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

webrtc/call/video_config.h

Index: webrtc/call/video_config.h
diff --git a/webrtc/config.h b/webrtc/call/video_config.h
similarity index 56%
copy from webrtc/config.h
copy to webrtc/call/video_config.h
index 962e0f2fb9ac844ff243c0425c1b9df4dca76a4d..cdff513263742ad2cecdcc8c20852e757183c87d 100644
--- a/webrtc/config.h
+++ b/webrtc/call/video_config.h
@@ -8,10 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// TODO(pbos): Move Config from common.h to here.
-
-#ifndef WEBRTC_CONFIG_H_
-#define WEBRTC_CONFIG_H_
+#ifndef WEBRTC_CALL_VIDEO_CONFIG_H_
+#define WEBRTC_CALL_VIDEO_CONFIG_H_
 
 #include <string>
 #include <vector>
@@ -25,114 +23,6 @@
 
 namespace webrtc {
 
-// Settings for NACK, see RFC 4585 for details.
-struct NackConfig {
-  NackConfig() : rtp_history_ms(0) {}
-  std::string ToString() const;
-  // Send side: the time RTP packets are stored for retransmissions.
-  // Receive side: the time the receiver is prepared to wait for
-  // retransmissions.
-  // Set to '0' to disable.
-  int rtp_history_ms;
-};
-
-// Settings for ULPFEC forward error correction.
-// Set the payload types to '-1' to disable.
-struct UlpfecConfig {
-  UlpfecConfig()
-      : ulpfec_payload_type(-1),
-        red_payload_type(-1),
-        red_rtx_payload_type(-1) {}
-  std::string ToString() const;
-  bool operator==(const UlpfecConfig& other) const;
-
-  // 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();
@@ -177,6 +67,7 @@ class VideoEncoderConfig {
     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;
@@ -263,4 +154,4 @@ class VideoEncoderConfig {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_CONFIG_H_
+#endif  // WEBRTC_CALL_VIDEO_CONFIG_H_