blob: d9b082bd903e34cc1444050dc3af3e9e39999157 [file] [log] [blame]
/*
* 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.
*/
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {}
constexpr size_t StreamId::kMaxSize;
bool StreamId::IsLegalName(rtc::ArrayView<const char> name) {
return (name.size() <= kMaxSize && name.size() > 0 &&
std::all_of(name.data(), name.data() + name.size(), isalnum));
}
void StreamId::Set(const char* data, size_t size) {
// If |data| contains \0, the stream id size might become less than |size|.
RTC_CHECK_LE(size, kMaxSize);
memcpy(value_, data, size);
if (size < kMaxSize)
value_[size] = 0;
}
// StreamId is used as member of RTPHeader that is sometimes copied with memcpy
// and thus assume trivial destructibility.
static_assert(std::is_trivially_destructible<StreamId>::value, "");
PayloadUnion::PayloadUnion(const AudioPayload& payload)
: audio_payload_(payload) {}
PayloadUnion::PayloadUnion(const VideoPayload& payload)
: video_payload_(payload) {}
PayloadUnion::PayloadUnion(const PayloadUnion&) = default;
PayloadUnion::PayloadUnion(PayloadUnion&&) = default;
PayloadUnion::~PayloadUnion() = default;
PayloadUnion& PayloadUnion::operator=(const PayloadUnion&) = default;
PayloadUnion& PayloadUnion::operator=(PayloadUnion&&) = default;
} // namespace webrtc