4ef438e2de
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
106 lines
2.7 KiB
C++
106 lines
2.7 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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};
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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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};
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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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std::string ToString() const;
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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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// RTP header extension to use for the video stream, see RFC 5285.
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struct RtpExtension {
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RtpExtension(const char* name, int id) : name(name), id(id) {}
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std::string ToString() const;
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// TODO(mflodman) Add API to query supported extensions.
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static const char* kTOffset;
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static const char* kAbsSendTime;
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std::string name;
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int id;
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};
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struct VideoStream {
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VideoStream()
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: width(0),
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height(0),
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max_framerate(-1),
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min_bitrate_bps(-1),
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target_bitrate_bps(-1),
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max_bitrate_bps(-1),
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max_qp(-1) {}
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std::string ToString() const;
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size_t width;
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size_t height;
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int max_framerate;
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int min_bitrate_bps;
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int target_bitrate_bps;
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int max_bitrate_bps;
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int max_qp;
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// Bitrate thresholds for enabling additional temporal layers.
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std::vector<int> temporal_layers;
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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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