Revert of Add task queue to Call.

webrtc/video/video_capture_input.cc

+/*
+ *  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/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,
+    rtc::VideoSinkInterface<VideoFrame>* 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_->OnFrame(video_frame);
+
+  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_.reset(new VideoFrame);
+  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_)
+    return false;
+
+  *video_frame = *captured_frame_;
+  captured_frame_.reset();
+  return true;
+}
+
+}  // namespace internal
+}  // namespace webrtc