Repo created

TMessagesProj/jni/voip/webrtc/modules/include/module_common_types.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 MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+#define MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+namespace webrtc {
+
+// Interface used by the CallStats class to distribute call statistics.
+// Callbacks will be triggered as soon as the class has been registered to a
+// CallStats object using RegisterStatsObserver.
+class CallStatsObserver {
+ public:
+  virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) = 0;
+
+  virtual ~CallStatsObserver() {}
+};
+
+// Interface used by NackModule and JitterBuffer.
+class NackSender {
+ public:
+  // If `buffering_allowed`, other feedback messages (e.g. key frame requests)
+  // may be added to the same outgoing feedback message. In that case, it's up
+  // to the user of the interface to ensure that when all buffer-able messages
+  // have been added, the feedback message is triggered.
+  virtual void SendNack(const std::vector<uint16_t>& sequence_numbers,
+                        bool buffering_allowed) = 0;
+
+ protected:
+  virtual ~NackSender() {}
+};
+
+// Interface used by NackModule and JitterBuffer.
+class KeyFrameRequestSender {
+ public:
+  virtual void RequestKeyFrame() = 0;
+
+ protected:
+  virtual ~KeyFrameRequestSender() {}
+};
+
+// Interface used by LossNotificationController to communicate to RtpRtcp.
+class LossNotificationSender {
+ public:
+  virtual ~LossNotificationSender() {}
+
+  virtual void SendLossNotification(uint16_t last_decoded_seq_num,
+                                    uint16_t last_received_seq_num,
+                                    bool decodability_flag,
+                                    bool buffering_allowed) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_INCLUDE_MODULE_COMMON_TYPES_H_

TMessagesProj/jni/voip/webrtc/modules/include/module_common_types_public.h

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+/*
+ *  Copyright (c) 2017 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 MODULES_INCLUDE_MODULE_COMMON_TYPES_PUBLIC_H_
+#define MODULES_INCLUDE_MODULE_COMMON_TYPES_PUBLIC_H_
+
+#include <limits>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+template <typename U>
+inline bool IsNewer(U value, U prev_value) {
+  static_assert(!std::numeric_limits<U>::is_signed, "U must be unsigned");
+  // kBreakpoint is the half-way mark for the type U. For instance, for a
+  // uint16_t it will be 0x8000, and for a uint32_t, it will be 0x8000000.
+  constexpr U kBreakpoint = (std::numeric_limits<U>::max() >> 1) + 1;
+  // Distinguish between elements that are exactly kBreakpoint apart.
+  // If t1>t2 and |t1-t2| = kBreakpoint: IsNewer(t1,t2)=true,
+  // IsNewer(t2,t1)=false
+  // rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false.
+  if (value - prev_value == kBreakpoint) {
+    return value > prev_value;
+  }
+  return value != prev_value &&
+         static_cast<U>(value - prev_value) < kBreakpoint;
+}
+
+// NB: Doesn't fulfill strict weak ordering requirements.
+//     Mustn't be used as std::map Compare function.
+inline bool IsNewerSequenceNumber(uint16_t sequence_number,
+                                  uint16_t prev_sequence_number) {
+  return IsNewer(sequence_number, prev_sequence_number);
+}
+
+// NB: Doesn't fulfill strict weak ordering requirements.
+//     Mustn't be used as std::map Compare function.
+inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
+  return IsNewer(timestamp, prev_timestamp);
+}
+
+inline uint16_t LatestSequenceNumber(uint16_t sequence_number1,
+                                     uint16_t sequence_number2) {
+  return IsNewerSequenceNumber(sequence_number1, sequence_number2)
+             ? sequence_number1
+             : sequence_number2;
+}
+
+inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) {
+  return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2;
+}
+
+}  // namespace webrtc
+#endif  // MODULES_INCLUDE_MODULE_COMMON_TYPES_PUBLIC_H_

TMessagesProj/jni/voip/webrtc/modules/include/module_fec_types.h

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+/*
+ *  Copyright (c) 2018 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 MODULES_INCLUDE_MODULE_FEC_TYPES_H_
+#define MODULES_INCLUDE_MODULE_FEC_TYPES_H_
+
+namespace webrtc {
+
+// Types for the FEC packet masks. The type `kFecMaskRandom` is based on a
+// random loss model. The type `kFecMaskBursty` is based on a bursty/consecutive
+// loss model. The packet masks are defined in
+// modules/rtp_rtcp/fec_private_tables_random(bursty).h
+enum FecMaskType {
+  kFecMaskRandom,
+  kFecMaskBursty,
+};
+
+// Struct containing forward error correction settings.
+struct FecProtectionParams {
+  int fec_rate = 0;
+  int max_fec_frames = 0;
+  FecMaskType fec_mask_type = FecMaskType::kFecMaskRandom;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_INCLUDE_MODULE_FEC_TYPES_H_