Repo created

TMessagesProj/jni/voip/webrtc/video/render/incoming_video_stream.cc

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+/*
+ *  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 "video/render/incoming_video_stream.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+#include "video/render/video_render_frames.h"
+
+namespace webrtc {
+
+IncomingVideoStream::IncomingVideoStream(
+    TaskQueueFactory* task_queue_factory,
+    int32_t delay_ms,
+    rtc::VideoSinkInterface<VideoFrame>* callback)
+    : render_buffers_(delay_ms),
+      callback_(callback),
+      incoming_render_queue_(task_queue_factory->CreateTaskQueue(
+          "IncomingVideoStream",
+          TaskQueueFactory::Priority::HIGH)) {}
+
+IncomingVideoStream::~IncomingVideoStream() {
+  RTC_DCHECK(main_thread_checker_.IsCurrent());
+  // The queue must be destroyed before its pointer is invalidated to avoid race
+  // between destructor and posting task to the task queue from itself.
+  // std::unique_ptr destructor does the same two operations in reverse order as
+  // it doesn't expect member would be used after its destruction has started.
+  incoming_render_queue_.get_deleter()(incoming_render_queue_.get());
+  incoming_render_queue_.release();
+}
+
+void IncomingVideoStream::OnFrame(const VideoFrame& video_frame) {
+  TRACE_EVENT0("webrtc", "IncomingVideoStream::OnFrame");
+  RTC_CHECK_RUNS_SERIALIZED(&decoder_race_checker_);
+  RTC_DCHECK(!incoming_render_queue_->IsCurrent());
+  // TODO(srte): Using video_frame = std::move(video_frame) would move the frame
+  // into the lambda instead of copying it, but it doesn't work unless we change
+  // OnFrame to take its frame argument by value instead of const reference.
+  incoming_render_queue_->PostTask([this, video_frame = video_frame]() mutable {
+    RTC_DCHECK_RUN_ON(incoming_render_queue_.get());
+    if (render_buffers_.AddFrame(std::move(video_frame)) == 1)
+      Dequeue();
+  });
+}
+
+void IncomingVideoStream::Dequeue() {
+  TRACE_EVENT0("webrtc", "IncomingVideoStream::Dequeue");
+  RTC_DCHECK_RUN_ON(incoming_render_queue_.get());
+  absl::optional<VideoFrame> frame_to_render = render_buffers_.FrameToRender();
+  if (frame_to_render)
+    callback_->OnFrame(*frame_to_render);
+
+  if (render_buffers_.HasPendingFrames()) {
+    uint32_t wait_time = render_buffers_.TimeToNextFrameRelease();
+    incoming_render_queue_->PostDelayedHighPrecisionTask(
+        [this]() { Dequeue(); }, TimeDelta::Millis(wait_time));
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/video/render/incoming_video_stream.h

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+/*
+ *  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.
+ */
+
+#ifndef VIDEO_RENDER_INCOMING_VIDEO_STREAM_H_
+#define VIDEO_RENDER_INCOMING_VIDEO_STREAM_H_
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/thread_annotations.h"
+#include "video/render/video_render_frames.h"
+
+namespace webrtc {
+
+class IncomingVideoStream : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  IncomingVideoStream(TaskQueueFactory* task_queue_factory,
+                      int32_t delay_ms,
+                      rtc::VideoSinkInterface<VideoFrame>* callback);
+  ~IncomingVideoStream() override;
+
+ private:
+  void OnFrame(const VideoFrame& video_frame) override;
+  void Dequeue();
+
+  SequenceChecker main_thread_checker_;
+  rtc::RaceChecker decoder_race_checker_;
+
+  VideoRenderFrames render_buffers_ RTC_GUARDED_BY(incoming_render_queue_);
+  rtc::VideoSinkInterface<VideoFrame>* const callback_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> incoming_render_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // VIDEO_RENDER_INCOMING_VIDEO_STREAM_H_

TMessagesProj/jni/voip/webrtc/video/render/video_render_frames.cc

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+/*
+ *  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 "video/render/video_render_frames.h"
+
+#include <type_traits>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+// Don't render frames with timestamp older than 500ms from now.
+const int kOldRenderTimestampMS = 500;
+// Don't render frames with timestamp more than 10s into the future.
+const int kFutureRenderTimestampMS = 10000;
+
+const uint32_t kEventMaxWaitTimeMs = 200;
+const uint32_t kMinRenderDelayMs = 10;
+const uint32_t kMaxRenderDelayMs = 500;
+const size_t kMaxIncomingFramesBeforeLogged = 100;
+
+uint32_t EnsureValidRenderDelay(uint32_t render_delay) {
+  return (render_delay < kMinRenderDelayMs || render_delay > kMaxRenderDelayMs)
+             ? kMinRenderDelayMs
+             : render_delay;
+}
+}  // namespace
+
+VideoRenderFrames::VideoRenderFrames(uint32_t render_delay_ms)
+    : render_delay_ms_(EnsureValidRenderDelay(render_delay_ms)) {}
+
+VideoRenderFrames::~VideoRenderFrames() {
+  frames_dropped_ += incoming_frames_.size();
+  RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DroppedFrames.RenderQueue",
+                            frames_dropped_);
+  RTC_LOG(LS_INFO) << "WebRTC.Video.DroppedFrames.RenderQueue "
+                   << frames_dropped_;
+}
+
+int32_t VideoRenderFrames::AddFrame(VideoFrame&& new_frame) {
+  const int64_t time_now = rtc::TimeMillis();
+
+  // Drop old frames only when there are other frames in the queue, otherwise, a
+  // really slow system never renders any frames.
+  if (!incoming_frames_.empty() &&
+      new_frame.render_time_ms() + kOldRenderTimestampMS < time_now) {
+    RTC_LOG(LS_WARNING) << "Too old frame, timestamp=" << new_frame.timestamp();
+    ++frames_dropped_;
+    return -1;
+  }
+
+  if (new_frame.render_time_ms() > time_now + kFutureRenderTimestampMS) {
+    RTC_LOG(LS_WARNING) << "Frame too long into the future, timestamp="
+                        << new_frame.timestamp();
+    ++frames_dropped_;
+    return -1;
+  }
+
+  if (new_frame.render_time_ms() < last_render_time_ms_) {
+    RTC_LOG(LS_WARNING) << "Frame scheduled out of order, render_time="
+                        << new_frame.render_time_ms()
+                        << ", latest=" << last_render_time_ms_;
+    // For more details, see bug:
+    // https://bugs.chromium.org/p/webrtc/issues/detail?id=7253
+    ++frames_dropped_;
+    return -1;
+  }
+
+  last_render_time_ms_ = new_frame.render_time_ms();
+  incoming_frames_.emplace_back(std::move(new_frame));
+
+  if (incoming_frames_.size() > kMaxIncomingFramesBeforeLogged) {
+    RTC_LOG(LS_WARNING) << "Stored incoming frames: "
+                        << incoming_frames_.size();
+  }
+  return static_cast<int32_t>(incoming_frames_.size());
+}
+
+absl::optional<VideoFrame> VideoRenderFrames::FrameToRender() {
+  absl::optional<VideoFrame> render_frame;
+  // Get the newest frame that can be released for rendering.
+  while (!incoming_frames_.empty() && TimeToNextFrameRelease() <= 0) {
+    if (render_frame) {
+      ++frames_dropped_;
+    }
+    render_frame = std::move(incoming_frames_.front());
+    incoming_frames_.pop_front();
+  }
+  return render_frame;
+}
+
+uint32_t VideoRenderFrames::TimeToNextFrameRelease() {
+  if (incoming_frames_.empty()) {
+    return kEventMaxWaitTimeMs;
+  }
+  const int64_t time_to_release = incoming_frames_.front().render_time_ms() -
+                                  render_delay_ms_ - rtc::TimeMillis();
+  return time_to_release < 0 ? 0u : static_cast<uint32_t>(time_to_release);
+}
+
+bool VideoRenderFrames::HasPendingFrames() const {
+  return !incoming_frames_.empty();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/video/render/video_render_frames.h

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+/*
+ *  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.
+ */
+
+#ifndef VIDEO_RENDER_VIDEO_RENDER_FRAMES_H_
+#define VIDEO_RENDER_VIDEO_RENDER_FRAMES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+
+#include "absl/types/optional.h"
+#include "api/video/video_frame.h"
+
+namespace webrtc {
+
+// Class definitions
+class VideoRenderFrames {
+ public:
+  explicit VideoRenderFrames(uint32_t render_delay_ms);
+  VideoRenderFrames(const VideoRenderFrames&) = delete;
+  ~VideoRenderFrames();
+
+  // Add a frame to the render queue
+  int32_t AddFrame(VideoFrame&& new_frame);
+
+  // Get a frame for rendering, or false if it's not time to render.
+  absl::optional<VideoFrame> FrameToRender();
+
+  // Returns the number of ms to next frame to render
+  uint32_t TimeToNextFrameRelease();
+
+  bool HasPendingFrames() const;
+
+ private:
+  // Sorted list with framed to be rendered, oldest first.
+  std::list<VideoFrame> incoming_frames_;
+
+  // Estimated delay from a frame is released until it's rendered.
+  const uint32_t render_delay_ms_;
+
+  int64_t last_render_time_ms_ = 0;
+  size_t frames_dropped_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // VIDEO_RENDER_VIDEO_RENDER_FRAMES_H_