eceb53241e
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1543004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4115 4adac7df-926f-26a2-2b94-8c16560cd09d
131 lines
3.4 KiB
C++
131 lines
3.4 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_COMMON_H_
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#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_COMMON_H_
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#include <string>
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#include "webrtc/common_types.h"
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namespace webrtc {
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class I420VideoFrame;
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namespace newapi {
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struct EncodedFrame;
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class I420FrameCallback {
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public:
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// This function is called with a I420 frame allowing the user to modify the
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// frame content.
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virtual void FrameCallback(I420VideoFrame* video_frame) = 0;
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protected:
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virtual ~I420FrameCallback() {}
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};
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class EncodedFrameObserver {
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public:
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virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) = 0;
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protected:
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virtual ~EncodedFrameObserver() {}
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};
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class VideoRenderer {
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public:
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// This function should return as soon as possible and not block until it's
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// time to render the frame.
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// TODO(mflodman) Remove time_to_render_ms when I420VideoFrame contains NTP.
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virtual void RenderFrame(const I420VideoFrame& video_frame,
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int time_to_render_ms) = 0;
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protected:
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virtual ~VideoRenderer() {}
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};
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class Transport {
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public:
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virtual bool SendRTP(const void* packet, size_t length) = 0;
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virtual bool SendRTCP(const void* packet, size_t length) = 0;
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protected:
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virtual ~Transport() {}
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};
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struct RtpStatistics {
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RtpStatistics()
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: ssrc(0),
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fraction_loss(0),
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cumulative_loss(0),
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extended_max_sequence_number(0) {}
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uint32_t ssrc;
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int fraction_loss;
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int cumulative_loss;
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int extended_max_sequence_number;
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std::string c_name;
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};
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// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
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// RTCP mode is described by RFC 5506.
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enum RtcpMode {
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kRtcpCompound,
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kRtcpReducedSize
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};
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// Settings for NACK, see RFC 4585 for details.
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struct NackConfig {
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NackConfig() : rtp_history_ms(0) {}
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// Send side: the time RTP packets are stored for retransmissions.
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// Receive side: the time the receiver is prepared to wait for
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// retransmissions.
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// Set to '0' to disable.
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int rtp_history_ms;
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};
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// Settings for forward error correction, see RFC 5109 for details. Set the
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// payload types to '-1' to disable.
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struct FecConfig {
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FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {}
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// Payload type used for ULPFEC packets.
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int ulpfec_payload_type;
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// Payload type used for RED packets.
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int red_payload_type;
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};
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// Settings for RTP retransmission payload format, see RFC 4588 for details.
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struct RtxConfig {
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RtxConfig() : ssrc(0), rtx_payload_type(0), video_payload_type(0) {}
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// SSRC to use for the RTX stream.
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uint32_t ssrc;
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// Payload type to use for the RTX stream.
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int rtx_payload_type;
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// Original video payload this RTX stream is used for.
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int video_payload_type;
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};
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// RTP header extension to use for the video stream, see RFC 5285.
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struct RtpExtension {
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RtpExtension() : id(0) {}
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// TODO(mflodman) Add API to query supported extensions.
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std::string name;
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int id;
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};
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} // namespace newapi
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_COMMON_H_
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