Expose ILBC codec in webrtc/api/audio_codecs/

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.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_ILBC_AUDIO_ENCODER_ILBC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
-class AudioEncoderIlbc final : public AudioEncoder {
+class AudioEncoderIlbcImpl final : public AudioEncoder {
  public:
-  struct Config {
-    bool IsOk() const;
+  static rtc::Optional<AudioEncoderIlbcConfig> SdpToConfig(
+      const SdpAudioFormat& format);
 
-    int payload_type = 102;
-    int frame_size_ms = 30;  // Valid values are 20, 30, 40, and 60 ms.
-    // Note that frame size 40 ms produces encodings with two 20 ms frames in
-    // them, and frame size 60 ms consists of two 30 ms frames.
-  };
-
-  explicit AudioEncoderIlbc(const Config& config);
-  explicit AudioEncoderIlbc(const CodecInst& codec_inst);
-  AudioEncoderIlbc(int payload_type, const SdpAudioFormat& format);
-  ~AudioEncoderIlbc() override;
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
+  explicit AudioEncoderIlbcImpl(const CodecInst& codec_inst);
+  AudioEncoderIlbcImpl(int payload_type, const SdpAudioFormat& format);
+  ~AudioEncoderIlbcImpl() override;
 
   static constexpr const char* GetPayloadName() { return "ILBC"; }
   static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
       const SdpAudioFormat& format);
 
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
   void Reset() override;
 
  private:
   size_t RequiredOutputSizeBytes() const;
 
-  static const size_t kMaxSamplesPerPacket = 480;
-  const Config config_;
+  static constexpr size_t kMaxSamplesPerPacket = 480;
+  const int frame_size_ms_;
+  const int payload_type_;
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   int16_t input_buffer_[kMaxSamplesPerPacket];
   IlbcEncoderInstance* encoder_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbc);
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbcImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_