Diff
Modified: trunk/LayoutTests/ChangeLog (276490 => 276491)
--- trunk/LayoutTests/ChangeLog 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/LayoutTests/ChangeLog 2021-04-23 09:23:04 UTC (rev 276491)
@@ -1,3 +1,17 @@
+2021-04-23 Philippe Normand <[email protected]>
+
+ [LibWebRTC][WPE][GTK] Enable vp9 and NO_MAIN_THREAD_WRAPPING
+ https://bugs.webkit.org/show_bug.cgi?id=222795
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ Update GLIB baselines for webrtc tests.
+
+ * platform/glib/TestExpectations:
+ * platform/glib/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt: Added.
+ * platform/glib/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-restartIce.https-expected.txt: Added.
+ * platform/glib/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setCodecPreferences-expected.txt: Added.
+
2021-04-22 Diego Pino Garcia <[email protected]>
[GTK] Unreviewed test gardening. Update baseline after r276448.
Modified: trunk/LayoutTests/platform/glib/TestExpectations (276490 => 276491)
--- trunk/LayoutTests/platform/glib/TestExpectations 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/LayoutTests/platform/glib/TestExpectations 2021-04-23 09:23:04 UTC (rev 276491)
@@ -1120,8 +1120,6 @@
webkit.org/b/210272 webrtc/datachannel/multiple-connections.html [ Timeout Pass ]
webkit.org/b/212809 [ Release ] webrtc/datachannel/gather-candidates-networkprocess-crash.html [ Timeout Pass ]
-webkit.org/b/213780 webrtc/vp9.html [ Failure ]
-
webkit.org/b/215005 webkit.org/b/218787 webrtc/h264-baseline.html [ Crash Timeout ]
# h264-high.html is marked as slow in the root expectation
webkit.org/b/215007 [ Release ] webrtc/h264-high.html [ Timeout ]
@@ -1138,7 +1136,7 @@
webkit.org/b/216763 webrtc/captureCanvas-webrtc-software-h264-high.html [ Crash Failure ]
-webkit.org/b/218221 webrtc/vp9-profile2.html [ Failure ]
+webkit.org/b/218221 webrtc/vp9-profile2.html [ Timeout Failure ]
# Release timing out.
webkit.org/b/219825 [ Release ] webrtc/sframe-keys.html [ Skip ]
@@ -2027,22 +2025,18 @@
webkit.org/b/186100 css3/color-filters/color-filter-color-property-list-item.html [ ImageOnlyFailure ]
webkit.org/b/186100 fast/hidpi/filters-turbulence.html [ ImageOnlyFailure ]
webkit.org/b/187044 webanimations/opacity-animation-yields-compositing-span.html [ Failure ]
-webkit.org/b/187064 webrtc/audio-peer-connection-g722.html
-webkit.org/b/187064 webrtc/captureCanvas-webrtc.html
+
+webkit.org/b/187064 webrtc/libwebrtc/descriptionGetters.html
webkit.org/b/187064 webrtc/datachannel/mdns-ice-candidates.html
-webkit.org/b/187064 webrtc/libwebrtc/descriptionGetters.html
-webkit.org/b/187064 webrtc/peer-connection-track-end.html
webkit.org/b/187064 webrtc/video-addTrack.html
-webkit.org/b/187064 webrtc/video-addTransceiver.html
webkit.org/b/187064 webrtc/video-disabled-black.html
-webkit.org/b/187064 webrtc/video.html
-webkit.org/b/187064 webrtc/video-mute.html
webkit.org/b/187064 webrtc/video-remote-mute.html
webkit.org/b/187064 webrtc/video-replace-muted-track.html
-webkit.org/b/187064 webrtc/video-replace-track.html
webkit.org/b/187064 webrtc/video-rotation.html
-webkit.org/b/187064 webrtc/video-unmute.html
webkit.org/b/187064 webrtc/video-with-data-channel.html
+webkit.org/b/187064 webrtc/video-replace-track.html [ Failure Pass ]
+webkit.org/b/187064 imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-getStats.https.html [ Failure Pass ]
+
webkit.org/b/187271 fast/text/emoji-with-joiner.html [ Failure ]
webkit.org/b/189343 imported/w3c/web-platform-tests/xhr/sync-no-timeout.any.worker.html [ Failure ]
webkit.org/b/190707 css-custom-properties-api/length2.html [ Failure ]
Added: trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt (0 => 276491)
--- trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt (rev 0)
+++ trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt 2021-04-23 09:23:04 UTC (rev 276491)
@@ -0,0 +1,18 @@
+
+PASS Creating first data channel should fire negotiationneeded event
+PASS calling createDataChannel twice should fire negotiationneeded event once
+PASS addTransceiver() should fire negotiationneeded event
+PASS Calling addTransceiver() twice should fire negotiationneeded event once
+PASS Calling both addTransceiver() and createDataChannel() should fire negotiationneeded event once
+PASS negotiationneeded event should not fire if signaling state is not stable
+PASS negotiationneeded event should fire only after signaling state goes back to stable after setRemoteDescription
+PASS negotiationneeded event should fire only after signaling state goes back to stable after setLocalDescription
+PASS negotiationneeded event should fire only after signalingstatechange event fires from setRemoteDescription
+PASS negotiationneeded event should fire only after signalingstatechange event fires from setLocalDescription
+PASS addTrack should cause negotiationneeded to fire
+PASS removeTrack should cause negotiationneeded to fire on the caller
+PASS removeTrack should cause negotiationneeded to fire on the callee
+PASS Updating the direction of the transceiver should cause negotiationneeded to fire
+PASS Calling setStreams should cause negotiationneeded to fire
+PASS Adding two transceivers, one at a time, results in the expected number of negotiationneeded events
+
Added: trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-restartIce.https-expected.txt (0 => 276491)
--- trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-restartIce.https-expected.txt (rev 0)
+++ trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-restartIce.https-expected.txt 2021-04-23 09:23:04 UTC (rev 276491)
@@ -0,0 +1,15 @@
+
+PASS restartIce() does not trigger negotiation ahead of initial negotiation
+PASS restartIce() has no effect on initial negotiation
+PASS restartIce() fires negotiationneeded after initial negotiation
+PASS restartIce() causes fresh ufrags
+PASS restartIce() works in have-local-offer
+PASS restartIce() works in initial have-local-offer
+PASS restartIce() works in have-remote-offer
+PASS restartIce() does nothing in initial have-remote-offer
+PASS restartIce() survives remote offer
+PASS restartIce() is satisfied by remote ICE restart
+PASS restartIce() trumps {iceRestart: false}
+FAIL restartIce() survives rollback promise_test: Unhandled rejection with value: object "InvalidStateError: Local description type 3 is incompatible with current signaling state 1"
+FAIL restartIce() survives remote offer containing partial restart assert_false: Restarted 2 expected false got true
+
Added: trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setCodecPreferences-expected.txt (0 => 276491)
--- trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setCodecPreferences-expected.txt (rev 0)
+++ trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setCodecPreferences-expected.txt 2021-04-23 09:23:04 UTC (rev 276491)
@@ -0,0 +1,21 @@
+
+PASS setCodecPreferences() on audio transceiver with codecs returned from RTCRtpSender.getCapabilities('audio') should succeed
+PASS setCodecPreferences() on video transceiver with codecs returned from RTCRtpReceiver.getCapabilities('video') should succeed
+PASS setCodecPreferences() with both sender receiver codecs combined should succeed
+PASS setCodecPreferences([]) should succeed
+PASS setCodecPreferences() with reordered codecs should succeed
+PASS setCodecPreferences() with only VP8 should succeed
+PASS setCodecPreferences() with only H264 should succeed
+PASS setCodecPreferences() should allow setting H264 as first codec
+PASS setCodecPreferences() should allow setting VP8 as first codec
+PASS setCodecPreferences() on audio transceiver with codecs returned from getCapabilities('video') should throw InvalidModificationError
+PASS setCodecPreferences() with user defined codec with invalid mimeType should throw InvalidModificationError
+PASS setCodecPreferences() with user defined codec should throw InvalidModificationError
+PASS setCodecPreferences() with user defined codec together with codecs returned from getCapabilities() should throw InvalidModificationError
+PASS setCodecPreferences() with modified codec clock rate should throw InvalidModificationError
+PASS setCodecPreferences() with modified codec channel count should throw InvalidModificationError
+PASS setCodecPreferences() with modified codec parameters should throw InvalidModificationError
+PASS setCodecPreferences() with modified codecs returned from getCapabilities() should throw InvalidModificationError
+PASS setCodecPreferences() modifies the order of audio codecs in createOffer
+PASS setCodecPreferences() modifies the order of video codecs in createOffer
+
Modified: trunk/Source/ThirdParty/libwebrtc/CMakeLists.txt (276490 => 276491)
--- trunk/Source/ThirdParty/libwebrtc/CMakeLists.txt 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/ThirdParty/libwebrtc/CMakeLists.txt 2021-04-23 09:23:04 UTC (rev 276491)
@@ -538,6 +538,10 @@
Source/webrtc/common_video/h264/pps_parser.cc
Source/webrtc/common_video/h264/sps_parser.cc
Source/webrtc/common_video/h264/sps_vui_rewriter.cc
+ Source/webrtc/common_video/h265/h265_common.cc
+ Source/webrtc/common_video/h265/h265_pps_parser.cc
+ Source/webrtc/common_video/h265/h265_sps_parser.cc
+ Source/webrtc/common_video/h265/h265_vps_parser.cc
Source/webrtc/common_video/incoming_video_stream.cc
Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
Source/webrtc/common_video/video_frame_buffer.cc
@@ -1042,8 +1046,9 @@
Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
Source/webrtc/modules/video_coding/codecs/vp9/svc_config.cc
Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator.cc
+ Source/webrtc/modules/video_coding/codecs/vp9/vp9.cc
+ Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
- Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
Source/webrtc/modules/video_coding/codec_timer.cc
Source/webrtc/modules/video_coding/decoder_database.cc
Source/webrtc/modules/video_coding/decoding_state.cc
@@ -1057,6 +1062,7 @@
Source/webrtc/modules/video_coding/generic_decoder.cc
Source/webrtc/modules/video_coding/h264_sprop_parameter_sets.cc
Source/webrtc/modules/video_coding/h264_sps_pps_tracker.cc
+ Source/webrtc/modules/video_coding/h265_vps_sps_pps_tracker.cc
Source/webrtc/modules/video_coding/histogram.cc
Source/webrtc/modules/video_coding/include/video_codec_interface.cc
Source/webrtc/modules/video_coding/inter_frame_delay.cc
@@ -1608,19 +1614,25 @@
set_target_properties(webrtc PROPERTIES C_VISIBILITY_PRESET hidden)
target_compile_definitions(webrtc PRIVATE
- OPENSSL_NO_ASM
DYNAMIC_ANNOTATIONS_ENABLED=1
EXPAT_RELATIVE_PATH
+ HAVE_CONFIG_H
HAVE_LRINT
HAVE_LRINTF
HAVE_NETINET_IN_H
HAVE_SCTP
+ HAVE_STDINT_H
+ HAVE_STDLIB_H
+ HAVE_UINT64_T
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
JSON_USE_EXCEPTION=0
NON_WINDOWS_DEFINE
+ NO_MAIN_THREAD_WRAPPING
+ OPENSSL_NO_ASM
OPUS_BUILD
OPUS_EXPORT=
+ RTC_ENABLE_VP9
VAR_ARRAYS
WEBRTC_APM_DEBUG_DUMP=0
WEBRTC_CODEC_G711
@@ -1639,19 +1651,16 @@
WEBRTC_USE_BUILTIN_ISAC_FIX=1
WEBRTC_USE_BUILTIN_ISAC_FLOAT=0
WTF_USE_DYNAMIC_ANNOTATIONS=1
- RTC_DISABLE_VP9
_GNU_SOURCE
)
if (APPLE)
target_compile_definitions(webrtc PRIVATE
- RTC_ENABLE_VP9
WEBRTC_ARCH_LITTLE_ENDIAN
WEBRTC_MAC
)
else ()
target_compile_definitions(webrtc PRIVATE
- DISABLE_H265
SCTP_PROCESS_LEVEL_LOCKS
SCTP_SIMPLE_ALLOCATOR
SCTP_USE_OPENSSL_SHA1
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (276490 => 276491)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2021-04-23 09:23:04 UTC (rev 276491)
@@ -1,3 +1,14 @@
+2021-04-23 Philippe Normand <[email protected]>
+
+ [LibWebRTC][WPE][GTK] Enable vp9 and NO_MAIN_THREAD_WRAPPING
+ https://bugs.webkit.org/show_bug.cgi?id=222795
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ * CMakeLists.txt: Enable NO_MAIN_THREAD_WRAPPING as in the XCode build and harmonize options
+ between Apple and non-Apple ports.
+ * Source/webrtc/common_video/h265/h265_pps_parser.h:
+
2021-04-21 Kimmo Kinnunen <[email protected]>
WebRTC should be compiled with thread-safe statics
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_pps_parser.h (276490 => 276491)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_pps_parser.h 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_pps_parser.h 2021-04-23 09:23:04 UTC (rev 276491)
@@ -13,6 +13,10 @@
#include "absl/types/optional.h"
+#ifdef HAVE_STDINT_H
+#include <stdint.h>
+#endif
+
namespace rtc {
class BitBuffer;
}
Modified: trunk/Source/WebCore/ChangeLog (276490 => 276491)
--- trunk/Source/WebCore/ChangeLog 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/WebCore/ChangeLog 2021-04-23 09:23:04 UTC (rev 276491)
@@ -1,3 +1,35 @@
+2021-04-23 Philippe Normand <[email protected]>
+
+ [LibWebRTC][WPE][GTK] Enable vp9 and NO_MAIN_THREAD_WRAPPING
+ https://bugs.webkit.org/show_bug.cgi?id=222795
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ Add VP9 encoding and decoding support in the WPE/GTK LibWebRTC backend. The underlying
+ encoding support is based on LibWebRTC's internal VP9 support. The GStreamer VPx encoders
+ lack simulcast and high profile support, so can't really be used anyway. The
+ GStreamerVP8Encoder was removed for this reason as well.
+
+ * platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp:
+ (WebCore::VP9Decoder::VP9Decoder):
+ (WebCore::VP9Decoder::Create):
+ (WebCore::GStreamerVideoDecoderFactory::CreateVideoDecoder):
+ (WebCore::GStreamerVideoDecoderFactory::GStreamerVideoDecoderFactory):
+ (WebCore::GStreamerVideoDecoderFactory::GetSupportedFormats const):
+ * platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.h:
+ * platform/mediastream/libwebrtc/GStreamerVideoEncoder.cpp:
+ * platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp:
+ (WebCore::GStreamerVideoEncoder::makeElement):
+ (WebCore::GStreamerVideoEncoder::createEncoder):
+ (WebCore::GStreamerVideoEncoder::SetRestrictionCaps):
+ (WebCore::GStreamerVideoEncoderFactory::CreateVideoEncoder):
+ (WebCore::GStreamerVideoEncoderFactory::GStreamerVideoEncoderFactory):
+ (WebCore::GStreamerVideoEncoderFactory::GetSupportedFormats const):
+ * platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.h:
+ * platform/mediastream/libwebrtc/LibWebRTCProviderGStreamer.cpp:
+ (WebCore::LibWebRTCProviderGStreamer::createDecoderFactory):
+ (WebCore::LibWebRTCProviderGStreamer::createEncoderFactory):
+
2021-04-22 Tyler Wilcock <[email protected]>
[css-counter-styles] Parse @counter-style descriptors
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp (276490 => 276491)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp 2021-04-23 09:23:04 UTC (rev 276491)
@@ -386,6 +386,34 @@
}
};
+class VP9Decoder : public GStreamerVideoDecoder {
+public:
+ VP9Decoder(bool isSupportingVP9Profile0 = true, bool isSupportingVP9Profile2 = true)
+ : m_isSupportingVP9Profile0(isSupportingVP9Profile0)
+ , m_isSupportingVP9Profile2(isSupportingVP9Profile2) { };
+
+ const gchar* Caps() final { return "video/x-vp9"; }
+ const gchar* Name() final { return cricket::kVp9CodecName; }
+ webrtc::VideoCodecType CodecType() final { return webrtc::kVideoCodecVP9; }
+ static std::unique_ptr<webrtc::VideoDecoder> Create()
+ {
+ return std::unique_ptr<webrtc::VideoDecoder>(new VP9Decoder());
+ }
+
+ std::vector<webrtc::SdpVideoFormat> ConfigureSupportedDecoder() final
+ {
+ std::vector<webrtc::SdpVideoFormat> formats;
+ if (m_isSupportingVP9Profile0)
+ formats.push_back(webrtc::SdpVideoFormat(cricket::kVp9CodecName, { { "profile-id", "0" } }));
+ if (m_isSupportingVP9Profile2)
+ formats.push_back(webrtc::SdpVideoFormat(cricket::kVp9CodecName, { { "profile-id", "2" } }));
+ return formats;
+ }
+private:
+ bool m_isSupportingVP9Profile0;
+ bool m_isSupportingVP9Profile2;
+};
+
std::unique_ptr<webrtc::VideoDecoder> GStreamerVideoDecoderFactory::CreateVideoDecoder(const webrtc::SdpVideoFormat& format)
{
webrtc::VideoDecoder* dec;
@@ -394,6 +422,8 @@
dec = new H264Decoder();
else if (format.name == cricket::kVp8CodecName)
return VP8Decoder::Create();
+ else if (format.name == cricket::kVp9CodecName)
+ return VP9Decoder::Create();
else {
GST_ERROR("Could not create decoder for %s", format.name.c_str());
@@ -403,7 +433,9 @@
return std::unique_ptr<webrtc::VideoDecoder>(dec);
}
-GStreamerVideoDecoderFactory::GStreamerVideoDecoderFactory()
+GStreamerVideoDecoderFactory::GStreamerVideoDecoderFactory(bool isSupportingVP9Profile0, bool isSupportingVP9Profile2)
+ : m_isSupportingVP9Profile0(isSupportingVP9Profile0)
+ , m_isSupportingVP9Profile2(isSupportingVP9Profile2)
{
ensureGStreamerInitialized();
@@ -417,6 +449,7 @@
std::vector<webrtc::SdpVideoFormat> formats;
VP8Decoder().AddDecoderIfSupported(formats);
+ VP9Decoder(m_isSupportingVP9Profile0, m_isSupportingVP9Profile2).AddDecoderIfSupported(formats);
H264Decoder().AddDecoderIfSupported(formats);
return formats;
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.h (276490 => 276491)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.h 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.h 2021-04-23 09:23:04 UTC (rev 276491)
@@ -24,8 +24,7 @@
#include "LibWebRTCMacros.h"
#include "api/video_codecs/video_decoder_factory.h"
-#include <gst/gst.h>
-#include <wtf/text/WTFString.h>
+#include <wtf/Forward.h>
namespace WebCore {
@@ -32,12 +31,14 @@
class GStreamerVideoDecoderFactory : public webrtc::VideoDecoderFactory {
WTF_MAKE_FAST_ALLOCATED;
public:
- GStreamerVideoDecoderFactory();
- static bool newSource(String trackId, GstElement *source);
+ GStreamerVideoDecoderFactory(bool isSupportingVP9Profile0, bool isSupportingVP9Profile2);
private:
std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder(const webrtc::SdpVideoFormat&) final;
std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const final;
+
+ bool m_isSupportingVP9Profile0;
+ bool m_isSupportingVP9Profile2;
};
}
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoder.cpp (276490 => 276491)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoder.cpp 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoder.cpp 2021-04-23 09:23:04 UTC (rev 276491)
@@ -35,7 +35,7 @@
#define KBIT_TO_BIT 1024
static GstStaticPadTemplate sinkTemplate = GST_STATIC_PAD_TEMPLATE("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS("video/x-raw(ANY)"));
-static GstStaticPadTemplate srcTemplate = GST_STATIC_PAD_TEMPLATE("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS("video/x-h264;video/x-vp8"));
+static GstStaticPadTemplate srcTemplate = GST_STATIC_PAD_TEMPLATE("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS("video/x-h264"));
using SetBitrateFunc = Function<void(GObject* encoder, const char* propertyName, int bitrate)>;
using SetupFunc = Function<void(WebKitWebrtcVideoEncoder*)>;
@@ -51,7 +51,7 @@
const char* keyframeIntervalPropertyName;
};
-enum EncoderId { None, X264, OpenH264, OmxH264, Vp8 };
+enum EncoderId { None, X264, OpenH264, OmxH264 };
class Encoders {
public:
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp (276490 => 276491)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp 2021-04-23 09:23:04 UTC (rev 276491)
@@ -28,12 +28,13 @@
#include "GStreamerVideoFrameLibWebRTC.h"
#include "LibWebRTCWebKitMacros.h"
#include "webrtc/common_video/h264/h264_common.h"
+#include "webrtc/media/base/vp9_profile.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
+#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/modules/video_coding/utility/simulcast_utility.h"
-
#include <gst/app/gstappsink.h>
#include <gst/app/gstappsrc.h>
@@ -107,7 +108,7 @@
SetRestrictionCaps(WTFMove(caps));
if (m_encoder)
- g_object_set(m_encoder, "bitrate", parameters.bitrate.get_sum_bps(), nullptr);
+ g_object_set(m_encoder.get(), "bitrate", parameters.bitrate.get_sum_bps(), nullptr);
}
GstElement* pipeline()
@@ -119,8 +120,7 @@
{
static Atomic<uint32_t> elementId;
auto name = makeString(Name(), "-enc-", factoryName, "-", elementId.exchangeAdd(1));
- auto elem = gst_element_factory_make(factoryName, name.utf8().data());
-
+ auto* elem = gst_element_factory_make(factoryName, name.utf8().data());
return elem;
}
@@ -131,7 +131,6 @@
if (webrtc::SimulcastUtility::NumberOfSimulcastStreams(*codecSettings) > 1) {
GST_ERROR("Simulcast not supported.");
-
return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
}
@@ -138,33 +137,31 @@
m_pipeline = makeElement("pipeline");
connectSimpleBusMessageCallback(m_pipeline.get());
- auto encoder = createEncoder();
- ASSERT(encoder);
- m_encoder = encoder.get();
+ m_encoder = createEncoder();
+ ASSERT(m_encoder);
- g_object_set(m_encoder, "keyframe-interval", KeyframeInterval(codecSettings), nullptr);
+ g_object_set(m_encoder.get(), "keyframe-interval", KeyframeInterval(codecSettings), nullptr);
m_src = makeElement("appsrc");
- g_object_set(m_src, "is-live", true, "format", GST_FORMAT_TIME, nullptr);
+ g_object_set(m_src.get(), "is-live", true, "format", GST_FORMAT_TIME, nullptr);
- auto videoconvert = makeElement("videoconvert");
- auto videoscale = makeElement("videoscale");
+ auto* videoconvert = makeElement("videoconvert");
+ auto* videoscale = makeElement("videoscale");
m_sink = makeElement("appsink");
- g_object_set(m_sink, "sync", FALSE, nullptr);
+ g_object_set(m_sink.get(), "sync", FALSE, nullptr);
m_capsFilter = makeElement("capsfilter");
if (m_restrictionCaps)
- g_object_set(m_capsFilter, "caps", m_restrictionCaps.get(), nullptr);
+ g_object_set(m_capsFilter.get(), "caps", m_restrictionCaps.get(), nullptr);
- gst_bin_add_many(GST_BIN(m_pipeline.get()), m_src, videoconvert, videoscale, m_capsFilter, encoder.leakRef(), m_sink, nullptr);
- if (!gst_element_link_many(m_src, videoconvert, videoscale, m_capsFilter, m_encoder, m_sink, nullptr)) {
- GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_VERBOSE, "webkit-webrtc-encoder.error");
-
+ gst_bin_add_many(GST_BIN_CAST(m_pipeline.get()), m_src.get(), videoconvert, videoscale, m_capsFilter.get(), m_encoder.get(), m_sink.get(), nullptr);
+ if (!gst_element_link_many(m_src.get(), videoconvert, videoscale, m_capsFilter.get(), m_encoder.get(), m_sink.get(), nullptr)) {
+ GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN_CAST(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_VERBOSE, "webkit-webrtc-encoder.error");
ASSERT_NOT_REACHED();
}
gst_element_set_state(m_pipeline.get(), GST_STATE_PLAYING);
-
+ GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN_CAST(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_VERBOSE, "webkit-webrtc-encoder");
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -236,13 +233,13 @@
if (!GST_CLOCK_TIME_IS_VALID(m_firstFramePts)) {
m_firstFramePts = GST_BUFFER_PTS(buffer);
- auto pad = adoptGRef(gst_element_get_static_pad(m_src, "src"));
+ auto pad = adoptGRef(gst_element_get_static_pad(m_src.get(), "src"));
gst_pad_set_offset(pad.get(), -m_firstFramePts);
}
for (auto frame_type : *frameTypes) {
if (frame_type == webrtc::VideoFrameType::kVideoFrameKey) {
- auto pad = adoptGRef(gst_element_get_static_pad(m_src, "src"));
+ auto pad = adoptGRef(gst_element_get_static_pad(m_src.get(), "src"));
auto forceKeyUnit = gst_video_event_new_downstream_force_key_unit(GST_CLOCK_TIME_NONE,
GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, FALSE, 1);
GST_INFO_OBJECT(m_pipeline.get(), "Requesting KEYFRAME!");
@@ -254,11 +251,11 @@
}
}
- res = returnFromFlowReturn(gst_app_src_push_sample(GST_APP_SRC(m_src), sample.get()));
+ res = returnFromFlowReturn(gst_app_src_push_sample(GST_APP_SRC(m_src.get()), sample.get()));
if (res != WEBRTC_VIDEO_CODEC_OK)
return res;
- auto encodedSample = adoptGRef(gst_app_sink_try_pull_sample(GST_APP_SINK(m_sink), 5 * GST_SECOND));
+ auto encodedSample = adoptGRef(gst_app_sink_try_pull_sample(GST_APP_SINK(m_sink.get()), 5 * GST_SECOND));
if (!encodedSample) {
GST_ERROR("Didn't get any encodedSample");
return WEBRTC_VIDEO_CODEC_ERROR;
@@ -293,15 +290,13 @@
GRefPtr<GstElement> createEncoder(void)
{
+ GRefPtr<GstElement> webrtcencoder = gst_element_factory_make("webrtcvideoencoder", NULL);
+ g_object_set(webrtcencoder.get(), "format", adoptGRef(gst_caps_from_string(Caps())).get(), NULL);
+
GRefPtr<GstElement> encoder = nullptr;
- GstElement* webrtcencoder = GST_ELEMENT(g_object_ref_sink(gst_element_factory_make("webrtcvideoencoder", NULL)));
-
- g_object_set(webrtcencoder, "format", adoptGRef(gst_caps_from_string(Caps())).get(), NULL);
- g_object_get(webrtcencoder, "encoder", &encoder.outPtr(), NULL);
-
+ g_object_get(webrtcencoder.get(), "encoder", &encoder.outPtr(), NULL);
if (!encoder) {
GST_INFO("No encoder found for %s", Caps());
-
return nullptr;
}
@@ -334,7 +329,7 @@
void SetRestrictionCaps(GRefPtr<GstCaps> caps)
{
if (m_restrictionCaps)
- g_object_set(m_capsFilter, "caps", m_restrictionCaps.get(), nullptr);
+ g_object_set(m_capsFilter.get(), "caps", m_restrictionCaps.get(), nullptr);
m_restrictionCaps = caps;
}
@@ -341,9 +336,9 @@
private:
GRefPtr<GstElement> m_pipeline;
- GstElement* m_src;
- GstElement* m_encoder;
- GstElement* m_capsFilter;
+ GRefPtr<GstElement> m_src;
+ GRefPtr<GstElement> m_encoder;
+ GRefPtr<GstElement> m_capsFilter;
webrtc::EncodedImageCallback* m_imageReadyCb;
GstClockTime m_firstFramePts;
@@ -351,7 +346,7 @@
webrtc::EncodedImage m_encodedFrame;
Lock m_bufferMapLock;
- GstElement* m_sink;
+ GRefPtr<GstElement> m_sink;
};
class GStreamerH264Encoder : public GStreamerVideoEncoder {
@@ -393,42 +388,17 @@
webrtc::H264PacketizationMode packetizationMode;
};
-class GStreamerVP8Encoder : public GStreamerVideoEncoder {
-public:
- GStreamerVP8Encoder() { }
- GStreamerVP8Encoder(const webrtc::SdpVideoFormat&) { }
- const gchar* Caps() final { return "video/x-vp8"; }
- const gchar* Name() final { return cricket::kVp8CodecName; }
- webrtc::VideoCodecType CodecType() final { return webrtc::kVideoCodecVP8; }
-
- int KeyframeInterval(const webrtc::VideoCodec* codecSettings) final
- {
- return codecSettings->VP8().keyFrameInterval;
+std::unique_ptr<webrtc::VideoEncoder> GStreamerVideoEncoderFactory::CreateVideoEncoder(const webrtc::SdpVideoFormat& format)
+{
+ // FIXME: vpxenc doesn't support simulcast nor SVC. vp9enc supports only profile 0. These
+ // shortcomings trigger webrtc/vp9.html and webrtc/simulcast-h264.html timeouts and most likely
+ // bad UX in WPE/GTK browsers. So for now we prefer to use LibWebRTC's VPx encoders.
+ if (format.name == cricket::kVp9CodecName) {
+ GST_INFO("Using VP9 Encoder from LibWebRTC.");
+ return webrtc::VP9Encoder::Create(cricket::VideoCodec(format));
}
- void PopulateCodecSpecific(webrtc::CodecSpecificInfo* codecSpecificInfos, const GstBuffer* buffer) final
- {
- codecSpecificInfos->codecType = webrtc::kVideoCodecVP8;
- webrtc::CodecSpecificInfoVP8* vp8Info = &(codecSpecificInfos->codecSpecific.VP8);
- vp8Info->temporalIdx = 0;
-
- vp8Info->keyIdx = webrtc::kNoKeyIdx;
- vp8Info->nonReference = GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT);
- }
-};
-
-std::unique_ptr<webrtc::VideoEncoder> GStreamerVideoEncoderFactory::CreateVideoEncoder(const webrtc::SdpVideoFormat& format)
-{
if (format.name == cricket::kVp8CodecName) {
- GRefPtr<GstElement> webrtcencoder = adoptGRef(GST_ELEMENT(g_object_ref_sink(gst_element_factory_make("webrtcvideoencoder", NULL))));
- GRefPtr<GstElement> encoder = nullptr;
-
- g_object_set(webrtcencoder.get(), "format", adoptGRef(gst_caps_from_string("video/x-vp8")).get(), NULL);
- g_object_get(webrtcencoder.get(), "encoder", &encoder.outPtr(), NULL);
-
- if (encoder)
- return makeUnique<GStreamerVP8Encoder>(format);
-
GST_INFO("Using VP8 Encoder from LibWebRTC.");
return makeUniqueWithoutFastMallocCheck<webrtc::LibvpxVp8Encoder>(webrtc::LibvpxInterface::CreateEncoder(), webrtc::VP8Encoder::Settings());
}
@@ -446,7 +416,9 @@
return nullptr;
}
-GStreamerVideoEncoderFactory::GStreamerVideoEncoderFactory()
+GStreamerVideoEncoderFactory::GStreamerVideoEncoderFactory(bool isSupportingVP9Profile0, bool isSupportingVP9Profile2)
+ : m_isSupportingVP9Profile0(isSupportingVP9Profile0)
+ , m_isSupportingVP9Profile2(isSupportingVP9Profile2)
{
ensureGStreamerInitialized();
@@ -464,6 +436,10 @@
std::vector<webrtc::SdpVideoFormat> supportedCodecs;
supportedCodecs.push_back(webrtc::SdpVideoFormat(cricket::kVp8CodecName));
+ if (m_isSupportingVP9Profile0)
+ supportedCodecs.push_back(webrtc::SdpVideoFormat(cricket::kVp9CodecName, {{"profile-id", "0"}}));
+ if (m_isSupportingVP9Profile2)
+ supportedCodecs.push_back(webrtc::SdpVideoFormat(cricket::kVp9CodecName, {{"profile-id", "2"}}));
// If OpenH264 is present, prefer it over the GStreamer encoders (x264enc, usually).
#if WEBKIT_LIBWEBRTC_OPENH264_ENCODER
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.h (276490 => 276491)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.h 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.h 2021-04-23 09:23:04 UTC (rev 276491)
@@ -23,7 +23,6 @@
#if USE(LIBWEBRTC) && USE(GSTREAMER)
#include "LibWebRTCMacros.h"
#include "api/video_codecs/video_encoder_factory.h"
-#include <gst/gst.h>
namespace WebCore {
@@ -30,7 +29,7 @@
class GStreamerVideoEncoderFactory final : public webrtc::VideoEncoderFactory {
WTF_MAKE_FAST_ALLOCATED;
public:
- GStreamerVideoEncoderFactory();
+ GStreamerVideoEncoderFactory(bool isSupportingVP9Profile0, bool isSupportingVP9Profile2);
private:
std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override;
@@ -37,10 +36,13 @@
std::unique_ptr<webrtc::VideoEncoder> CreateVideoEncoder(const webrtc::SdpVideoFormat&) final;
CodecInfo QueryVideoEncoder(const webrtc::SdpVideoFormat&) const override
{
- GST_FIXME("Detect whether the decoder is HW accelerated");
+ // FIXME: Detect whether the decoder is HW accelerated.
return { false };
}
+
+ bool m_isSupportingVP9Profile0;
+ bool m_isSupportingVP9Profile2;
};
}
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProviderGStreamer.cpp (276490 => 276491)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProviderGStreamer.cpp 2021-04-23 07:16:49 UTC (rev 276490)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProviderGStreamer.cpp 2021-04-23 09:23:04 UTC (rev 276491)
@@ -28,6 +28,7 @@
#if USE(LIBWEBRTC) && USE(GSTREAMER)
+#include "RuntimeEnabledFeatures.h"
#include <wtf/UniqueRef.h>
namespace WebCore {
@@ -44,12 +45,12 @@
std::unique_ptr<webrtc::VideoDecoderFactory> LibWebRTCProviderGStreamer::createDecoderFactory()
{
- return makeUnique<GStreamerVideoDecoderFactory>();
+ return makeUnique<GStreamerVideoDecoderFactory>(isSupportingVP9Profile0(), isSupportingVP9Profile2());
}
std::unique_ptr<webrtc::VideoEncoderFactory> LibWebRTCProviderGStreamer::createEncoderFactory()
{
- return makeUnique<GStreamerVideoEncoderFactory>();
+ return makeUnique<GStreamerVideoEncoderFactory>(isSupportingVP9Profile0(), isSupportingVP9Profile2());
}
} // namespace WebCore
