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Diffstat (limited to 'webrtc/modules/video_coding/main/test/rtp_player.h')
-rw-r--r-- | webrtc/modules/video_coding/main/test/rtp_player.h | 97 |
1 files changed, 0 insertions, 97 deletions
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 7459231416..0000000000 --- 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/interface/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_ |