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

webrtc/call/config.cc

 /*
  *  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.
  */
-#include "webrtc/config.h"
+#include "webrtc/call/config.h"
 
 #include <algorithm>
 #include <sstream>
 #include <string>
 
 #include "webrtc/rtc_base/checks.h"
 
 namespace webrtc {
 std::string NackConfig::ToString() const {
   std::stringstream ss;
@@ skipping 10 matching lines @@
   ss << '}';
   return ss.str();
 }
 
 bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
   return ulpfec_payload_type == other.ulpfec_payload_type &&
          red_payload_type == other.red_payload_type &&
          red_rtx_payload_type == other.red_rtx_payload_type;
 }
 
-std::string RtpExtension::ToString() const {
-  std::stringstream ss;
-  ss << "{uri: " << uri;
-  ss << ", id: " << id;
-  if (encrypt) {
-    ss << ", encrypt";
-  }
-  ss << '}';
-  return ss.str();
-}
-
-const char RtpExtension::kAudioLevelUri[] =
-    "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
-const int RtpExtension::kAudioLevelDefaultId = 1;
-
-const char RtpExtension::kTimestampOffsetUri[] =
-    "urn:ietf:params:rtp-hdrext:toffset";
-const int RtpExtension::kTimestampOffsetDefaultId = 2;
-
-const char RtpExtension::kAbsSendTimeUri[] =
-    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
-const int RtpExtension::kAbsSendTimeDefaultId = 3;
-
-const char RtpExtension::kVideoRotationUri[] = "urn:3gpp:video-orientation";
-const int RtpExtension::kVideoRotationDefaultId = 4;
-
-const char RtpExtension::kTransportSequenceNumberUri[] =
-    "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
-const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
-
-// This extension allows applications to adaptively limit the playout delay
-// on frames as per the current needs. For example, a gaming application
-// has very different needs on end-to-end delay compared to a video-conference
-// application.
-const char RtpExtension::kPlayoutDelayUri[] =
-    "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
-const int RtpExtension::kPlayoutDelayDefaultId = 6;
-
-const char RtpExtension::kVideoContentTypeUri[] =
-    "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
-const int RtpExtension::kVideoContentTypeDefaultId = 7;
-
-const char RtpExtension::kVideoTimingUri[] =
-    "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
-const int RtpExtension::kVideoTimingDefaultId = 8;
-
-const char RtpExtension::kEncryptHeaderExtensionsUri[] =
-    "urn:ietf:params:rtp-hdrext:encrypt";
-
-const int RtpExtension::kMinId = 1;
-const int RtpExtension::kMaxId = 14;
-
-bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
-  return uri == webrtc::RtpExtension::kAudioLevelUri ||
-         uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
-}
-
-bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
-  return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
-         uri == webrtc::RtpExtension::kAbsSendTimeUri ||
-         uri == webrtc::RtpExtension::kVideoRotationUri ||
-         uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
-         uri == webrtc::RtpExtension::kPlayoutDelayUri ||
-         uri == webrtc::RtpExtension::kVideoContentTypeUri ||
-         uri == webrtc::RtpExtension::kVideoTimingUri;
-}
-
-bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
-  return uri == webrtc::RtpExtension::kAudioLevelUri ||
-         uri == webrtc::RtpExtension::kTimestampOffsetUri ||
-#if !defined(ENABLE_EXTERNAL_AUTH)
-         // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri"
-         // here and filter out later if external auth is really used in
-         // srtpfilter. External auth is used by Chromium and replaces the
-         // extension header value of "kAbsSendTimeUri", so it must not be
-         // encrypted (which can't be done by Chromium).
-         uri == webrtc::RtpExtension::kAbsSendTimeUri ||
-#endif
-         uri == webrtc::RtpExtension::kVideoRotationUri ||
-         uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
-         uri == webrtc::RtpExtension::kPlayoutDelayUri ||
-         uri == webrtc::RtpExtension::kVideoContentTypeUri;
-}
-
-const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
-    const std::vector<RtpExtension>& extensions,
-    const std::string& uri) {
-  for (const auto& extension : extensions) {
-    if (extension.uri == uri) {
-      return &extension;
-    }
-  }
-  return nullptr;
-}
-
-std::vector<RtpExtension> RtpExtension::FilterDuplicateNonEncrypted(
-    const std::vector<RtpExtension>& extensions) {
-  std::vector<RtpExtension> filtered;
-  for (auto extension = extensions.begin(); extension != extensions.end();
-      ++extension) {
-    if (extension->encrypt) {
-      filtered.push_back(*extension);
-      continue;
-    }
-
-    // Only add non-encrypted extension if no encrypted with the same URI
-    // is also present...
-    if (std::find_if(extension + 1, extensions.end(),
-        [extension](const RtpExtension& check) {
-          return extension->uri == check.uri;
-        }) != extensions.end()) {
-      continue;
-    }
-
-    // ...and has not been added before.
-    if (!FindHeaderExtensionByUri(filtered, extension->uri)) {
-      filtered.push_back(*extension);
-    }
-  }
-  return filtered;
-}
-
 VideoStream::VideoStream()
     : width(0),
       height(0),
       max_framerate(-1),
       min_bitrate_bps(-1),
       target_bitrate_bps(-1),
       max_bitrate_bps(-1),
       max_qp(-1) {}
 
 VideoStream::~VideoStream() = default;
@@ skipping 101 matching lines @@
 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings(
     const VideoCodecVP9& specifics)
     : specifics_(specifics) {}
 
 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
     VideoCodecVP9* vp9_settings) const {
   *vp9_settings = specifics_;
 }
 
 }  // namespace webrtc