Bug 1389598 - Part 2: Remove gonk references from media/ r=jesup

--HG-- extra : rebase_source : d1af2d0987038e1c0b0b0c971d0d2e4e9f08364a
dothq · Aug 12, 2017 · 5051390 · 5051390
1 parent 79e2533
commit 5051390
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Showing 16 changed files with 10 additions and 224 deletions.
diff --git a/build/gyp.mozbuild b/build/gyp.mozbuild
@@ -60,7 +60,6 @@ gyp_vars.update({
     'arm_neon': 0,
     'arm_neon_optional': 1,
 
-    'moz_widget_toolkit_gonk': 0,
     'moz_webrtc_omx': 0,
     'moz_webrtc_mediacodec': 0,
 

diff --git a/media/mtransport/common.build b/media/mtransport/common.build
@@ -28,11 +28,6 @@ mtransport_lcppsrcs = [
     'transportlayerprsock.cpp',
 ]
 
-if CONFIG['MOZ_WIDGET_TOOLKIT'] == 'gonk':
-    mtransport_lcppsrcs += [
-        'gonk_addrs.cpp',
-    ]
-
 mtransport_cppsrcs = [
     '/media/mtransport/%s' % s for s in sorted(mtransport_lcppsrcs)
 ]

diff --git a/media/mtransport/gonk_addrs.cpp b/media/mtransport/gonk_addrs.cpp
diff --git a/media/mtransport/third_party/nICEr/nicer.gyp b/media/mtransport/third_party/nICEr/nicer.gyp
@@ -7,7 +7,6 @@
 #
 {
   'variables' : {
-    'build_with_gonk%': 0,
     'have_ethtool_cmd_speed_hi%': 1
   },
   'targets' : [
@@ -234,18 +233,6 @@
 
 		 'sources': [
 		 ],
-              }],
-              ['moz_widget_toolkit_gonk==1', {
-                'defines' : [
-                  'WEBRTC_GONK',
-                  'NO_REG_RPC',
-                ],
-             }],
-             # Gonk has its own nr_stun_get_addrs implementation.
-             ['build_with_gonk==1', {
-               'defines': [
-                  "USE_PLATFORM_NR_STUN_GET_ADDRS",
-               ]
              }],
              ['have_ethtool_cmd_speed_hi==0', {
                'defines': [

diff --git a/media/webrtc/signaling/signaling.gyp b/media/webrtc/signaling/signaling.gyp
@@ -39,16 +39,6 @@
     'chromium_code': 1,
   },
 
-  'target_defaults': {
-    'conditions': [
-      ['moz_widget_toolkit_gonk==1', {
-        'defines' : [
-          'WEBRTC_GONK',
-       ],
-      }],
-    ],
-  },
-
   'targets': [
 
     #
@@ -296,7 +286,7 @@
           'cflags_mozilla': [
           ],
         }],
-        ['OS=="android" or moz_widget_toolkit_gonk==1', {
+        ['OS=="android"', {
           'cflags_mozilla': [
             # This warning complains about important MOZ_EXPORT attributes
             # on forward declarations for Android API types.

diff --git a/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp b/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
@@ -2259,12 +2259,8 @@ JsepSessionImpl::SetupDefaultCodecs()
       48000,
       2,
       960,
-#ifdef WEBRTC_GONK
       // TODO Move this elsewhere to be adaptive to rate - Bug 1207925
-      16000 // B2G uses lower capture sampling rate
-#else
       40000
-#endif
       ));
 
   mSupportedCodecs.values.push_back(new JsepAudioCodecDescription(

diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
@@ -25,10 +25,6 @@
 #include "MediaStreamVideoSink.h"
 #include "VideoUtils.h"
 #include "VideoStreamTrack.h"
-#ifdef WEBRTC_GONK
-#include "GrallocImages.h"
-#include "mozilla/layers/GrallocTextureClient.h"
-#endif
 
 #include "nsError.h"
 #include "AudioSegment.h"

diff --git a/media/webrtc/trunk/build/common.gypi b/media/webrtc/trunk/build/common.gypi
@@ -428,7 +428,7 @@
         }],
 
         # Flags to use X11 on non-Mac POSIX platforms
-        ['OS=="win" or OS=="mac" or OS=="ios" or OS=="android" or moz_widget_toolkit_gonk==1', {
+        ['OS=="win" or OS=="mac" or OS=="ios" or OS=="android"', {
           'use_glib%': 0,
           'use_x11%': 0,
         }, {

diff --git a/media/webrtc/trunk/webrtc/common_audio/resampler/resampler.cc b/media/webrtc/trunk/webrtc/common_audio/resampler/resampler.cc
@@ -23,7 +23,7 @@
 // TODO(jesup) better adjust per platform ability
 // Note: if these are changed (higher), you may need to change the
 // KernelDelay values in the unit tests here and in output_mixer.
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_GONK)
+#if defined(WEBRTC_ANDROID)
 #define RESAMPLER_QUALITY 2
 #else
 #define RESAMPLER_QUALITY 3

diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/gonk/audio_manager.cc b/media/webrtc/trunk/webrtc/modules/audio_device/gonk/audio_manager.cc
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/gonk/audio_manager.h b/media/webrtc/trunk/webrtc/modules/audio_device/gonk/audio_manager.h
diff --git a/media/webrtc/trunk/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc b/media/webrtc/trunk/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc
@@ -29,7 +29,7 @@
 
 namespace webrtc_adm_linux_pulse {
 
-#if defined(__OpenBSD__) || defined(WEBRTC_GONK)
+#if defined(__OpenBSD__)
 LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so")
 #else
 LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")

diff --git a/media/webrtc/trunk/webrtc/modules/video_capture/video_capture.h b/media/webrtc/trunk/webrtc/modules/video_capture/video_capture.h
@@ -17,7 +17,7 @@
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/video_capture/video_capture_defines.h"
 
-#if defined(ANDROID) && !defined(WEBRTC_GONK)
+#if defined(ANDROID)
 #include <jni.h>
 #endif
 

diff --git a/media/webrtc/trunk/webrtc/system_wrappers/include/static_instance.h b/media/webrtc/trunk/webrtc/system_wrappers/include/static_instance.h
@@ -13,7 +13,7 @@
 
 #include <assert.h>
 
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_GONK)
+#if defined(WEBRTC_ANDROID)
 #define OS_LINUX
 #endif
 #include "base/singleton.h"

diff --git a/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_features.cc b/media/webrtc/trunk/webrtc/system_wrappers/source/cpu_features.cc
@@ -67,7 +67,7 @@ static int GetCPUInfo(CPUFeature feature) {
   return 0;
 }
 
-#if !defined(WEBRTC_GONK) && !defined(ANDROID)
+#if !defined(ANDROID)
 #ifdef WEBRTC_ARCH_ARM_V7
 uint64_t WebRtc_GetCPUFeaturesARM(void) {
   return kCPUFeatureARMv7
@@ -77,7 +77,7 @@ uint64_t WebRtc_GetCPUFeaturesARM(void) {
          | kCPUFeatureVFPv3;
 }
 #endif // WEBRTC_ARCH_ARM_V7
-#endif // !WEBRTC_GONK && !ANDROID
+#endif // !ANDROID
 
 #endif
 

diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_defines.h b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_defines.h
@@ -60,13 +60,13 @@ enum { kVoiceEngineMaxModuleVersionSize = 960 };
 // Audio processing
 const NoiseSuppression::Level kDefaultNsMode = NoiseSuppression::kModerate;
 const GainControl::Mode kDefaultAgcMode =
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_GONK)
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
     GainControl::kAdaptiveDigital;
 #else
     GainControl::kAdaptiveAnalog;
 #endif
 const bool kDefaultAgcState =
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_GONK)
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
     false;
 #else
     true;