Index: webrtc/video/video_capture_input.cc |
diff --git a/webrtc/video/video_capture_input.cc b/webrtc/video/video_capture_input.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..8f574e21154d7abddca59dcf46b28d476866d5dc |
--- /dev/null |
+++ b/webrtc/video/video_capture_input.cc |
@@ -0,0 +1,109 @@ |
+/* |
+ * 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 |