commit d8d3248d95437863f3448affad7a1bc2238a1959
author Elad Alon <eladalon@webrtc.org> Mon Feb 18 22:45:57 2019
committer Commit Bot <commit-bot@chromium.org> Tue Feb 19 08:51:20 2019
tree 3ab1dc5fcaff70ea651489e5ad2622186f777f26
parent 1925b5a1d3083c12dfd0c9467351f352784cd149

Reland "Delete test/constants.h"

This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6.

Reason for revert: Failing tests fixed.

Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}

TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}

test/call_test.h

diff --git a/test/call_test.h b/test/call_test.h
index fb26051..dbe8e07 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/test/video/function_video_decoder_factory.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "api/video/video_bitrate_allocator_factory.h"
@@ -71,6 +72,8 @@
   static const std::map<uint8_t, MediaType> payload_type_map_;
 
  protected:
+  void RegisterRtpExtension(const RtpExtension& extension);
+
   // RunBaseTest overwrites the audio_state of the send and receive Call configs
   // to simplify test code.
   void RunBaseTest(BaseTest* test);
@@ -216,6 +219,13 @@
   SingleThreadedTaskQueueForTesting task_queue_;
 
  private:
+  absl::optional<RtpExtension> GetRtpExtensionByUri(
+      const std::string& uri) const;
+
+  void AddRtpExtensionByUri(const std::string& uri,
+                            std::vector<RtpExtension>* extensions) const;
+
+  std::vector<RtpExtension> rtp_extensions_;
   rtc::scoped_refptr<AudioProcessing> apm_send_;
   rtc::scoped_refptr<AudioProcessing> apm_recv_;
   rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;