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| 1 /* | |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #include "webrtc/video/video_capture_input.h" | |
| 12 | |
| 13 #include "webrtc/base/checks.h" | |
| 14 #include "webrtc/base/logging.h" | |
| 15 #include "webrtc/base/trace_event.h" | |
| 16 #include "webrtc/modules/include/module_common_types.h" | |
| 17 #include "webrtc/modules/video_capture/video_capture_factory.h" | |
| 18 #include "webrtc/modules/video_processing/include/video_processing.h" | |
| 19 #include "webrtc/video/overuse_frame_detector.h" | |
| 20 #include "webrtc/video/send_statistics_proxy.h" | |
| 21 #include "webrtc/video/vie_encoder.h" | |
| 22 | |
| 23 namespace webrtc { | |
| 24 | |
| 25 namespace internal { | |
| 26 VideoCaptureInput::VideoCaptureInput( | |
| 27 rtc::Event* capture_event, | |
| 28 rtc::VideoSinkInterface<VideoFrame>* local_renderer, | |
| 29 SendStatisticsProxy* stats_proxy, | |
| 30 OveruseFrameDetector* overuse_detector) | |
| 31 : local_renderer_(local_renderer), | |
| 32 stats_proxy_(stats_proxy), | |
| 33 capture_event_(capture_event), | |
| 34 // TODO(danilchap): Pass clock from outside to ensure it is same clock | |
| 35 // rtcp module use to calculate offset since last frame captured | |
| 36 // to estimate rtp timestamp for SenderReport. | |
| 37 clock_(Clock::GetRealTimeClock()), | |
| 38 last_captured_timestamp_(0), | |
| 39 delta_ntp_internal_ms_(clock_->CurrentNtpInMilliseconds() - | |
| 40 clock_->TimeInMilliseconds()), | |
| 41 overuse_detector_(overuse_detector) {} | |
| 42 | |
| 43 VideoCaptureInput::~VideoCaptureInput() { | |
| 44 } | |
| 45 | |
| 46 void VideoCaptureInput::IncomingCapturedFrame(const VideoFrame& video_frame) { | |
| 47 // TODO(pbos): Remove local rendering, it should be handled by the client code | |
| 48 // if required. | |
| 49 if (local_renderer_) | |
| 50 local_renderer_->OnFrame(video_frame); | |
| 51 | |
| 52 stats_proxy_->OnIncomingFrame(video_frame.width(), video_frame.height()); | |
| 53 | |
| 54 VideoFrame incoming_frame = video_frame; | |
| 55 | |
| 56 // Local time in webrtc time base. | |
| 57 int64_t current_time = clock_->TimeInMilliseconds(); | |
| 58 incoming_frame.set_render_time_ms(current_time); | |
| 59 | |
| 60 // Capture time may come from clock with an offset and drift from clock_. | |
| 61 int64_t capture_ntp_time_ms; | |
| 62 if (video_frame.ntp_time_ms() != 0) { | |
| 63 capture_ntp_time_ms = video_frame.ntp_time_ms(); | |
| 64 } else if (video_frame.render_time_ms() != 0) { | |
| 65 capture_ntp_time_ms = video_frame.render_time_ms() + delta_ntp_internal_ms_; | |
| 66 } else { | |
| 67 capture_ntp_time_ms = current_time + delta_ntp_internal_ms_; | |
| 68 } | |
| 69 incoming_frame.set_ntp_time_ms(capture_ntp_time_ms); | |
| 70 | |
| 71 // Convert NTP time, in ms, to RTP timestamp. | |
| 72 const int kMsToRtpTimestamp = 90; | |
| 73 incoming_frame.set_timestamp( | |
| 74 kMsToRtpTimestamp * static_cast<uint32_t>(incoming_frame.ntp_time_ms())); | |
| 75 | |
| 76 rtc::CritScope lock(&crit_); | |
| 77 if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) { | |
| 78 // We don't allow the same capture time for two frames, drop this one. | |
| 79 LOG(LS_WARNING) << "Same/old NTP timestamp (" | |
| 80 << incoming_frame.ntp_time_ms() | |
| 81 << " <= " << last_captured_timestamp_ | |
| 82 << ") for incoming frame. Dropping."; | |
| 83 return; | |
| 84 } | |
| 85 | |
| 86 captured_frame_.reset(new VideoFrame); | |
| 87 captured_frame_->ShallowCopy(incoming_frame); | |
| 88 last_captured_timestamp_ = incoming_frame.ntp_time_ms(); | |
| 89 | |
| 90 overuse_detector_->FrameCaptured(*captured_frame_); | |
| 91 | |
| 92 TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", video_frame.render_time_ms(), | |
| 93 "render_time", video_frame.render_time_ms()); | |
| 94 | |
| 95 capture_event_->Set(); | |
| 96 } | |
| 97 | |
| 98 bool VideoCaptureInput::GetVideoFrame(VideoFrame* video_frame) { | |
| 99 rtc::CritScope lock(&crit_); | |
| 100 if (!captured_frame_) | |
| 101 return false; | |
| 102 | |
| 103 *video_frame = *captured_frame_; | |
| 104 captured_frame_.reset(); | |
| 105 return true; | |
| 106 } | |
| 107 | |
| 108 } // namespace internal | |
| 109 } // namespace webrtc | |
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Chromium Code Reviews
Issue 2060403002:
Add task queue to Call. (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@move_getpadding