/*
 *  Copyright (c) 2017 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 API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_

#include <memory>
#include <vector>

#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"

namespace webrtc {

// L16 encoder API for use as a template parameter to
// CreateAudioEncoderFactory<...>().
struct RTC_EXPORT AudioEncoderL16 {
  struct Config {
    bool IsOk() const {
      return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
              sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
             num_channels >= 1 &&
             num_channels <= AudioEncoder::kMaxNumberOfChannels &&
             frame_size_ms > 0 && frame_size_ms <= 120 &&
             frame_size_ms % 10 == 0;
    }
    int sample_rate_hz = 8000;
    int num_channels = 1;
    int frame_size_ms = 10;
  };
  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
  static AudioCodecInfo QueryAudioEncoder(const Config& config);
  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
      const Config& config,
      int payload_type,
      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
      const FieldTrialsView* field_trials = nullptr);
};

}  // namespace webrtc

#endif  // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
