplink packet loss fraction to this encoder to allow it to adapt.
  // |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
  virtual void OnReceivedUplinkPacketLossFraction(
      float uplink_packet_loss_fraction);

  // Provides 1st-order-FEC-recoverable uplink packet loss rate to this encoder
  // to allow it to adapt.
  // |uplink_recoverable_packet_loss_fraction| is in the range [0.0, 1.0].
  virtual void OnReceivedUplinkRecoverablePacketLossFraction(
      float uplink_recoverable_packet_loss_fraction);

  // Provides target audio bitrate to this encoder to allow it to adapt.
  virtual void OnReceivedTargetAudioBitrate(int target_bps);

  // Provides target audio bitrate and corresponding probing interval of
  // the bandwidth estimator to this encoder to allow it to adapt.
  virtual void OnReceivedUplinkBandwidth(
      int target_audio_bitrate_bps,
      rtc::Optional<int64_t> probing_interval_ms);

  // Provides RTT to this encoder to allow it to adapt.
  virtual void OnReceivedRtt(int rtt_ms);

  // Provides overhead to this encoder to adapt. The overhead is the number of
  // bytes that will be added to each packet the encoder generates.
  virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);

  // To allow encoder to adapt its frame length, it must be provided the frame
  // length range that receivers can accept.
  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                           int max_frame_length_ms);

 protected:
  // Subclasses implement this to perform the actual encoding. Called by
  // Encode().
  virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                 rtc::ArrayView<const int16_t> audio,
                                 rtc::Buffer* encoded) = 0;
};
}  // namespace webrtc
#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_