ccd42840bc
Note, this CL does not contain any tests. Those are implemeted as call tests and will be submitted when the receive stream is wired up as well. BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5559006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
99 lines
2.8 KiB
C++
99 lines
2.8 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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// TODO(pbos): Move Config from common.h to here.
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#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
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#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
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#include <string>
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#include <vector>
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#include "webrtc/common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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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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struct StreamStats {
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StreamStats() : key_frames(0), delta_frames(0), bitrate_bps(0) {}
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uint32_t key_frames;
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uint32_t delta_frames;
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int32_t bitrate_bps;
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StreamDataCounters rtp_stats;
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RtcpStatistics rtcp_stats;
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bool operator==(const StreamStats& other) const {
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return key_frames == other.key_frames &&
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delta_frames == other.delta_frames &&
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bitrate_bps == other.bitrate_bps && rtp_stats == other.rtp_stats &&
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rtcp_stats == other.rtcp_stats;
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}
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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() : rtx_payload_type(0), video_payload_type(0) {}
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// SSRCs to use for the RTX streams.
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std::vector<uint32_t> ssrcs;
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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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static const char* kTOffset;
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static const char* kAbsSendTime;
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RtpExtension(const char* name, int id) : name(name), id(id) {}
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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 webrtc
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#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
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