modules/video_coding refactorings

webrtc/modules/video_coding/main/test/rtp_player.h

Index: webrtc/modules/video_coding/main/test/rtp_player.h
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.h b/webrtc/modules/video_coding/main/test/rtp_player.h
deleted file mode 100644
index a2ecadd6151ce4521b5b9e8303c6ec169f0424f2..0000000000000000000000000000000000000000
--- a/webrtc/modules/video_coding/main/test/rtp_player.h
+++ /dev/null
@@ -1,97 +0,0 @@
-/*
- *  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_VIDEO_CODING_TEST_RTP_PLAYER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
-
-#include <string>
-#include <vector>
-
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
-
-namespace webrtc {
-class Clock;
-
-namespace rtpplayer {
-
-class PayloadCodecTuple {
- public:
-  PayloadCodecTuple(uint8_t payload_type, const std::string& codec_name,
-                    VideoCodecType codec_type)
-      : name_(codec_name),
-        payload_type_(payload_type),
-        codec_type_(codec_type) {
-  }
-
-  const std::string& name() const { return name_; }
-  uint8_t payload_type() const { return payload_type_; }
-  VideoCodecType codec_type() const { return codec_type_; }
-
- private:
-  std::string name_;
-  uint8_t payload_type_;
-  VideoCodecType codec_type_;
-};
-
-typedef std::vector<PayloadCodecTuple> PayloadTypes;
-typedef std::vector<PayloadCodecTuple>::const_iterator PayloadTypesIterator;
-
-// Implemented by RtpPlayer and given to client as a means to retrieve
-// information about a specific RTP stream.
-class RtpStreamInterface {
- public:
-  virtual ~RtpStreamInterface() {}
-
-  // Ask for missing packets to be resent.
-  virtual void ResendPackets(const uint16_t* sequence_numbers,
-                             uint16_t length) = 0;
-
-  virtual uint32_t ssrc() const = 0;
-  virtual const PayloadTypes& payload_types() const = 0;
-};
-
-// Implemented by a sink. Wraps RtpData because its d-tor is protected.
-class PayloadSinkInterface : public RtpData {
- public:
-  virtual ~PayloadSinkInterface() {}
-};
-
-// Implemented to provide a sink for RTP data, such as hooking up a VCM to
-// the incoming RTP stream.
-class PayloadSinkFactoryInterface {
- public:
-  virtual ~PayloadSinkFactoryInterface() {}
-
-  // Return NULL if failed to create sink. 'stream' is guaranteed to be
-  // around for as long as the RtpData. The returned object is owned by
-  // the caller (RtpPlayer).
-  virtual PayloadSinkInterface* Create(RtpStreamInterface* stream) = 0;
-};
-
-// The client's view of an RtpPlayer.
-class RtpPlayerInterface {
- public:
-  virtual ~RtpPlayerInterface() {}
-
-  virtual int NextPacket(int64_t timeNow) = 0;
-  virtual uint32_t TimeUntilNextPacket() const = 0;
-  virtual void Print() const = 0;
-};
-
-RtpPlayerInterface* Create(const std::string& inputFilename,
-    PayloadSinkFactoryInterface* payloadSinkFactory, Clock* clock,
-    const PayloadTypes& payload_types, float lossRate, int64_t rttMs,
-    bool reordering);
-
-}  // namespace rtpplayer
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_