chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format.cc

/*
 *  Copyright (c) 2014 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/source/rtp_format.h"

#include <memory>

#include "absl/types/variant.h"
#include "modules/rtp_rtcp/source/rtp_format_h264.h"
#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
#include "modules/rtp_rtcp/source/rtp_packetizer_av1.h"
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
#include "rtc_base/checks.h"
#ifdef RTC_ENABLE_H265
#include "modules/rtp_rtcp/source/rtp_packetizer_h265.h"
#endif

namespace webrtc {

std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
    absl::optional<VideoCodecType> type,
    rtc::ArrayView<const uint8_t> payload,
    PayloadSizeLimits limits,
    // Codec-specific details.
    const RTPVideoHeader& rtp_video_header,
    bool enable_av1_even_split) { … }

std::vector<int> RtpPacketizer::SplitAboutEqually(
    int payload_len,
    const PayloadSizeLimits& limits) { … }

}  // namespace webrtc