8eb37a39e7
Reason for revert: Failed on Win 10 Chrome FYI. https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio # # Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138 # last system error: 87 # Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295) # WebRtcBrowserTest # Original issue's description: > - Add task queue to Call with the intent of replacing the use of one of the process threads. > > - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue. > > - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread. > > - VideoEncoderConfig and VideoSendStream::Config support move semantics. > > - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames. > > BUG=webrtc:5687 > > Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473 > Cr-Commit-Position: refs/heads/master@{#13767} TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5687 Review-Url: https://codereview.webrtc.org/2248713003 Cr-Commit-Position: refs/heads/master@{#13774}
110 lines
3.9 KiB
C++
110 lines
3.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video/video_capture_input.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/trace_event.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_capture/video_capture_factory.h"
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#include "webrtc/modules/video_processing/include/video_processing.h"
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#include "webrtc/video/overuse_frame_detector.h"
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#include "webrtc/video/send_statistics_proxy.h"
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#include "webrtc/video/vie_encoder.h"
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namespace webrtc {
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namespace internal {
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VideoCaptureInput::VideoCaptureInput(
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rtc::Event* capture_event,
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rtc::VideoSinkInterface<VideoFrame>* local_renderer,
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SendStatisticsProxy* stats_proxy,
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OveruseFrameDetector* overuse_detector)
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: local_renderer_(local_renderer),
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stats_proxy_(stats_proxy),
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capture_event_(capture_event),
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// TODO(danilchap): Pass clock from outside to ensure it is same clock
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// rtcp module use to calculate offset since last frame captured
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// to estimate rtp timestamp for SenderReport.
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clock_(Clock::GetRealTimeClock()),
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last_captured_timestamp_(0),
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delta_ntp_internal_ms_(clock_->CurrentNtpInMilliseconds() -
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clock_->TimeInMilliseconds()),
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overuse_detector_(overuse_detector) {}
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VideoCaptureInput::~VideoCaptureInput() {
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}
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void VideoCaptureInput::IncomingCapturedFrame(const VideoFrame& video_frame) {
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// TODO(pbos): Remove local rendering, it should be handled by the client code
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// if required.
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if (local_renderer_)
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local_renderer_->OnFrame(video_frame);
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stats_proxy_->OnIncomingFrame(video_frame.width(), video_frame.height());
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VideoFrame incoming_frame = video_frame;
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// Local time in webrtc time base.
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int64_t current_time = clock_->TimeInMilliseconds();
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incoming_frame.set_render_time_ms(current_time);
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// Capture time may come from clock with an offset and drift from clock_.
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int64_t capture_ntp_time_ms;
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if (video_frame.ntp_time_ms() != 0) {
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capture_ntp_time_ms = video_frame.ntp_time_ms();
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} else if (video_frame.render_time_ms() != 0) {
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capture_ntp_time_ms = video_frame.render_time_ms() + delta_ntp_internal_ms_;
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} else {
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capture_ntp_time_ms = current_time + delta_ntp_internal_ms_;
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}
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incoming_frame.set_ntp_time_ms(capture_ntp_time_ms);
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// Convert NTP time, in ms, to RTP timestamp.
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const int kMsToRtpTimestamp = 90;
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incoming_frame.set_timestamp(
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kMsToRtpTimestamp * static_cast<uint32_t>(incoming_frame.ntp_time_ms()));
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rtc::CritScope lock(&crit_);
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if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) {
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// We don't allow the same capture time for two frames, drop this one.
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LOG(LS_WARNING) << "Same/old NTP timestamp ("
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<< incoming_frame.ntp_time_ms()
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<< " <= " << last_captured_timestamp_
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<< ") for incoming frame. Dropping.";
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return;
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}
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captured_frame_.reset(new VideoFrame);
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captured_frame_->ShallowCopy(incoming_frame);
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last_captured_timestamp_ = incoming_frame.ntp_time_ms();
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overuse_detector_->FrameCaptured(*captured_frame_);
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TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", video_frame.render_time_ms(),
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"render_time", video_frame.render_time_ms());
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capture_event_->Set();
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}
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bool VideoCaptureInput::GetVideoFrame(VideoFrame* video_frame) {
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rtc::CritScope lock(&crit_);
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if (!captured_frame_)
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return false;
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*video_frame = *captured_frame_;
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captured_frame_.reset();
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return true;
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}
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} // namespace internal
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} // namespace webrtc
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