Remove include dirs from modules/{media_file,pacing}

webrtc/modules/pacing/include/paced_sender.h

-/*
- *  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 WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
-#define WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_
-
-#include <list>
-#include <set>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/include/module.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-class BitrateProber;
-class Clock;
-class CriticalSectionWrapper;
-
-namespace paced_sender {
-class IntervalBudget;
-struct Packet;
-class PacketQueue;
-}  // namespace paced_sender
-
-class PacedSender : public Module, public RtpPacketSender {
- public:
-  class Callback {
-   public:
-    // Note: packets sent as a result of a callback should not pass by this
-    // module again.
-    // Called when it's time to send a queued packet.
-    // Returns false if packet cannot be sent.
-    virtual bool TimeToSendPacket(uint32_t ssrc,
-                                  uint16_t sequence_number,
-                                  int64_t capture_time_ms,
-                                  bool retransmission) = 0;
-    // Called when it's a good time to send a padding data.
-    // Returns the number of bytes sent.
-    virtual size_t TimeToSendPadding(size_t bytes) = 0;
-
-   protected:
-    virtual ~Callback() {}
-  };
-
-  static const int64_t kDefaultMaxQueueLengthMs = 2000;
-  // Pace in kbits/s until we receive first estimate.
-  static const int kDefaultInitialPaceKbps = 2000;
-  // Pacing-rate relative to our target send rate.
-  // Multiplicative factor that is applied to the target bitrate to calculate
-  // the number of bytes that can be transmitted per interval.
-  // Increasing this factor will result in lower delays in cases of bitrate
-  // overshoots from the encoder.
-  static const float kDefaultPaceMultiplier;
-
-  static const size_t kMinProbePacketSize = 200;
-
-  PacedSender(Clock* clock,
-              Callback* callback,
-              int bitrate_kbps,
-              int max_bitrate_kbps,
-              int min_bitrate_kbps);
-
-  virtual ~PacedSender();
-
-  // Temporarily pause all sending.
-  void Pause();
-
-  // Resume sending packets.
-  void Resume();
-
-  // Enable bitrate probing. Enabled by default, mostly here to simplify
-  // testing. Must be called before any packets are being sent to have an
-  // effect.
-  void SetProbingEnabled(bool enabled);
-
-  // Set target bitrates for the pacer.
-  // We will pace out bursts of packets at a bitrate of |max_bitrate_kbps|.
-  // |bitrate_kbps| is our estimate of what we are allowed to send on average.
-  // Padding packets will be utilized to reach |min_bitrate| unless enough media
-  // packets are available.
-  void UpdateBitrate(int bitrate_kbps,
-                     int max_bitrate_kbps,
-                     int min_bitrate_kbps);
-
-  // Returns true if we send the packet now, else it will add the packet
-  // information to the queue and call TimeToSendPacket when it's time to send.
-  void InsertPacket(RtpPacketSender::Priority priority,
-                    uint32_t ssrc,
-                    uint16_t sequence_number,
-                    int64_t capture_time_ms,
-                    size_t bytes,
-                    bool retransmission) override;
-
-  // Returns the time since the oldest queued packet was enqueued.
-  virtual int64_t QueueInMs() const;
-
-  virtual size_t QueueSizePackets() const;
-
-  // Returns the number of milliseconds it will take to send the current
-  // packets in the queue, given the current size and bitrate, ignoring prio.
-  virtual int64_t ExpectedQueueTimeMs() const;
-
-  // Returns the number of milliseconds until the module want a worker thread
-  // to call Process.
-  int64_t TimeUntilNextProcess() override;
-
-  // Process any pending packets in the queue(s).
-  int32_t Process() override;
-
- private:
-  // Updates the number of bytes that can be sent for the next time interval.
-  void UpdateBytesPerInterval(int64_t delta_time_in_ms)
-      EXCLUSIVE_LOCKS_REQUIRED(critsect_);
-
-  bool SendPacket(const paced_sender::Packet& packet)
-      EXCLUSIVE_LOCKS_REQUIRED(critsect_);
-  void SendPadding(size_t padding_needed) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
-
-  Clock* const clock_;
-  Callback* const callback_;
-
-  rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
-  bool paused_ GUARDED_BY(critsect_);
-  bool probing_enabled_;
-  // This is the media budget, keeping track of how many bits of media
-  // we can pace out during the current interval.
-  rtc::scoped_ptr<paced_sender::IntervalBudget> media_budget_
-      GUARDED_BY(critsect_);
-  // This is the padding budget, keeping track of how many bits of padding we're
-  // allowed to send out during the current interval. This budget will be
-  // utilized when there's no media to send.
-  rtc::scoped_ptr<paced_sender::IntervalBudget> padding_budget_
-      GUARDED_BY(critsect_);
-
-  rtc::scoped_ptr<BitrateProber> prober_ GUARDED_BY(critsect_);
-  int bitrate_bps_ GUARDED_BY(critsect_);
-
-  int64_t time_last_update_us_ GUARDED_BY(critsect_);
-
-  rtc::scoped_ptr<paced_sender::PacketQueue> packets_ GUARDED_BY(critsect_);
-  uint64_t packet_counter_;
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_PACING_INCLUDE_PACED_SENDER_H_