Repo created

TMessagesProj/jni/voip/webrtc/api/call/audio_sink.h

@@ -0,0 +1,48 @@
+/*
+ *  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 API_CALL_AUDIO_SINK_H_
+#define API_CALL_AUDIO_SINK_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// Represents a simple push audio sink.
+class AudioSinkInterface {
+ public:
+  virtual ~AudioSinkInterface() {}
+
+  struct Data {
+    Data(const int16_t* data,
+         size_t samples_per_channel,
+         int sample_rate,
+         size_t channels,
+         uint32_t timestamp)
+        : data(data),
+          samples_per_channel(samples_per_channel),
+          sample_rate(sample_rate),
+          channels(channels),
+          timestamp(timestamp) {}
+
+    const int16_t* data;         // The actual 16bit audio data.
+    size_t samples_per_channel;  // Number of frames in the buffer.
+    int sample_rate;             // Sample rate in Hz.
+    size_t channels;             // Number of channels in the audio data.
+    uint32_t timestamp;          // The RTP timestamp of the first sample.
+  };
+
+  virtual void OnData(const Data& audio) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_CALL_AUDIO_SINK_H_

TMessagesProj/jni/voip/webrtc/api/call/bitrate_allocation.h

@@ -0,0 +1,45 @@
+/*
+ *  Copyright 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 API_CALL_BITRATE_ALLOCATION_H_
+#define API_CALL_BITRATE_ALLOCATION_H_
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+// BitrateAllocationUpdate provides information to allocated streams about their
+// bitrate allocation. It originates from the BitrateAllocater class and is
+// propagated from there.
+struct BitrateAllocationUpdate {
+  // The allocated target bitrate. Media streams should produce this amount of
+  // data. (Note that this may include packet overhead depending on
+  // configuration.)
+  DataRate target_bitrate = DataRate::Zero();
+  // The allocated part of the estimated link capacity. This is more stable than
+  // the target as it is based on the underlying link capacity estimate. This
+  // should be used to change encoder configuration when the cost of change is
+  // high.
+  DataRate stable_target_bitrate = DataRate::Zero();
+  // Predicted packet loss ratio.
+  double packet_loss_ratio = 0;
+  // Predicted round trip time.
+  TimeDelta round_trip_time = TimeDelta::PlusInfinity();
+  // `bwe_period` is deprecated, use `stable_target_bitrate` allocation instead.
+  TimeDelta bwe_period = TimeDelta::PlusInfinity();
+  // Congestion window pushback bitrate reduction fraction. Used in
+  // VideoStreamEncoder to reduce the bitrate by the given fraction
+  // by dropping frames.
+  double cwnd_reduce_ratio = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_CALL_BITRATE_ALLOCATION_H_

TMessagesProj/jni/voip/webrtc/api/call/transport.cc

@@ -0,0 +1,23 @@
+/*
+ *  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.
+ */
+
+#include "api/call/transport.h"
+
+#include <cstdint>
+
+namespace webrtc {
+
+PacketOptions::PacketOptions() = default;
+
+PacketOptions::PacketOptions(const PacketOptions&) = default;
+
+PacketOptions::~PacketOptions() = default;
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/call/transport.h

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2013 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 API_CALL_TRANSPORT_H_
+#define API_CALL_TRANSPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// TODO(holmer): Look into unifying this with the PacketOptions in
+// asyncpacketsocket.h.
+struct PacketOptions {
+  PacketOptions();
+  PacketOptions(const PacketOptions&);
+  ~PacketOptions();
+
+  // A 16 bits positive id. Negative ids are invalid and should be interpreted
+  // as packet_id not being set.
+  int packet_id = -1;
+  // Additional data bound to the RTP packet for use in application code,
+  // outside of WebRTC.
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data;
+  // Whether this is a retransmission of an earlier packet.
+  bool is_retransmit = false;
+  bool included_in_feedback = false;
+  bool included_in_allocation = false;
+  // Whether this packet can be part of a packet batch at lower levels.
+  bool batchable = false;
+  // Whether this packet is the last of a batch.
+  bool last_packet_in_batch = false;
+};
+
+class Transport {
+ public:
+  virtual bool SendRtp(rtc::ArrayView<const uint8_t> packet,
+                       const PacketOptions& options) = 0;
+  virtual bool SendRtcp(rtc::ArrayView<const uint8_t> packet) = 0;
+
+ protected:
+  virtual ~Transport() {}
+};
+
+}  // namespace webrtc
+
+#endif  // API_CALL_TRANSPORT_H_