audio_encoder_g722.h

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/*

 *  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 MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_

#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_

#include <memory>

#include <utility>

#include "absl/types/optional.h"

#include "api/audio_codecs/audio_encoder.h"

#include "api/audio_codecs/g722/audio_encoder_g722_config.h"

#include "api/units/time_delta.h"

#include "modules/audio_coding/codecs/g722/g722_interface.h"

#include "rtc_base/buffer.h"

namespace webrtc {

class AudioEncoderG722Impl final : public AudioEncoder {

 public:

  AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);

  ~AudioEncoderG722Impl() override;

  AudioEncoderG722Impl(const AudioEncoderG722Impl&) = delete;

  AudioEncoderG722Impl& operator=(const AudioEncoderG722Impl&) = delete;

  int SampleRateHz() const override;

  size_t NumChannels() const override;

  int RtpTimestampRateHz() const override;

  size_t Num10MsFramesInNextPacket() const override;

  size_t Max10MsFramesInAPacket() const override;

  int GetTargetBitrate() const override;

  void Reset() override;

  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()

      const override;

 protected:

  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,

                         rtc::ArrayView<const int16_t> audio,

                         rtc::Buffer* encoded) override;

 private:

  // The encoder state for one channel.

  struct EncoderState {

    G722EncInst* encoder;

    std::unique_ptr<int16_t[]> speech_buffer;  // Queued up for encoding.

    rtc::Buffer encoded_buffer;                // Already encoded.

    EncoderState();

    ~EncoderState();

};

  size_t SamplesPerChannel() const;

  const size_t num_channels_;

  const int payload_type_;

  const size_t num_10ms_frames_per_packet_;

  size_t num_10ms_frames_buffered_;

  uint32_t first_timestamp_in_buffer_;

  const std::unique_ptr<EncoderState[]> encoders_;

  rtc::Buffer interleave_buffer_;

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_