Attachment 356793 Details for Bug 192316 – GTK patch

[patch] GTK patch

patch (text/plain), 10.68 KB, created by Alejandro G. Castro on 2018-12-07 02:56:07 PST

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Creator: Alejandro G. Castro

Created: 2018-12-07 02:56:07 PST

Size: 10.68 KB

patch

obsolete

>diff --git a/Source/ThirdParty/libwebrtc/CMakeLists.txt b/Source/ThirdParty/libwebrtc/CMakeLists.txt
>index fa21b279c79..6c0856e53cd 100644
>--- a/Source/ThirdParty/libwebrtc/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/CMakeLists.txt
>@@ -24,6 +24,11 @@ endif ()
> set(webrtc_SOURCES
>     Source/third_party/abseil-cpp/absl/base/dynamic_annotations.cc
>     Source/third_party/abseil-cpp/absl/base/internal/raw_logging.cc
>+    Source/third_party/abseil-cpp/absl/base/internal/throw_delegate.cc
>+    Source/third_party/abseil-cpp/absl/strings/ascii.cc
>+    Source/third_party/abseil-cpp/absl/strings/internal/memutil.cc
>+    Source/third_party/abseil-cpp/absl/strings/match.cc
>+    Source/third_party/abseil-cpp/absl/strings/string_view.cc
>     Source/third_party/abseil-cpp/absl/types/bad_optional_access.cc
>     Source/third_party/abseil-cpp/absl/types/bad_variant_access.cc
>     Source/third_party/abseil-cpp/absl/types/optional.cc
>@@ -364,12 +369,15 @@ set(webrtc_SOURCES
>     Source/webrtc/api/audio_options.cc
>     Source/webrtc/api/call/transport.cc
>     Source/webrtc/api/candidate.cc
>+    Source/webrtc/api/create_peerconnection_factory.cc
>+    Source/webrtc/api/crypto/cryptooptions.cc
>     Source/webrtc/api/datachannelinterface.cc
>     Source/webrtc/api/jsep.cc
>     Source/webrtc/api/jsepicecandidate.cc
>     Source/webrtc/api/mediaconstraintsinterface.cc
>     Source/webrtc/api/mediastreaminterface.cc
>     Source/webrtc/api/mediatypes.cc
>+    Source/webrtc/api/media_transport_interface.cc
>     Source/webrtc/api/peerconnectioninterface.cc
>     Source/webrtc/api/proxy.cc
>     Source/webrtc/api/rtcerror.cc
>@@ -380,13 +388,17 @@ set(webrtc_SOURCES
>     Source/webrtc/api/rtptransceiverinterface.cc
>     Source/webrtc/api/statstypes.cc
>     Source/webrtc/api/transport/bitrate_settings.cc
>+    Source/webrtc/api/transport/goog_cc_factory.cc
>     Source/webrtc/api/transport/network_types.cc
>     Source/webrtc/api/units/data_rate.cc
>     Source/webrtc/api/units/data_size.cc
>     Source/webrtc/api/units/time_delta.cc
>     Source/webrtc/api/units/timestamp.cc
>+    Source/webrtc/api/video/builtin_video_bitrate_allocator_factory.cc
>     Source/webrtc/api/video/color_space.cc
>+    Source/webrtc/api/video/encoded_image.cc
>     Source/webrtc/api/video/encoded_frame.cc
>+    Source/webrtc/api/video/hdr_metadata.cc
>     Source/webrtc/api/video/i010_buffer.cc
>     Source/webrtc/api/video/i420_buffer.cc
>     Source/webrtc/api/video/video_bitrate_allocation.cc
>@@ -401,6 +413,7 @@ set(webrtc_SOURCES
>     Source/webrtc/api/video_codecs/video_decoder_factory.cc
>     Source/webrtc/api/video_codecs/builtin_video_decoder_factory.cc
>     Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc
>+    Source/webrtc/api/video_codecs/create_vp8_temporal_layers.cc
>     Source/webrtc/api/video_codecs/sdp_video_format.cc
>     Source/webrtc/api/video_codecs/video_codec.cc
>     Source/webrtc/api/video_codecs/video_decoder.cc
>@@ -512,7 +525,6 @@ set(webrtc_SOURCES
>     Source/webrtc/common_video/i420_buffer_pool.cc
>     Source/webrtc/common_video/incoming_video_stream.cc
>     Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
>-    Source/webrtc/common_video/video_frame.cc
>     Source/webrtc/common_video/video_frame_buffer.cc
>     Source/webrtc/common_video/video_render_frames.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc
>@@ -914,6 +926,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
>     Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>     Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>+    Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc
>     Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>     Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.cc
>     Source/webrtc/modules/congestion_controller/bbr/bbr_factory.cc
>@@ -924,6 +937,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>@@ -931,6 +945,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
>     Source/webrtc/modules/congestion_controller/receive_side_congestion_controller.cc
>+    Source/webrtc/modules/congestion_controller/rtp/control_handler.cc
>     Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc
>     Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc
>     Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
>@@ -965,6 +980,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc
>     Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
>     Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_size.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
>@@ -1096,6 +1112,7 @@ set(webrtc_SOURCES
>     Source/webrtc/p2p/base/basicpacketsocketfactory.cc
>     Source/webrtc/p2p/base/dtlstransport.cc
>     Source/webrtc/p2p/base/dtlstransportinternal.cc
>+    Source/webrtc/p2p/base/icecredentialsiterator.cc
>     Source/webrtc/p2p/base/icetransportinternal.cc
>     Source/webrtc/p2p/base/p2pconstants.cc
>     Source/webrtc/p2p/base/p2ptransportchannel.cc
>@@ -1184,19 +1201,22 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/event_tracer.cc
>     Source/webrtc/rtc_base/experiments/alr_experiment.cc
>     Source/webrtc/rtc_base/experiments/congestion_controller_experiment.cc
>+    Source/webrtc/rtc_base/experiments/cpu_speed_experiment.cc
>     Source/webrtc/rtc_base/experiments/field_trial_parser.cc
>     Source/webrtc/rtc_base/experiments/field_trial_units.cc
>+    Source/webrtc/rtc_base/experiments/jitter_upper_bound_experiment.cc
>+    Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment.cc
>     Source/webrtc/rtc_base/experiments/quality_scaling_experiment.cc
>     Source/webrtc/rtc_base/experiments/rtt_mult_experiment.cc
>     Source/webrtc/rtc_base/file.cc
>     Source/webrtc/rtc_base/file_posix.cc
>-    Source/webrtc/rtc_base/fileutils.cc
>     Source/webrtc/rtc_base/firewallsocketserver.cc
>     Source/webrtc/rtc_base/flags.cc
>     Source/webrtc/rtc_base/helpers.cc
>     Source/webrtc/rtc_base/httpcommon.cc
>     Source/webrtc/rtc_base/ifaddrs_converter.cc
>     Source/webrtc/rtc_base/ipaddress.cc
>+    Source/webrtc/rtc_base/key_derivation.cc
>     Source/webrtc/rtc_base/location.cc
>     Source/webrtc/rtc_base/logsinks.cc
>     Source/webrtc/rtc_base/logging.cc
>@@ -1212,9 +1232,11 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/nethelpers.cc
>     Source/webrtc/rtc_base/network.cc
>     Source/webrtc/rtc_base/networkmonitor.cc
>+    Source/webrtc/rtc_base/network/sent_packet.cc
>     Source/webrtc/rtc_base/nullsocketserver.cc
>     Source/webrtc/rtc_base/numerics/exp_filter.cc
>     Source/webrtc/rtc_base/numerics/histogram_percentile_counter.cc
>+    Source/webrtc/rtc_base/numerics/moving_average.cc
>     Source/webrtc/rtc_base/numerics/sample_counter.cc
>     Source/webrtc/rtc_base/openssladapter.cc
>     Source/webrtc/rtc_base/opensslcertificate.cc
>@@ -1223,6 +1245,7 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/opensslsessioncache.cc
>     Source/webrtc/rtc_base/opensslstreamadapter.cc
>     Source/webrtc/rtc_base/opensslutility.cc
>+    Source/webrtc/rtc_base/openssl_key_derivation_hkdf.cc
>     Source/webrtc/rtc_base/physicalsocketserver.cc
>     Source/webrtc/rtc_base/platform_file.cc
>     Source/webrtc/rtc_base/platform_thread.cc
>@@ -1288,6 +1311,7 @@ set(webrtc_SOURCES
>     Source/webrtc/video/call_stats.cc
>     Source/webrtc/video/encoder_rtcp_feedback.cc
>     Source/webrtc/video/overuse_frame_detector.cc
>+    Source/webrtc/video/frame_dumping_decoder.cc
>     Source/webrtc/video/quality_threshold.cc
>     Source/webrtc/video/receive_statistics_proxy.cc
>     Source/webrtc/video/report_block_stats.cc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>index 44ec3f50c19..58b3345f9db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>@@ -75,7 +75,12 @@ CriticalSection& logCriticalScope() {
> void LogSink::OnLogMessage(const std::string& msg,
>                            LoggingSeverity severity,
>                            const char* tag) {
>-  OnLogMessage(tag + (": " + msg));
>+  OnLogMessage(tag + (": " + msg), severity);
>+}
>+
>+void LogSink::OnLogMessage(const std::string& msg,
>+                           LoggingSeverity /* severity */) {
>+  OnLogMessage(msg);
> }
> 
> /////////////////////////////////////////////////////////////////////////////
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>index 905bbdac0e2..27b8801bb0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>@@ -22,7 +22,13 @@
> #include <type_traits>
> #include <utility>
> 
>+#if defined(WEBRTC_LINUX)
>+#include <event2/event.h>
>+#include <event2/event_compat.h>
>+#include <event2/event_struct.h>
>+#else
> #include "base/third_party/libevent/event.h"
>+#endif
> #include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>index ac6445fa00c..bc12b6cdf0a 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>@@ -186,11 +186,6 @@ public:
>         return WEBRTC_VIDEO_CODEC_OK;
>     }
> 
>-    int32_t SetChannelParameters(uint32_t, int64_t) final
>-    {
>-        return WEBRTC_VIDEO_CODEC_OK;
>-    }
>-
>     int32_t Encode(const webrtc::VideoFrame& frame,
>         const webrtc::CodecSpecificInfo* codecInfo,
>         const std::vector<webrtc::FrameType>* frameTypes) final

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Attachments on bug 192316: 356390 | 356436 | 356469 | 356793 | 356816 | 356841