Remove useless debugging code
It would be enough to say we're removing EnableSrtpDebugging because
it's never called, but the story is a bit more interesting.
libsrtp's debugging facilities are gated behind the reasonably-named
ENABLE_DEBUGGING macro:
https://chromium.googlesource.com/chromium/deps/libsrtp/+/b17c065a8a63725eca216fd461984f82c331dd07/srtp/crypto/include/err.h#186
This code was imported to WebRTC from libjingle, but neither WebRTC or
Chromium ever set ENABLE_DEBUGGING. Even if someone had ever called
EnableSrtpDebugging, it wouldn't have done anything.
BUG=0
Review-Url: https://codereview.webrtc.org/2409513002
Cr-Commit-Position: refs/heads/master@{#14592}
diff --git a/webrtc/pc/srtpfilter.cc b/webrtc/pc/srtpfilter.cc
index 3f38d11..f1ecff1 100644
--- a/webrtc/pc/srtpfilter.cc
+++ b/webrtc/pc/srtpfilter.cc
@@ -25,9 +25,6 @@
#include "webrtc/media/base/rtputils.h"
#include "webrtc/pc/externalhmac.h"
-// Enable this line to turn on SRTP debugging
-// #define SRTP_DEBUG
-
#ifdef HAVE_SRTP
extern "C" {
#ifdef SRTP_RELATIVE_PATH
@@ -38,16 +35,6 @@
#include "third_party/libsrtp/include/srtp_priv.h"
#endif // SRTP_RELATIVE_PATH
}
-
-#if !defined(NDEBUG)
-extern "C" srtp_debug_module_t mod_srtp;
-extern "C" srtp_debug_module_t mod_auth;
-extern "C" srtp_debug_module_t mod_cipher;
-extern "C" srtp_debug_module_t mod_stat;
-extern "C" srtp_debug_module_t mod_alloc;
-extern "C" srtp_debug_module_t mod_aes_icm;
-extern "C" srtp_debug_module_t mod_aes_hmac;
-#endif
#endif // HAVE_SRTP
namespace cricket {
@@ -65,21 +52,6 @@
#endif // !HAVE_SRTP
-void EnableSrtpDebugging() {
-#ifdef HAVE_SRTP
-#if !defined(NDEBUG)
- debug_on(mod_srtp);
- debug_on(mod_auth);
- debug_on(mod_cipher);
- debug_on(mod_stat);
- debug_on(mod_alloc);
- debug_on(mod_aes_icm);
- // debug_on(mod_aes_cbc);
- // debug_on(mod_hmac);
-#endif
-#endif // HAVE_SRTP
-}
-
// NOTE: This is called from ChannelManager D'tor.
void ShutdownSrtp() {
#ifdef HAVE_SRTP
diff --git a/webrtc/pc/srtpfilter.h b/webrtc/pc/srtpfilter.h
index 26e1e24..abea507 100644
--- a/webrtc/pc/srtpfilter.h
+++ b/webrtc/pc/srtpfilter.h
@@ -40,7 +40,6 @@
class SrtpSession;
class SrtpStat;
-void EnableSrtpDebugging();
void ShutdownSrtp();
// Class to transform SRTP to/from RTP.
