audio_coding: rename interface -> include

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/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-
-namespace webrtc {
-
-struct CodecInst;
-
-class AudioEncoderOpus final : public AudioEncoder {
- public:
-  enum ApplicationMode {
-    kVoip = 0,
-    kAudio = 1,
-  };
-
-  struct Config {
-    bool IsOk() const;
-    int frame_size_ms = 20;
-    int num_channels = 1;
-    int payload_type = 120;
-    ApplicationMode application = kVoip;
-    int bitrate_bps = 64000;
-    bool fec_enabled = false;
-    int max_playback_rate_hz = 48000;
-    int complexity = kDefaultComplexity;
-    bool dtx_enabled = false;
-
-   private:
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
-    // If we are on Android, iOS and/or ARM, use a lower complexity setting as
-    // default, to save encoder complexity.
-    static const int kDefaultComplexity = 5;
-#else
-    static const int kDefaultComplexity = 9;
-#endif
-  };
-
-  explicit AudioEncoderOpus(const Config& config);
-  explicit AudioEncoderOpus(const CodecInst& codec_inst);
-  ~AudioEncoderOpus() override;
-
-  size_t MaxEncodedBytes() const override;
-  int SampleRateHz() const override;
-  int NumChannels() const override;
-  size_t Num10MsFramesInNextPacket() const override;
-  size_t Max10MsFramesInAPacket() const override;
-  int GetTargetBitrate() const override;
-
-  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             const int16_t* audio,
-                             size_t max_encoded_bytes,
-                             uint8_t* encoded) override;
-
-  void Reset() override;
-  bool SetFec(bool enable) override;
-
-  // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
-  // being inactive. During that, it still sends 2 packets (one for content, one
-  // for signaling) about every 400 ms.
-  bool SetDtx(bool enable) override;
-
-  bool SetApplication(Application application) override;
-  void SetMaxPlaybackRate(int frequency_hz) override;
-  void SetProjectedPacketLossRate(double fraction) override;
-  void SetTargetBitrate(int target_bps) override;
-
-  // Getters for testing.
-  double packet_loss_rate() const { return packet_loss_rate_; }
-  ApplicationMode application() const { return config_.application; }
-  bool dtx_enabled() const { return config_.dtx_enabled; }
-
- private:
-  int Num10msFramesPerPacket() const;
-  int SamplesPer10msFrame() const;
-  bool RecreateEncoderInstance(const Config& config);
-
-  Config config_;
-  double packet_loss_rate_;
-  std::vector<int16_t> input_buffer_;
-  OpusEncInst* inst_;
-  uint32_t first_timestamp_in_buffer_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_