webrtc_m130/webrtc/video/video_capture_input.cc
Peter Boström a4c76882b9 Move encoder thread to VideoSendStream.
Makes VideoCaptureInput easier to test and enables running more things
outside VideoCaptureInput on the encoder thread in the future
(initializing encoders and reconfiguring them, for instance).

BUG=webrtc:5410, webrtc:5494
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1763693002 .

Cr-Commit-Position: refs/heads/master@{#11860}
2016-03-03 15:29:09 +00:00

109 lines
4.0 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/video/video_capture_input.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_capture/video_capture_factory.h"
#include "webrtc/modules/video_processing/include/video_processing.h"
#include "webrtc/modules/video_render/video_render_defines.h"
#include "webrtc/video/overuse_frame_detector.h"
#include "webrtc/video/send_statistics_proxy.h"
#include "webrtc/video/vie_encoder.h"
namespace webrtc {
namespace internal {
VideoCaptureInput::VideoCaptureInput(rtc::Event* capture_event,
VideoRenderer* local_renderer,
SendStatisticsProxy* stats_proxy,
OveruseFrameDetector* overuse_detector)
: local_renderer_(local_renderer),
stats_proxy_(stats_proxy),
capture_event_(capture_event),
// TODO(danilchap): Pass clock from outside to ensure it is same clock
// rtcp module use to calculate offset since last frame captured
// to estimate rtp timestamp for SenderReport.
clock_(Clock::GetRealTimeClock()),
last_captured_timestamp_(0),
delta_ntp_internal_ms_(clock_->CurrentNtpInMilliseconds() -
clock_->TimeInMilliseconds()),
overuse_detector_(overuse_detector) {}
VideoCaptureInput::~VideoCaptureInput() {
}
void VideoCaptureInput::IncomingCapturedFrame(const VideoFrame& video_frame) {
// TODO(pbos): Remove local rendering, it should be handled by the client code
// if required.
if (local_renderer_)
local_renderer_->RenderFrame(video_frame, 0);
stats_proxy_->OnIncomingFrame(video_frame.width(), video_frame.height());
VideoFrame incoming_frame = video_frame;
// Local time in webrtc time base.
int64_t current_time = clock_->TimeInMilliseconds();
incoming_frame.set_render_time_ms(current_time);
// Capture time may come from clock with an offset and drift from clock_.
int64_t capture_ntp_time_ms;
if (video_frame.ntp_time_ms() != 0) {
capture_ntp_time_ms = video_frame.ntp_time_ms();
} else if (video_frame.render_time_ms() != 0) {
capture_ntp_time_ms = video_frame.render_time_ms() + delta_ntp_internal_ms_;
} else {
capture_ntp_time_ms = current_time + delta_ntp_internal_ms_;
}
incoming_frame.set_ntp_time_ms(capture_ntp_time_ms);
// Convert NTP time, in ms, to RTP timestamp.
const int kMsToRtpTimestamp = 90;
incoming_frame.set_timestamp(
kMsToRtpTimestamp * static_cast<uint32_t>(incoming_frame.ntp_time_ms()));
rtc::CritScope lock(&crit_);
if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) {
// We don't allow the same capture time for two frames, drop this one.
LOG(LS_WARNING) << "Same/old NTP timestamp ("
<< incoming_frame.ntp_time_ms()
<< " <= " << last_captured_timestamp_
<< ") for incoming frame. Dropping.";
return;
}
captured_frame_.ShallowCopy(incoming_frame);
last_captured_timestamp_ = incoming_frame.ntp_time_ms();
overuse_detector_->FrameCaptured(captured_frame_);
TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", video_frame.render_time_ms(),
"render_time", video_frame.render_time_ms());
capture_event_->Set();
}
bool VideoCaptureInput::GetVideoFrame(VideoFrame* video_frame) {
rtc::CritScope lock(&crit_);
if (captured_frame_.IsZeroSize())
return false;
*video_frame = captured_frame_;
captured_frame_.Reset();
return true;
}
} // namespace internal
} // namespace webrtc