initial_build_fix.patch

Index: source/third_party/libsrtp/libsrtp.gyp
===================================================================
--- source.orig/third_party/libsrtp/libsrtp.gyp
+++ source/third_party/libsrtp/libsrtp.gyp
@@ -188,7 +188,7 @@
       'conditions': [
         ['use_openssl==1', {
           'dependencies': [
-            '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl',
+#            '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl',
           ],
           'sources!': [
             'srtp/crypto/cipher/aes_cbc.c',
@@ -296,7 +296,7 @@
       'conditions': [
         ['use_openssl==1', {
           'dependencies': [
-            '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl',
+#            '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl',
           ],
         }],
       ],
Index: source/webrtc/base/json.h
===================================================================
--- source.orig/webrtc/base/json.h
+++ source/webrtc/base/json.h
@@ -17,7 +17,7 @@
 #if !defined(WEBRTC_EXTERNAL_JSON)
 #include "json/json.h"
 #else
-#include "third_party/jsoncpp/json.h"
+#include "json/json.h"
 #endif
 
 namespace rtc {
Index: source/webrtc/base/opensslstreamadapter.cc
===================================================================
--- source.orig/webrtc/base/opensslstreamadapter.cc
+++ source/webrtc/base/opensslstreamadapter.cc
@@ -19,9 +19,14 @@
 #include <openssl/tls1.h>
 #include <openssl/x509v3.h>
 #ifndef OPENSSL_IS_BORINGSSL
+#include <openssl/ssl.h>
 #include <openssl/dtls1.h>
 #endif
 
+#ifndef DTLS1_2_VERSION
+#define DTLS1_2_VERSION 0xFEFD
+#endif
+
 #include <memory>
 #include <vector>
 
Index: source/webrtc/base/task_queue_libevent.cc
===================================================================
--- source.orig/webrtc/base/task_queue_libevent.cc
+++ source/webrtc/base/task_queue_libevent.cc
@@ -14,7 +14,7 @@
 #include <string.h>
 #include <unistd.h>
 
-#include "base/third_party/libevent/event.h"
+#include <event.h>
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue_posix.h"
Index: source/webrtc/media/sctp/sctpdataengine.cc
===================================================================
--- source.orig/webrtc/media/sctp/sctpdataengine.cc
+++ source/webrtc/media/sctp/sctpdataengine.cc
@@ -17,7 +17,8 @@
 #include <sstream>
 #include <vector>
 
-#include "usrsctplib/usrsctp.h"
+#include <usrsctp.h>
+
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/criticalsection.h"
Index: source/webrtc/modules/audio_coding/neteq/neteq.gypi
===================================================================
--- source.orig/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ source/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -132,6 +132,14 @@
         'time_stretch.cc',
         'time_stretch.h',
       ],
+      'conditions': [
+        ['include_opus==0', {
+          'sources!': [
+            'audio_classifier.cc',
+            'audio_classifier.h',
+          ],
+        }],
+      ],
     },
   ], # targets
   'conditions': [
Index: source/webrtc/modules/audio_device/audio_device.gypi
===================================================================
--- source.orig/webrtc/modules/audio_device/audio_device.gypi
+++ source/webrtc/modules/audio_device/audio_device.gypi
@@ -113,6 +113,13 @@
                 ],
               },
             }],
+            ['use_x11==1', {
+              'link_settings': {
+                'libraries': [
+                  '-lX11',
+                ],
+              },
+            }],
             ['OS=="linux"', {
               'sources': [
                 'linux/alsasymboltable_linux.cc',
@@ -129,7 +136,7 @@
               ],
               'link_settings': {
                 'libraries': [
-                  '-ldl','-lX11',
+                  '-ldl',
                 ],
               },
               'conditions': [
Index: source/webrtc/api/api.gyp
===================================================================
--- source.orig/webrtc/api/api.gyp
+++ source/webrtc/api/api.gyp
@@ -117,7 +117,7 @@
 
             ],
           },
-          'includes': ['../../build/java.gypi'],
+#          'includes': ['../../build/java.gypi'],
         }, # libjingle_peerconnection_java
       ]
     }],
Index: source/webrtc/api/api_tests.gyp
===================================================================
--- source.orig/webrtc/api/api_tests.gyp
+++ source/webrtc/api/api_tests.gyp
@@ -147,8 +147,8 @@
             'never_lint': 1,
           },
           'includes': [
-            '../../build/java_apk.gypi',
-            '../../build/android/test_runner.gypi',
+#            '../../build/java_apk.gypi',
+#            '../../build/android/test_runner.gypi',
           ],
         },
       ],  # targets
Index: source/webrtc/base/testutils.h
===================================================================
--- source.orig/webrtc/base/testutils.h
+++ source/webrtc/base/testutils.h
@@ -14,6 +14,8 @@
 // Utilities for testing rtc infrastructure in unittests
 
 #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
+
+#if defined(USE_X11)
 #include <X11/Xlib.h>
 #include <X11/extensions/Xrandr.h>
 
@@ -21,6 +23,7 @@
 #undef None
 #undef Bool
 #endif
+#endif
 
 #include <algorithm>
 #include <map>
@@ -598,7 +601,7 @@ inline AssertionResult CmpHelperFileEq(c
     return; \
   } \
 
-#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID) && defined(USE_X11)
 struct XDisplay {
   XDisplay() : display_(XOpenDisplay(NULL)) { }
   ~XDisplay() { if (display_) XCloseDisplay(display_); }
@@ -612,7 +615,7 @@ struct XDisplay {
 // Returns true if screencasting is available. When false, anything that uses
 // screencasting features may fail.
 inline bool IsScreencastingAvailable() {
-#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID) && defined(USE_X11)
   XDisplay display;
   if (!display.IsValid()) {
     LOG(LS_WARNING) << "No X Display available.";
Index: source/webrtc/test/test.gyp
===================================================================
--- source.orig/webrtc/test/test.gyp
+++ source/webrtc/test/test.gyp
@@ -328,7 +328,16 @@
        'win/d3d_renderer.h',
      ],
      'conditions': [
-       ['OS=="linux"', {
+       ['OS=="linux" and use_x11==0', {
+         'sources!': [
+           'gl/gl_renderer.cc',
+           'gl/gl_renderer.h',
+           'linux/glx_renderer.cc',
+           'linux/glx_renderer.h',
+           'linux/video_renderer_linux.cc',
+         ],
+       }],
+       ['OS=="linux" and use_x11==1', {
          'sources!': [
            'null_platform_renderer.cc',
          ],
@@ -374,7 +383,7 @@
      ],
      'direct_dependent_settings': {
        'conditions': [
-         ['OS=="linux"', {
+         ['OS=="linux" and use_x11==1', {
            'libraries': [
              '-lXext',
              '-lX11',
Index: source/webrtc/modules/audio_coding/audio_coding_tests.gypi
===================================================================
--- source.orig/webrtc/modules/audio_coding/audio_coding_tests.gypi
+++ source/webrtc/modules/audio_coding/audio_coding_tests.gypi
@@ -19,7 +19,6 @@
       'dependencies': [
         'audio_processing',
         'isac_fix',
-        'webrtc_opus',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
         '<(webrtc_root)/test/test.gyp:test_support_main',
@@ -31,6 +30,15 @@
         'codecs/tools/audio_codec_speed_test.cc',
       ],
       'conditions': [
+        ['include_opus==1', {
+          'dependencies': [
+            'webrtc_opus',
+          ],
+        }, {
+          'sources!': [
+            'codecs/opus/opus_speed_test.cc',
+          ],
+        }],
         ['OS=="android"', {
           'dependencies': [
             '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
Index: source/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
===================================================================
--- source.orig/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
+++ source/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
@@ -59,6 +59,36 @@
         },
       ],
     }],
+    ['include_opus==1', {
+      'targets': [
+        {
+          'target_name': 'audio_classifier_test',
+          'type': 'executable',
+          'dependencies': [
+            'neteq',
+            'webrtc_opus',
+          ],
+          'sources': [
+            'test/audio_classifier_test.cc',
+          ],
+        },
+        {
+          'target_name': 'neteq_opus_quality_test',
+          'type': 'executable',
+          'dependencies': [
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+            '<(webrtc_root)/test/test.gyp:test_support_main',
+            'neteq',
+            'neteq_test_support',
+            'webrtc_opus',
+          ],
+          'sources': [
+            'test/neteq_opus_quality_test.cc',
+          ],
+        },
+      ],
+    }]
   ],
   'targets': [
     {
@@ -74,7 +104,6 @@
         'isac',
         'neteq_test_tools',  # Test helpers
         'pcm16b',
-        'webrtc_opus',
       ],
       'defines': [
         'CODEC_ILBC',
@@ -91,7 +120,6 @@
         'CODEC_CNGCODEC32',
         'CODEC_ATEVENT_DECODE',
         'CODEC_RED',
-        'CODEC_OPUS',
       ],
       'include_dirs': [
         'include',
@@ -105,6 +133,16 @@
       'msvs_disabled_warnings': [
         4267,  # size_t to int truncation.
       ],
+      'conditions': [
+        ['include_opus==1', {
+          'dependencies': [
+            'webrtc_opus',
+          ],
+          'defines': [
+            'CODEC_OPUS',
+          ],
+        }],
+      ],
     },
 
     {
@@ -181,18 +219,6 @@
     },
 
     {
-      'target_name': 'audio_classifier_test',
-      'type': 'executable',
-      'dependencies': [
-        'neteq',
-        'webrtc_opus',
-      ],
-      'sources': [
-        'test/audio_classifier_test.cc',
-      ],
-    },
-
-    {
       'target_name': 'neteq_test_support',
       'type': 'static_library',
       'dependencies': [
@@ -226,22 +252,6 @@
       ],
     },
 
-    {
-      'target_name': 'neteq_opus_quality_test',
-      'type': 'executable',
-      'dependencies': [
-        '<(DEPTH)/testing/gtest.gyp:gtest',
-        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-        '<(webrtc_root)/test/test.gyp:test_support_main',
-        'neteq',
-        'neteq_test_support',
-        'webrtc_opus',
-      ],
-      'sources': [
-        'test/neteq_opus_quality_test.cc',
-      ],
-    },
-
     {
       'target_name': 'neteq_isac_quality_test',
       'type': 'executable',
Index: source/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
===================================================================
--- source.orig/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ source/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -444,6 +444,7 @@ class AudioDecoderG722StereoTest : publi
 class AudioDecoderOpusTest : public AudioDecoderTest {
  protected:
   AudioDecoderOpusTest() : AudioDecoderTest() {
+#ifdef WEBRTC_CODEC_OPUS
     codec_input_rate_hz_ = 48000;
     frame_size_ = 480;
     data_length_ = 10 * frame_size_;
@@ -453,12 +454,14 @@ class AudioDecoderOpusTest : public Audi
     config.payload_type = payload_type_;
     config.application = AudioEncoderOpus::kVoip;
     audio_encoder_.reset(new AudioEncoderOpus(config));
+#endif
   }
 };
 
 class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
  protected:
   AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
+#ifdef WEBRTC_CODEC_OPUS
     channels_ = 2;
     delete decoder_;
     decoder_ = new AudioDecoderOpus(2);
@@ -468,6 +471,7 @@ class AudioDecoderOpusStereoTest : publi
     config.payload_type = payload_type_;
     config.application = AudioEncoderOpus::kAudio;
     audio_encoder_.reset(new AudioEncoderOpus(config));
+#endif
   }
 };
 
Index: source/webrtc/modules/audio_coding/test/Tester.cc
===================================================================
--- source.orig/webrtc/modules/audio_coding/test/Tester.cc
+++ source/webrtc/modules/audio_coding/test/Tester.cc
@@ -120,6 +120,7 @@ TEST(AudioCodingModuleTest, TestWebRtcVa
   Trace::ReturnTrace();
 }
 
+#ifdef WEBRTC_CODEC_OPUS
 TEST(AudioCodingModuleTest, TestOpusDtx) {
   Trace::CreateTrace();
   Trace::SetTraceFile((webrtc::test::OutputPath() +
@@ -135,6 +136,7 @@ TEST(AudioCodingModuleTest, TestOpus) {
   webrtc::OpusTest().Perform();
   Trace::ReturnTrace();
 }
+#endif
 
 TEST(AudioCodingModuleTest, TestPacketLoss) {
   Trace::CreateTrace();
Index: source/webrtc/modules/modules.gyp
===================================================================
--- source.orig/webrtc/modules/modules.gyp
+++ source/webrtc/modules/modules.gyp
@@ -100,6 +100,11 @@
             'video_coding/codecs/vp8/test/vp8_impl_unittest.cc',
           ],
           'conditions': [
+            ['include_opus!=1', {
+              'sources!': [
+                'audio_coding/test/opus_test.cc',
+              ],
+            }],
             ['OS=="android"', {
               'dependencies': [
                 '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
@@ -387,8 +392,17 @@
             'video_processing/test/denoiser_test.cc',
             'video_processing/test/video_processing_unittest.cc',
             'video_processing/test/video_processing_unittest.h',
+            'audio_processing/test/bitexactness_tools.cc',
+            'audio_processing/test/audio_buffer_tools.cc',
           ],
           'conditions': [
+            ['include_opus!=1', {
+              'sources!': [
+                'audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
+                'audio_coding/codecs/opus/opus_unittest.cc',
+                'audio_coding/neteq/audio_classifier_unittest.cc',
+              ],
+            }],
             ['libvpx_build_vp9==1', {
               'sources': [
                 'video_coding/codecs/vp9/screenshare_layers_unittest.cc',
Index: source/webrtc/modules/video_capture/video_capture.gypi
===================================================================
--- source.orig/webrtc/modules/video_capture/video_capture.gypi
+++ source/webrtc/modules/video_capture/video_capture.gypi
@@ -37,7 +37,7 @@
       # Default video capture module implementation that only supports external
       # capture.
       'target_name': 'video_capture',
-      'type': 'static_library',
+      'type': 'none',
       'dependencies': [
         'video_capture_module',
       ],
@@ -188,7 +188,7 @@
                 '-lpthread -lm',
               ],
             }],
-            ['OS=="linux"', {
+            ['OS=="linux" and use_x11==1', {
               'libraries': [
                 '-lrt',
                 '-lXext',
Index: source/webrtc/base/array_view_unittest.cc
===================================================================
--- source.orig/webrtc/base/array_view_unittest.cc
+++ source/webrtc/base/array_view_unittest.cc
@@ -131,7 +131,7 @@ TEST(ArrayViewTest, TestRtcBuffer) {
   EXPECT_EQ(10u, y.size());
   EXPECT_EQ(b.data(), y.data());
   // ArrayView<char> d = b;  // Compile error, because incompatible types.
-  const rtc::Buffer cb = "very const";
+  const rtc::Buffer cb("very const");
   Call<const uint8_t>(cb);
   // Call<uint8_t>(cb);  // Compile error, because can't drop const.
   ArrayView<const uint8_t> z = cb;
Index: source/webrtc/modules/audio_coding/audio_coding.gypi
===================================================================
--- source.orig/webrtc/modules/audio_coding/audio_coding.gypi
+++ source/webrtc/modules/audio_coding/audio_coding.gypi
@@ -161,6 +161,9 @@
           '../include',
           '<(webrtc_root)',
         ],
+        'defines': [
+          '<@(audio_coding_defines)',
+        ],
       },
       'conditions': [
         ['include_opus==1', {