webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.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_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_

#include <vector>

#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"

namespace webrtc {

// NOTE: This class has neither ThreadChecker, nor locks. The owner of an
// AudioEncoderOpus object must ensure that it is not accessed concurrently.

class AudioEncoderOpus final : public AudioEncoder {
 public:
  enum ApplicationMode {
    kVoip = 0,
    kAudio = 1,
  };

  struct Config {
    Config();
    bool IsOk() const;
    int frame_size_ms;
    int num_channels;
    int payload_type;
    ApplicationMode application;
    int bitrate_bps;
    bool fec_enabled;
    int max_playback_rate_hz;
  };

  explicit AudioEncoderOpus(const Config& config);
  virtual ~AudioEncoderOpus() OVERRIDE;

  virtual int sample_rate_hz() const OVERRIDE;
  virtual int num_channels() const OVERRIDE;
  virtual int Num10MsFramesInNextPacket() const OVERRIDE;
  virtual int Max10MsFramesInAPacket() const OVERRIDE;
  void SetTargetBitrate(int bits_per_second) override;
  void SetProjectedPacketLossRate(double fraction) override;

 protected:
  virtual bool EncodeInternal(uint32_t rtp_timestamp,
                              const int16_t* audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded,
                              EncodedInfo* info) OVERRIDE;

 private:
  const int num_10ms_frames_per_packet_;
  const int num_channels_;
  const int payload_type_;
  const ApplicationMode application_;
  const int samples_per_10ms_frame_;
  std::vector<int16_t> input_buffer_;
  OpusEncInst* inst_;
  uint32_t first_timestamp_in_buffer_;
  double packet_loss_rate_;
};

}  // namespace webrtc
#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_