webrtc/logging/rtc_event_log/rtc_stream_config.h
Yves Gerey 988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00

60 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 "api/rtp_headers.h"
#include "api/rtpparameters.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(const std::string& 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_