Move RtpExtension to api/ directory and config.h/.cc to call/.

webrtc/call/config.h

 /*
  *  Copyright (c) 2013 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.
  */
 
 // TODO(pbos): Move Config from common.h to here.

[the sun, 2017/08/28 18:52:56]

obsolete comment

[stefan-webrtc, 2017/08/29 08:37:17]

I don't know if it's obsolete to be honest, not clear what Config it refers to.
:)

I'll delete it since it's hard to understand anyway.

[the sun, 2017/08/28 18:52:56]

Add another comment here, or file an issue, that the contents of this file
should be split out.

[stefan-webrtc, 2017/08/29 08:37:17]

I had to split them already because of dependency issues.

-#ifndef WEBRTC_CONFIG_H_
-#define WEBRTC_CONFIG_H_
+#ifndef WEBRTC_CALL_CONFIG_H_
+#define WEBRTC_CALL_CONFIG_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/rtc_base/basictypes.h"
 #include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/refcount.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/typedefs.h"
@@ skipping 24 matching lines @@
   // Payload type used for ULPFEC packets.
   int ulpfec_payload_type;
 
   // Payload type used for RED packets.
   int red_payload_type;
 
   // RTX payload type for RED payload.
   int red_rtx_payload_type;
 };
 
-// RTP header extension, see RFC 5285.
-struct RtpExtension {
-  RtpExtension() {}
-  RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
-  RtpExtension(const std::string& uri, int id, bool encrypt) : uri(uri),
-      id(id), encrypt(encrypt) {}
-  std::string ToString() const;
-  bool operator==(const RtpExtension& rhs) const {
-    return uri == rhs.uri && id == rhs.id && encrypt == rhs.encrypt;
-  }
-  static bool IsSupportedForAudio(const std::string& uri);
-  static bool IsSupportedForVideo(const std::string& uri);
-  // Return "true" if the given RTP header extension URI may be encrypted.
-  static bool IsEncryptionSupported(const std::string& uri);
-
-  // Returns the named header extension if found among all extensions,
-  // nullptr otherwise.
-  static const RtpExtension* FindHeaderExtensionByUri(
-    const std::vector<RtpExtension>& extensions,
-    const std::string& uri);
-
-  // Return a list of RTP header extensions with the non-encrypted extensions
-  // removed if both the encrypted and non-encrypted extension is present for
-  // the same URI.
-  static std::vector<RtpExtension> FilterDuplicateNonEncrypted(
-    const std::vector<RtpExtension>& extensions);
-
-  // Header extension for audio levels, as defined in:
-  // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
-  static const char kAudioLevelUri[];
-  static const int kAudioLevelDefaultId;
-
-  // Header extension for RTP timestamp offset, see RFC 5450 for details:
-  // http://tools.ietf.org/html/rfc5450
-  static const char kTimestampOffsetUri[];
-  static const int kTimestampOffsetDefaultId;
-
-  // Header extension for absolute send time, see url for details:
-  // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-  static const char kAbsSendTimeUri[];
-  static const int kAbsSendTimeDefaultId;
-
-  // Header extension for coordination of video orientation, see url for
-  // details:
-  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf
-  static const char kVideoRotationUri[];
-  static const int kVideoRotationDefaultId;
-
-  // Header extension for video content type. E.g. default or screenshare.
-  static const char kVideoContentTypeUri[];
-  static const int kVideoContentTypeDefaultId;
-
-  // Header extension for video timing.
-  static const char kVideoTimingUri[];
-  static const int kVideoTimingDefaultId;
-
-  // Header extension for transport sequence number, see url for details:
-  // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
-  static const char kTransportSequenceNumberUri[];
-  static const int kTransportSequenceNumberDefaultId;
-
-  static const char kPlayoutDelayUri[];
-  static const int kPlayoutDelayDefaultId;
-
-  // Encryption of Header Extensions, see RFC 6904 for details:
-  // https://tools.ietf.org/html/rfc6904
-  static const char kEncryptHeaderExtensionsUri[];
-
-  // Inclusive min and max IDs for one-byte header extensions, per RFC5285.
-  static const int kMinId;
-  static const int kMaxId;
-
-  std::string uri;
-  int id = 0;
-  bool encrypt = false;
-};
-
 struct VideoStream {
   VideoStream();
   ~VideoStream();
   std::string ToString() const;
 
   size_t width;
   size_t height;
   int max_framerate;
 
   int min_bitrate_bps;
@@ skipping 24 matching lines @@
   class EncoderSpecificSettings : public rtc::RefCountInterface {
    public:
     // TODO(pbos): Remove FillEncoderSpecificSettings as soon as VideoCodec is
     // not in use and encoder implementations ask for codec-specific structs
     // directly.
     void FillEncoderSpecificSettings(VideoCodec* codec_struct) const;
 
     virtual void FillVideoCodecVp8(VideoCodecVP8* vp8_settings) const;
     virtual void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const;
     virtual void FillVideoCodecH264(VideoCodecH264* h264_settings) const;
+
    private:
     ~EncoderSpecificSettings() override {}
     friend class VideoEncoderConfig;
   };
 
   class H264EncoderSpecificSettings : public EncoderSpecificSettings {
    public:
     explicit H264EncoderSpecificSettings(const VideoCodecH264& specifics);
     void FillVideoCodecH264(VideoCodecH264* h264_settings) const override;
 
@@ skipping 66 matching lines @@
   size_t number_of_streams;
 
  private:
   // Access to the copy constructor is private to force use of the Copy()
   // method for those exceptional cases where we do use it.
   VideoEncoderConfig(const VideoEncoderConfig&);
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_CONFIG_H_
+#endif  // WEBRTC_CALL_CONFIG_H_