mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
277766f55e
Bug: webrtc:13579 Change-Id: Ibc5fa7842d52321d61cc4cdd4770635af737ddff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267170 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37533}
62 lines
1.5 KiB
C++
62 lines
1.5 KiB
C++
/*
|
|
* Copyright (c) 2017 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 LOGGING_RTC_EVENT_LOG_RTC_STREAM_CONFIG_H_
|
|
#define LOGGING_RTC_EVENT_LOG_RTC_STREAM_CONFIG_H_
|
|
|
|
#include <stdint.h>
|
|
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "absl/strings/string_view.h"
|
|
#include "api/rtp_headers.h"
|
|
#include "api/rtp_parameters.h"
|
|
|
|
namespace webrtc {
|
|
namespace rtclog {
|
|
|
|
struct StreamConfig {
|
|
StreamConfig();
|
|
StreamConfig(const StreamConfig& other);
|
|
~StreamConfig();
|
|
|
|
bool operator==(const StreamConfig& other) const;
|
|
bool operator!=(const StreamConfig& other) const;
|
|
|
|
uint32_t local_ssrc = 0;
|
|
uint32_t remote_ssrc = 0;
|
|
uint32_t rtx_ssrc = 0;
|
|
std::string rsid;
|
|
|
|
bool remb = false;
|
|
std::vector<RtpExtension> rtp_extensions;
|
|
|
|
RtcpMode rtcp_mode = RtcpMode::kReducedSize;
|
|
|
|
struct Codec {
|
|
Codec(absl::string_view payload_name,
|
|
int payload_type,
|
|
int rtx_payload_type);
|
|
|
|
bool operator==(const Codec& other) const;
|
|
|
|
std::string payload_name;
|
|
int payload_type;
|
|
int rtx_payload_type;
|
|
};
|
|
|
|
std::vector<Codec> codecs;
|
|
};
|
|
|
|
} // namespace rtclog
|
|
} // namespace webrtc
|
|
|
|
#endif // LOGGING_RTC_EVENT_LOG_RTC_STREAM_CONFIG_H_
|