09315705b9
This CL includes Call tests that test both send and receive sides. BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
79 lines
2.2 KiB
C++
79 lines
2.2 KiB
C++
/*
|
|
* Copyright (c) 2013 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.
|
|
*/
|
|
|
|
// TODO(pbos): Move Config from common.h to here.
|
|
|
|
#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
|
|
#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
|
|
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
struct RtpStatistics {
|
|
RtpStatistics()
|
|
: ssrc(0),
|
|
fraction_loss(0),
|
|
cumulative_loss(0),
|
|
extended_max_sequence_number(0) {}
|
|
uint32_t ssrc;
|
|
int fraction_loss;
|
|
int cumulative_loss;
|
|
int extended_max_sequence_number;
|
|
std::string c_name;
|
|
};
|
|
|
|
struct StreamStats {
|
|
StreamStats() : key_frames(0), delta_frames(0), bitrate_bps(0) {}
|
|
uint32_t key_frames;
|
|
uint32_t delta_frames;
|
|
int32_t bitrate_bps;
|
|
StreamDataCounters rtp_stats;
|
|
RtcpStatistics rtcp_stats;
|
|
};
|
|
|
|
// Settings for NACK, see RFC 4585 for details.
|
|
struct NackConfig {
|
|
NackConfig() : rtp_history_ms(0) {}
|
|
// Send side: the time RTP packets are stored for retransmissions.
|
|
// Receive side: the time the receiver is prepared to wait for
|
|
// retransmissions.
|
|
// Set to '0' to disable.
|
|
int rtp_history_ms;
|
|
};
|
|
|
|
// Settings for forward error correction, see RFC 5109 for details. Set the
|
|
// payload types to '-1' to disable.
|
|
struct FecConfig {
|
|
FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {}
|
|
// Payload type used for ULPFEC packets.
|
|
int ulpfec_payload_type;
|
|
|
|
// Payload type used for RED packets.
|
|
int red_payload_type;
|
|
};
|
|
|
|
// RTP header extension to use for the video stream, see RFC 5285.
|
|
struct RtpExtension {
|
|
static const char* kTOffset;
|
|
static const char* kAbsSendTime;
|
|
RtpExtension(const char* name, int id) : name(name), id(id) {}
|
|
// TODO(mflodman) Add API to query supported extensions.
|
|
std::string name;
|
|
int id;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
|