Repo created

TMessagesProj/jni/voip/webrtc/common_video/include/bitrate_adjuster.h

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+/*
+ *  Copyright 2016 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 COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
+#define COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/rtc_export.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Certain hardware encoders tend to consistently overshoot the bitrate that
+// they are configured to encode at. This class estimates an adjusted bitrate
+// that when set on the encoder will produce the desired bitrate.
+class RTC_EXPORT BitrateAdjuster {
+ public:
+  // min_adjusted_bitrate_pct and max_adjusted_bitrate_pct are the lower and
+  // upper bound outputted adjusted bitrates as a percentage of the target
+  // bitrate.
+  BitrateAdjuster(float min_adjusted_bitrate_pct,
+                  float max_adjusted_bitrate_pct);
+  virtual ~BitrateAdjuster() {}
+
+  static const uint32_t kBitrateUpdateIntervalMs;
+  static const uint32_t kBitrateUpdateFrameInterval;
+  static const float kBitrateTolerancePct;
+  static const float kBytesPerMsToBitsPerSecond;
+
+  // Sets the desired bitrate in bps (bits per second).
+  // Should be called at least once before Update.
+  void SetTargetBitrateBps(uint32_t bitrate_bps);
+  uint32_t GetTargetBitrateBps() const;
+
+  // Returns the adjusted bitrate in bps.
+  uint32_t GetAdjustedBitrateBps() const;
+
+  // Returns what we think the current bitrate is.
+  absl::optional<uint32_t> GetEstimatedBitrateBps();
+
+  // This should be called after each frame is encoded. The timestamp at which
+  // it is called is used to estimate the output bitrate of the encoder.
+  // Should be called from only one thread.
+  void Update(size_t frame_size);
+
+ private:
+  // Returns true if the bitrate is within kBitrateTolerancePct of bitrate_bps.
+  bool IsWithinTolerance(uint32_t bitrate_bps, uint32_t target_bitrate_bps);
+
+  // Returns smallest possible adjusted value.
+  uint32_t GetMinAdjustedBitrateBps() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Returns largest possible adjusted value.
+  uint32_t GetMaxAdjustedBitrateBps() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  void Reset();
+  void UpdateBitrate(uint32_t current_time_ms)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  mutable Mutex mutex_;
+  const float min_adjusted_bitrate_pct_;
+  const float max_adjusted_bitrate_pct_;
+  // The bitrate we want.
+  volatile uint32_t target_bitrate_bps_ RTC_GUARDED_BY(mutex_);
+  // The bitrate we use to get what we want.
+  volatile uint32_t adjusted_bitrate_bps_ RTC_GUARDED_BY(mutex_);
+  // The target bitrate that the adjusted bitrate was computed from.
+  volatile uint32_t last_adjusted_target_bitrate_bps_ RTC_GUARDED_BY(mutex_);
+  // Used to estimate bitrate.
+  RateStatistics bitrate_tracker_ RTC_GUARDED_BY(mutex_);
+  // The last time we tried to adjust the bitrate.
+  uint32_t last_bitrate_update_time_ms_ RTC_GUARDED_BY(mutex_);
+  // The number of frames since the last time we tried to adjust the bitrate.
+  uint32_t frames_since_last_update_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_

TMessagesProj/jni/voip/webrtc/common_video/include/quality_limitation_reason.h

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+/*
+ *  Copyright 2019 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 COMMON_VIDEO_INCLUDE_QUALITY_LIMITATION_REASON_H_
+#define COMMON_VIDEO_INCLUDE_QUALITY_LIMITATION_REASON_H_
+
+namespace webrtc {
+
+// https://w3c.github.io/webrtc-stats/#rtcqualitylimitationreason-enum
+enum class QualityLimitationReason {
+  kNone,
+  kCpu,
+  kBandwidth,
+  kOther,
+};
+
+}  // namespace webrtc
+
+#endif  // COMMON_VIDEO_INCLUDE_QUALITY_LIMITATION_REASON_H_

TMessagesProj/jni/voip/webrtc/common_video/include/video_frame_buffer.h

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+/*
+ *  Copyright (c) 2015 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 COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
+#define COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
+
+#include <stdint.h>
+
+#include <functional>
+
+#include "api/scoped_refptr.h"
+#include "api/video/video_frame_buffer.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<I420BufferInterface> WrapI420Buffer(
+    int width,
+    int height,
+    const uint8_t* y_plane,
+    int y_stride,
+    const uint8_t* u_plane,
+    int u_stride,
+    const uint8_t* v_plane,
+    int v_stride,
+    std::function<void()> no_longer_used);
+
+rtc::scoped_refptr<I422BufferInterface> WrapI422Buffer(
+    int width,
+    int height,
+    const uint8_t* y_plane,
+    int y_stride,
+    const uint8_t* u_plane,
+    int u_stride,
+    const uint8_t* v_plane,
+    int v_stride,
+    std::function<void()> no_longer_used);
+
+rtc::scoped_refptr<I444BufferInterface> WrapI444Buffer(
+    int width,
+    int height,
+    const uint8_t* y_plane,
+    int y_stride,
+    const uint8_t* u_plane,
+    int u_stride,
+    const uint8_t* v_plane,
+    int v_stride,
+    std::function<void()> no_longer_used);
+
+rtc::scoped_refptr<I420ABufferInterface> WrapI420ABuffer(
+    int width,
+    int height,
+    const uint8_t* y_plane,
+    int y_stride,
+    const uint8_t* u_plane,
+    int u_stride,
+    const uint8_t* v_plane,
+    int v_stride,
+    const uint8_t* a_plane,
+    int a_stride,
+    std::function<void()> no_longer_used);
+
+rtc::scoped_refptr<PlanarYuvBuffer> WrapYuvBuffer(
+    VideoFrameBuffer::Type type,
+    int width,
+    int height,
+    const uint8_t* y_plane,
+    int y_stride,
+    const uint8_t* u_plane,
+    int u_stride,
+    const uint8_t* v_plane,
+    int v_stride,
+    std::function<void()> no_longer_used);
+
+rtc::scoped_refptr<I010BufferInterface> WrapI010Buffer(
+    int width,
+    int height,
+    const uint16_t* y_plane,
+    int y_stride,
+    const uint16_t* u_plane,
+    int u_stride,
+    const uint16_t* v_plane,
+    int v_stride,
+    std::function<void()> no_longer_used);
+
+rtc::scoped_refptr<I210BufferInterface> WrapI210Buffer(
+    int width,
+    int height,
+    const uint16_t* y_plane,
+    int y_stride,
+    const uint16_t* u_plane,
+    int u_stride,
+    const uint16_t* v_plane,
+    int v_stride,
+    std::function<void()> no_longer_used);
+
+rtc::scoped_refptr<I410BufferInterface> WrapI410Buffer(
+    int width,
+    int height,
+    const uint16_t* y_plane,
+    int y_stride,
+    const uint16_t* u_plane,
+    int u_stride,
+    const uint16_t* v_plane,
+    int v_stride,
+    std::function<void()> no_longer_used);
+}  // namespace webrtc
+
+#endif  // COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_

TMessagesProj/jni/voip/webrtc/common_video/include/video_frame_buffer_pool.h

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+/*
+ *  Copyright (c) 2015 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 COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_POOL_H_
+#define COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_POOL_H_
+
+#include <stddef.h>
+
+#include <list>
+
+#include "api/scoped_refptr.h"
+#include "api/video/i010_buffer.h"
+#include "api/video/i210_buffer.h"
+#include "api/video/i410_buffer.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/i422_buffer.h"
+#include "api/video/i444_buffer.h"
+#include "api/video/nv12_buffer.h"
+#include "rtc_base/race_checker.h"
+
+namespace webrtc {
+
+// Simple buffer pool to avoid unnecessary allocations of video frame buffers.
+// The pool manages the memory of the I420Buffer/NV12Buffer returned from
+// Create(I420|NV12)Buffer. When the buffer is destructed, the memory is
+// returned to the pool for use by subsequent calls to Create(I420|NV12)Buffer.
+// If the resolution passed to Create(I420|NV12)Buffer changes or requested
+// pixel format changes, old buffers will be purged from the pool.
+// Note that Create(I420|NV12)Buffer will crash if more than
+// kMaxNumberOfFramesBeforeCrash are created. This is to prevent memory leaks
+// where frames are not returned.
+class VideoFrameBufferPool {
+ public:
+  VideoFrameBufferPool();
+  explicit VideoFrameBufferPool(bool zero_initialize);
+  VideoFrameBufferPool(bool zero_initialize, size_t max_number_of_buffers);
+  ~VideoFrameBufferPool();
+
+  // Returns a buffer from the pool. If no suitable buffer exist in the pool
+  // and there are less than `max_number_of_buffers` pending, a buffer is
+  // created. Returns null otherwise.
+  rtc::scoped_refptr<I420Buffer> CreateI420Buffer(int width, int height);
+  rtc::scoped_refptr<I422Buffer> CreateI422Buffer(int width, int height);
+  rtc::scoped_refptr<I444Buffer> CreateI444Buffer(int width, int height);
+  rtc::scoped_refptr<I010Buffer> CreateI010Buffer(int width, int height);
+  rtc::scoped_refptr<I210Buffer> CreateI210Buffer(int width, int height);
+  rtc::scoped_refptr<I410Buffer> CreateI410Buffer(int width, int height);
+  rtc::scoped_refptr<NV12Buffer> CreateNV12Buffer(int width, int height);
+
+  // Changes the max amount of buffers in the pool to the new value.
+  // Returns true if change was successful and false if the amount of already
+  // allocated buffers is bigger than new value.
+  bool Resize(size_t max_number_of_buffers);
+
+  // Clears buffers_ and detaches the thread checker so that it can be reused
+  // later from another thread.
+  void Release();
+
+ private:
+  rtc::scoped_refptr<VideoFrameBuffer>
+  GetExistingBuffer(int width, int height, VideoFrameBuffer::Type type);
+
+  rtc::RaceChecker race_checker_;
+  std::list<rtc::scoped_refptr<VideoFrameBuffer>> buffers_;
+  // If true, newly allocated buffers are zero-initialized. Note that recycled
+  // buffers are not zero'd before reuse. This is required of buffers used by
+  // FFmpeg according to http://crbug.com/390941, which only requires it for the
+  // initial allocation (as shown by FFmpeg's own buffer allocation code). It
+  // has to do with "Use-of-uninitialized-value" on "Linux_msan_chrome".
+  const bool zero_initialize_;
+  // Max number of buffers this pool can have pending.
+  size_t max_number_of_buffers_;
+};
+
+}  // namespace webrtc
+
+#endif  // COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_POOL_H_