1 | /* |
2 | * Copyright (C) 2017 Igalia S.L. All rights reserved. |
3 | * Copyright (C) 2017 Apple Inc. All rights reserved. |
4 | * |
5 | * Redistribution and use in source and binary forms, with or without |
6 | * modification, are permitted, provided that the following conditions |
7 | * are required to be met: |
8 | * |
9 | * 1. Redistributions of source code must retain the above copyright |
10 | * notice, this list of conditions and the following disclaimer. |
11 | * 2. Redistributions in binary form must reproduce the above copyright |
12 | * notice, this list of conditions and the following disclaimer in the |
13 | * documentation and/or other materials provided with the distribution. |
14 | * |
15 | * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY |
16 | * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
17 | * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
18 | * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR |
19 | * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL |
20 | * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR |
21 | * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER |
22 | * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, |
23 | * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE |
24 | * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
25 | */ |
26 | |
27 | |
28 | #include "config.h" |
29 | |
30 | #if USE(LIBWEBRTC) && USE(GSTREAMER) |
31 | #include "RealtimeOutgoingVideoSourceLibWebRTC.h" |
32 | |
33 | #include "GStreamerVideoFrameLibWebRTC.h" |
34 | #include "MediaSampleGStreamer.h" |
35 | |
36 | namespace WebCore { |
37 | |
38 | Ref<RealtimeOutgoingVideoSource> RealtimeOutgoingVideoSource::create(Ref<MediaStreamTrackPrivate>&& videoSource) |
39 | { |
40 | return RealtimeOutgoingVideoSourceLibWebRTC::create(WTFMove(videoSource)); |
41 | } |
42 | |
43 | Ref<RealtimeOutgoingVideoSourceLibWebRTC> RealtimeOutgoingVideoSourceLibWebRTC::create(Ref<MediaStreamTrackPrivate>&& videoSource) |
44 | { |
45 | return adoptRef(*new RealtimeOutgoingVideoSourceLibWebRTC(WTFMove(videoSource))); |
46 | } |
47 | |
48 | RealtimeOutgoingVideoSourceLibWebRTC::RealtimeOutgoingVideoSourceLibWebRTC(Ref<MediaStreamTrackPrivate>&& videoSource) |
49 | : RealtimeOutgoingVideoSource(WTFMove(videoSource)) |
50 | { |
51 | } |
52 | |
53 | void RealtimeOutgoingVideoSourceLibWebRTC::sampleBufferUpdated(MediaStreamTrackPrivate&, MediaSample& sample) |
54 | { |
55 | if (isSilenced()) |
56 | return; |
57 | |
58 | switch (sample.videoRotation()) { |
59 | case MediaSample::VideoRotation::None: |
60 | m_currentRotation = webrtc::kVideoRotation_0; |
61 | break; |
62 | case MediaSample::VideoRotation::UpsideDown: |
63 | m_currentRotation = webrtc::kVideoRotation_180; |
64 | break; |
65 | case MediaSample::VideoRotation::Right: |
66 | m_currentRotation = webrtc::kVideoRotation_90; |
67 | break; |
68 | case MediaSample::VideoRotation::Left: |
69 | m_currentRotation = webrtc::kVideoRotation_270; |
70 | break; |
71 | } |
72 | |
73 | ASSERT(sample.platformSample().type == PlatformSample::GStreamerSampleType); |
74 | auto& mediaSample = static_cast<MediaSampleGStreamer&>(sample); |
75 | auto frameBuffer(GStreamerVideoFrameLibWebRTC::create(gst_sample_ref(mediaSample.platformSample().sample.gstSample))); |
76 | |
77 | sendFrame(WTFMove(frameBuffer)); |
78 | } |
79 | |
80 | rtc::scoped_refptr<webrtc::VideoFrameBuffer> RealtimeOutgoingVideoSourceLibWebRTC::createBlackFrame(size_t width, size_t height) |
81 | { |
82 | GstVideoInfo info; |
83 | |
84 | gst_video_info_set_format(&info, GST_VIDEO_FORMAT_RGB, width, height); |
85 | |
86 | GRefPtr<GstBuffer> buffer = adoptGRef(gst_buffer_new_allocate(nullptr, info.size, nullptr)); |
87 | GRefPtr<GstCaps> caps = adoptGRef(gst_video_info_to_caps(&info)); |
88 | |
89 | auto map = GstMappedBuffer::create(buffer.get(), GST_MAP_WRITE); |
90 | memset(map->data(), 0, info.size); |
91 | |
92 | return GStreamerVideoFrameLibWebRTC::create(gst_sample_new(buffer.get(), caps.get(), NULL, NULL)); |
93 | } |
94 | |
95 | } // namespace WebCore |
96 | |
97 | #endif // USE(LIBWEBRTC) |
98 | |