Created
December 9, 2016 06:50
-
-
Save kapejod/d99ec6e6ce206bd584055ec75ad63740 to your computer and use it in GitHub Desktop.
How i have built WebRTC for tvOs (some time ago....might not apply anymore)
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
diff --git a/webrtc/base/posix.cc b/webrtc/base/posix.cc | |
index 0eb24ee..7e48273 100644 | |
--- a/webrtc/base/posix.cc | |
+++ b/webrtc/base/posix.cc | |
@@ -44,7 +44,7 @@ enum { | |
bool RunAsDaemon(const char *file, const char *const argv[]) { | |
// Fork intermediate child to daemonize. | |
- pid_t pid = fork(); | |
+ pid_t pid = -1; | |
if (pid < 0) { | |
LOG_ERR(LS_ERROR) << "fork()"; | |
return false; | |
@@ -68,7 +68,7 @@ bool RunAsDaemon(const char *file, const char *const argv[]) { | |
#endif | |
// Fork again to become a daemon. | |
- pid = fork(); | |
+ pid = -1; | |
// It is important that everything here use _exit() and not exit(), because | |
// exit() would call the destructors of all global variables in the whole | |
// process, which is both unnecessary and unsafe. | |
@@ -81,7 +81,7 @@ bool RunAsDaemon(const char *file, const char *const argv[]) { | |
// WEBRTC_POSIX requires the args to be typed as non-const for historical | |
// reasons, but it mandates that the actual implementation be const, so | |
// the cast is safe. | |
- execvp(file, const_cast<char *const *>(argv)); | |
+ // execvp(file, const_cast<char *const *>(argv)); | |
_exit(255); // if execvp failed | |
} | |
diff --git a/webrtc/build/ios/client.webrtc/iOS64_Debug.json b/webrtc/build/ios/client.webrtc/iOS64_Debug.json | |
index 657ba91..110f80f 100644 | |
--- a/webrtc/build/ios/client.webrtc/iOS64_Debug.json | |
+++ b/webrtc/build/ios/client.webrtc/iOS64_Debug.json | |
@@ -12,7 +12,7 @@ | |
}, | |
"compiler": "ninja", | |
"configuration": "Debug", | |
- "sdk": "iphoneos9.0", | |
+ "sdk": "tvos9.0", | |
"tests": [ | |
] | |
} | |
diff --git a/webrtc/build/ios/client.webrtc/iOS64_Release.json b/webrtc/build/ios/client.webrtc/iOS64_Release.json | |
index 097a1b1..d9e1095 100644 | |
--- a/webrtc/build/ios/client.webrtc/iOS64_Release.json | |
+++ b/webrtc/build/ios/client.webrtc/iOS64_Release.json | |
@@ -12,7 +12,7 @@ | |
}, | |
"compiler": "ninja", | |
"configuration": "Release", | |
- "sdk": "iphoneos9.0", | |
+ "sdk": "appletvos9.1", | |
"tests": [ | |
] | |
} | |
diff --git a/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.m b/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.m | |
index 6f52657..e855efb 100644 | |
--- a/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.m | |
+++ b/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.m | |
@@ -120,9 +120,7 @@ static CGFloat const kAppLabelHeight = 20; | |
UILabel *_appLabel; | |
ARDRoomTextField *_roomText; | |
UILabel *_callOptionsLabel; | |
- UISwitch *_audioOnlySwitch; | |
UILabel *_audioOnlyLabel; | |
- UISwitch *_loopbackSwitch; | |
UILabel *_loopbackLabel; | |
UIButton *_startCallButton; | |
UIButton *_audioLoopButton; | |
@@ -162,9 +160,6 @@ static CGFloat const kAppLabelHeight = 20; | |
[_callOptionsLabel sizeToFit]; | |
[self addSubview:_callOptionsLabel]; | |
- _audioOnlySwitch = [[UISwitch alloc] initWithFrame:CGRectZero]; | |
- [_audioOnlySwitch sizeToFit]; | |
- [self addSubview:_audioOnlySwitch]; | |
_audioOnlyLabel = [[UILabel alloc] initWithFrame:CGRectZero]; | |
_audioOnlyLabel.text = @"Audio only"; | |
@@ -173,9 +168,6 @@ static CGFloat const kAppLabelHeight = 20; | |
[_audioOnlyLabel sizeToFit]; | |
[self addSubview:_audioOnlyLabel]; | |
- _loopbackSwitch = [[UISwitch alloc] initWithFrame:CGRectZero]; | |
- [_loopbackSwitch sizeToFit]; | |
- [self addSubview:_loopbackSwitch]; | |
_loopbackLabel = [[UILabel alloc] initWithFrame:CGRectZero]; | |
_loopbackLabel.text = @"Loopback mode"; | |
@@ -242,34 +234,6 @@ static CGFloat const kAppLabelHeight = 20; | |
CGFloat audioOnlyTop = | |
CGRectGetMaxY(_callOptionsLabel.frame) + kCallControlMargin * 2; | |
- CGRect audioOnlyRect = CGRectMake(kCallControlMargin * 3, | |
- audioOnlyTop, | |
- _audioOnlySwitch.frame.size.width, | |
- _audioOnlySwitch.frame.size.height); | |
- _audioOnlySwitch.frame = audioOnlyRect; | |
- CGFloat audioOnlyLabelCenterX = CGRectGetMaxX(audioOnlyRect) + | |
- kCallControlMargin + _audioOnlyLabel.frame.size.width / 2; | |
- _audioOnlyLabel.center = CGPointMake(audioOnlyLabelCenterX, | |
- CGRectGetMidY(audioOnlyRect)); | |
- | |
- CGFloat loopbackModeTop = | |
- CGRectGetMaxY(_audioOnlySwitch.frame) + kCallControlMargin; | |
- CGRect loopbackModeRect = CGRectMake(kCallControlMargin * 3, | |
- loopbackModeTop, | |
- _loopbackSwitch.frame.size.width, | |
- _loopbackSwitch.frame.size.height); | |
- _loopbackSwitch.frame = loopbackModeRect; | |
- CGFloat loopbackModeLabelCenterX = CGRectGetMaxX(loopbackModeRect) + | |
- kCallControlMargin + _loopbackLabel.frame.size.width / 2; | |
- _loopbackLabel.center = CGPointMake(loopbackModeLabelCenterX, | |
- CGRectGetMidY(loopbackModeRect)); | |
- | |
- CGFloat audioLoopTop = | |
- CGRectGetMaxY(loopbackModeRect) + kCallControlMargin * 3; | |
- _audioLoopButton.frame = CGRectMake(kCallControlMargin, | |
- audioLoopTop, | |
- _audioLoopButton.frame.size.width, | |
- _audioLoopButton.frame.size.height); | |
CGFloat startCallTop = | |
CGRectGetMaxY(_audioLoopButton.frame) + kCallControlMargin * 3; | |
@@ -307,14 +271,14 @@ static CGFloat const kAppLabelHeight = 20; | |
- (void)onStartCall:(id)sender { | |
NSString *room = _roomText.roomText; | |
// If this is a loopback call, allow a generated room name. | |
- if (!room.length && _loopbackSwitch.isOn) { | |
+ if (!room.length) { | |
room = [[NSUUID UUID] UUIDString]; | |
} | |
room = [room stringByReplacingOccurrencesOfString:@"-" withString:@""]; | |
[_delegate mainView:self | |
didInputRoom:room | |
- isLoopback:_loopbackSwitch.isOn | |
- isAudioOnly:_audioOnlySwitch.isOn]; | |
+ isLoopback:YES | |
+ isAudioOnly:NO]; | |
} | |
@end | |
diff --git a/webrtc/examples/objc/AppRTCDemo/ios/ARDMainViewController.m b/webrtc/examples/objc/AppRTCDemo/ios/ARDMainViewController.m | |
index 8de6f6a..b883eea 100644 | |
--- a/webrtc/examples/objc/AppRTCDemo/ios/ARDMainViewController.m | |
+++ b/webrtc/examples/objc/AppRTCDemo/ios/ARDMainViewController.m | |
@@ -80,12 +80,6 @@ | |
#pragma mark - Private | |
- (void)showAlertWithMessage:(NSString*)message { | |
- UIAlertView* alertView = [[UIAlertView alloc] initWithTitle:nil | |
- message:message | |
- delegate:nil | |
- cancelButtonTitle:@"OK" | |
- otherButtonTitles:nil]; | |
- [alertView show]; | |
} | |
@end | |
diff --git a/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m b/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m | |
index 51290a0..5b2a7a3 100644 | |
--- a/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m | |
+++ b/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m | |
@@ -185,12 +185,6 @@ | |
} | |
- (void)showAlertWithMessage:(NSString*)message { | |
- UIAlertView* alertView = [[UIAlertView alloc] initWithTitle:nil | |
- message:message | |
- delegate:nil | |
- cancelButtonTitle:@"OK" | |
- otherButtonTitles:nil]; | |
- [alertView show]; | |
} | |
@end | |
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc | |
index 02de6b8..c80113b 100644 | |
--- a/webrtc/media/engine/webrtcvideoengine2.cc | |
+++ b/webrtc/media/engine/webrtcvideoengine2.cc | |
@@ -163,8 +163,7 @@ bool CodecIsInternallySupported(const std::string& codec_name) { | |
return true; | |
} | |
if (CodecNamesEq(codec_name, kH264CodecName)) { | |
- return webrtc::H264Encoder::IsSupported() && | |
- webrtc::H264Decoder::IsSupported(); | |
+ return false; | |
} | |
return false; | |
} | |
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm | |
index 4390f49..0411f53 100644 | |
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm | |
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm | |
@@ -120,13 +120,9 @@ static bool VerifyAudioSession(RTCAudioSession* session) { | |
} | |
// Ensure that the required category and mode are actually activated. | |
- if (![session.category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) { | |
+ if (![session.category isEqualToString:AVAudioSessionCategoryPlayback) { | |
LOG(LS_ERROR) | |
- << "Failed to set category to AVAudioSessionCategoryPlayAndRecord"; | |
- return false; | |
- } | |
- if (![session.mode isEqualToString:AVAudioSessionModeVoiceChat]) { | |
- LOG(LS_ERROR) << "Failed to set mode to AVAudioSessionModeVoiceChat"; | |
+ << "Failed to set category to AVAudioSessionCategoryPlayback"; | |
return false; | |
} | |
return true; | |
@@ -156,19 +152,12 @@ static bool ActivateAudioSession(RTCAudioSession* session, bool activate) { | |
// audio sessions which are also nonmixable. | |
if (session.category != AVAudioSessionCategoryPlayAndRecord) { | |
error = nil; | |
- success = [session setCategory:AVAudioSessionCategoryPlayAndRecord | |
- withOptions:AVAudioSessionCategoryOptionAllowBluetooth | |
+ success = [session setCategory:AVAudioSessionCategoryPlayback | |
+ withOptions: 0 | |
error:&error]; | |
RTC_DCHECK(CheckAndLogError(success, error)); | |
} | |
- // Specify mode for two-way voice communication (e.g. VoIP). | |
- if (session.mode != AVAudioSessionModeVoiceChat) { | |
- error = nil; | |
- success = [session setMode:AVAudioSessionModeVoiceChat error:&error]; | |
- RTC_DCHECK(CheckAndLogError(success, error)); | |
- } | |
- | |
// Set the session's sample rate or the hardware sample rate. | |
// It is essential that we use the same sample rate as stream format | |
// to ensure that the I/O unit does not have to do sample rate conversion. | |
@@ -442,37 +431,12 @@ int32_t AudioDeviceIOS::StopRecording() { | |
// Change the default receiver playout route to speaker. | |
int32_t AudioDeviceIOS::SetLoudspeakerStatus(bool enable) { | |
LOGI() << "SetLoudspeakerStatus(" << enable << ")"; | |
- | |
- RTCAudioSession* session = [RTCAudioSession sharedInstance]; | |
- [session lockForConfiguration]; | |
- NSString* category = session.category; | |
- AVAudioSessionCategoryOptions options = session.categoryOptions; | |
- // Respect old category options if category is | |
- // AVAudioSessionCategoryPlayAndRecord. Otherwise reset it since old options | |
- // might not be valid for this category. | |
- if ([category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) { | |
- if (enable) { | |
- options |= AVAudioSessionCategoryOptionDefaultToSpeaker; | |
- } else { | |
- options &= ~AVAudioSessionCategoryOptionDefaultToSpeaker; | |
- } | |
- } else { | |
- options = AVAudioSessionCategoryOptionDefaultToSpeaker; | |
- } | |
- NSError* error = nil; | |
- BOOL success = [session setCategory:AVAudioSessionCategoryPlayAndRecord | |
- withOptions:options | |
- error:&error]; | |
- ios::CheckAndLogError(success, error); | |
- [session unlockForConfiguration]; | |
- return (error == nil) ? 0 : -1; | |
+ return 0; | |
} | |
int32_t AudioDeviceIOS::GetLoudspeakerStatus(bool& enabled) const { | |
LOGI() << "GetLoudspeakerStatus"; | |
- RTCAudioSession* session = [RTCAudioSession sharedInstance]; | |
- AVAudioSessionCategoryOptions options = session.categoryOptions; | |
- enabled = options & AVAudioSessionCategoryOptionDefaultToSpeaker; | |
+ enabled = 1; | |
return 0; | |
} | |
diff --git a/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm b/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm | |
index ea7c546..1fa6182 100644 | |
--- a/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm | |
+++ b/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm | |
@@ -266,7 +266,7 @@ NSInteger const kRTCAudioSessionErrorLockRequired = -1; | |
if (![self checkLock:outError]) { | |
return NO; | |
} | |
- return [self.session setCategory:category withOptions:options error:outError]; | |
+ return [self.session setCategory:category error:outError]; | |
} | |
- (BOOL)setMode:(NSString *)mode error:(NSError **)outError { | |
diff --git a/webrtc/modules/video_capture/video_capture_factory.cc b/webrtc/modules/video_capture/video_capture_factory.cc | |
index 618c08b..8cc9222 100644 | |
--- a/webrtc/modules/video_capture/video_capture_factory.cc | |
+++ b/webrtc/modules/video_capture/video_capture_factory.cc | |
@@ -17,25 +17,17 @@ namespace webrtc | |
VideoCaptureModule* VideoCaptureFactory::Create(const int32_t id, | |
const char* deviceUniqueIdUTF8) { | |
-#if defined(ANDROID) | |
return nullptr; | |
-#else | |
- return videocapturemodule::VideoCaptureImpl::Create(id, deviceUniqueIdUTF8); | |
-#endif | |
} | |
VideoCaptureModule* VideoCaptureFactory::Create(const int32_t id, | |
VideoCaptureExternal*& externalCapture) { | |
- return videocapturemodule::VideoCaptureImpl::Create(id, externalCapture); | |
+ return nullptr; | |
} | |
VideoCaptureModule::DeviceInfo* VideoCaptureFactory::CreateDeviceInfo( | |
const int32_t id) { | |
-#if defined(ANDROID) | |
return nullptr; | |
-#else | |
- return videocapturemodule::VideoCaptureImpl::CreateDeviceInfo(id); | |
-#endif | |
} | |
} // namespace webrtc | |
diff --git a/webrtc/modules/video_coding/codec_database.cc b/webrtc/modules/video_coding/codec_database.cc | |
index a5a7c1e..777184c 100644 | |
--- a/webrtc/modules/video_coding/codec_database.cc | |
+++ b/webrtc/modules/video_coding/codec_database.cc | |
@@ -586,10 +586,7 @@ VCMGenericDecoder* VCMCodecDataBase::CreateDecoder(VideoCodecType type) const { | |
case kVideoCodecI420: | |
return new VCMGenericDecoder(new I420Decoder()); | |
case kVideoCodecH264: | |
- if (H264Decoder::IsSupported()) { | |
- return new VCMGenericDecoder(H264Decoder::Create()); | |
- } | |
- break; | |
+ return nullptr; | |
default: | |
break; | |
} | |
diff --git a/webrtc/video/overuse_frame_detector.cc b/webrtc/video/overuse_frame_detector.cc | |
index 522a505..4f56137 100644 | |
--- a/webrtc/video/overuse_frame_detector.cc | |
+++ b/webrtc/video/overuse_frame_detector.cc | |
@@ -56,7 +56,7 @@ CpuOveruseOptions::CpuOveruseOptions() | |
min_frame_samples(120), | |
min_process_count(3), | |
high_threshold_consecutive_count(2) { | |
-#if defined(WEBRTC_MAC) | |
+#if defined(WEBRTC_MAC_NOTHANKS) | |
// This is proof-of-concept code for letting the physical core count affect | |
// the interval into which we attempt to scale. For now, the code is Mac OS | |
// specific, since that's the platform were we saw most problems. | |
diff --git a/webrtc/video/video_decoder.cc b/webrtc/video/video_decoder.cc | |
index 8e0a503..5b9bedc 100644 | |
--- a/webrtc/video/video_decoder.cc | |
+++ b/webrtc/video/video_decoder.cc | |
@@ -19,13 +19,11 @@ | |
namespace webrtc { | |
VideoDecoder* VideoDecoder::Create(VideoDecoder::DecoderType codec_type) { | |
switch (codec_type) { | |
- case kH264: | |
- RTC_DCHECK(H264Decoder::IsSupported()); | |
- return H264Decoder::Create(); | |
case kVp8: | |
return VP8Decoder::Create(); | |
case kVp9: | |
return VP9Decoder::Create(); | |
+ case kH264: | |
case kUnsupportedCodec: | |
LOG(LS_ERROR) << "Creating NullVideoDecoder for unsupported codec."; | |
return new NullVideoDecoder(); | |
diff --git a/webrtc/video/video_encoder.cc b/webrtc/video/video_encoder.cc | |
index e85e3d9..cf34141 100644 | |
--- a/webrtc/video/video_encoder.cc | |
+++ b/webrtc/video/video_encoder.cc | |
@@ -19,13 +19,11 @@ | |
namespace webrtc { | |
VideoEncoder* VideoEncoder::Create(VideoEncoder::EncoderType codec_type) { | |
switch (codec_type) { | |
- case kH264: | |
- RTC_DCHECK(H264Encoder::IsSupported()); | |
- return H264Encoder::Create(); | |
case kVp8: | |
return VP8Encoder::Create(); | |
case kVp9: | |
return VP9Encoder::Create(); | |
+ case kH264: | |
case kUnsupportedCodec: | |
RTC_NOTREACHED(); | |
return nullptr; |
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
listcontains() { | |
for word in $1; do | |
[[ $word = $2 ]] && return 0 | |
done | |
return 1 | |
} | |
ninjas=`find src/out_tvos -name '*.ninja'` | |
ninjas=$(echo "$ninjas" | tr ' ' '\n' | sort -u | tr '\n' ' ') | |
exclude_ninjas="src/out_tvos/Release-iphoneos/obj/webrtc/modules/webrtc_h264_video_toolbox.ninja src/out_tvos/Release-iphoneos/obj/webrtc/modules/webrtc_h264.ninja src/out_tvos/Release-iphoneos/obj/webrtc/modules/video_capture_module_internal_impl.ninja" | |
echo $ninjas | |
for ninja in $ninjas; do | |
if listcontains "$exclude_ninjas" "$ninja"; | |
then | |
echo "ignoring $ninja" | |
else | |
sed -i -- "s/iPhoneOS.platform\/Developer\/SDKs\/iPhoneOS9.2.sdk/AppleTVOS.platform\/Developer\/SDKs\/AppleTVOS9.1.sdk/g" $ninja | |
sed -i -- "s/libwebrtc_h264_video_toolbox.a//g" $ninja | |
sed -i -- "s/libwebrtc_h264.a//g" $ninja | |
sed -i -- "s/-framework VideoToolbox//g" $ninja | |
sed -i -- "s/iphoneos-version/appletvos-version/g" $ninja | |
sed -i -- "s/libvideo_capture_module_internal_impl.a//g" $ninja | |
fi | |
done | |
sed -i -- "s/obj\/talk\/app\/webrtc\/objc\/libjingle_peerconnection_objc.avfoundationvideocapturer.o $/g" src/out_tvos/Release-iphoneos/obj/talk/app/webrtc/ibjingle_peerconnection_objc.ninja | |
sed -i -- "s/obj\/talk\/app\/webrtc\/objc\/libjingle_peerconnection_objc.RTCAVFoundationVideoSource.o $/g" src/out_tvos/Release-iphoneos/obj/talk/app/webrtc/ibjingle_peerconnection_objc.ninja |
Sign up for free
to join this conversation on GitHub.
Already have an account?
Sign in to comment