modules/audio_coding: Remove some codec include dirs

webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.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_G722_INCLUDE_AUDIO_ENCODER_G722_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_
-
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
-
-namespace webrtc {
-
-struct CodecInst;
-
-class AudioEncoderG722 final : public AudioEncoder {
- public:
-  struct Config {
-    bool IsOk() const;
-
-    int payload_type = 9;
-    int frame_size_ms = 20;
-    int num_channels = 1;
-  };
-
-  explicit AudioEncoderG722(const Config& config);
-  explicit AudioEncoderG722(const CodecInst& codec_inst);
-  ~AudioEncoderG722() override;
-
-  size_t MaxEncodedBytes() const override;
-  int SampleRateHz() const override;
-  int NumChannels() const override;
-  int RtpTimestampRateHz() const override;
-  size_t Num10MsFramesInNextPacket() const override;
-  size_t Max10MsFramesInAPacket() const override;
-  int GetTargetBitrate() const override;
-  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             rtc::ArrayView<const int16_t> audio,
-                             size_t max_encoded_bytes,
-                             uint8_t* encoded) override;
-  void Reset() override;
-
- private:
-  // The encoder state for one channel.
-  struct EncoderState {
-    G722EncInst* encoder;
-    rtc::scoped_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
-    rtc::Buffer encoded_buffer;                 // Already encoded.
-    EncoderState();
-    ~EncoderState();
-  };
-
-  size_t SamplesPerChannel() const;
-
-  const int 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 rtc::scoped_ptr<EncoderState[]> encoders_;
-  rtc::Buffer interleave_buffer_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_