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/*
* 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) {
if (!type) {
// Use raw packetizer.
return std::make_unique<RtpPacketizerGeneric>(payload, limits);
}
switch (*type) {
case kVideoCodecH264: {
const auto& h264 =
absl::get<RTPVideoHeaderH264>(rtp_video_header.video_type_header);
return std::make_unique<RtpPacketizerH264>(payload, limits,
h264.packetization_mode);
}
case kVideoCodecVP8: {
const auto& vp8 =
absl::get<RTPVideoHeaderVP8>(rtp_video_header.video_type_header);
return std::make_unique<RtpPacketizerVp8>(payload, limits, vp8);
}
case kVideoCodecVP9: {
const auto& vp9 =
absl::get<RTPVideoHeaderVP9>(rtp_video_header.video_type_header);
return std::make_unique<RtpPacketizerVp9>(payload, limits, vp9);
}
case kVideoCodecAV1:
return std::make_unique<RtpPacketizerAv1>(
payload, limits, rtp_video_header.frame_type,
rtp_video_header.is_last_frame_in_picture);
#ifdef RTC_ENABLE_H265
case kVideoCodecH265: {
return std::make_unique<RtpPacketizerH265>(payload, limits);
}
#endif
default: {
return std::make_unique<RtpPacketizerGeneric>(payload, limits,
rtp_video_header);
}
}
}
std::vector<int> RtpPacketizer::SplitAboutEqually(
int payload_len,
const PayloadSizeLimits& limits) {
RTC_DCHECK_GT(payload_len, 0);
// First or last packet larger than normal are unsupported.
RTC_DCHECK_GE(limits.first_packet_reduction_len, 0);
RTC_DCHECK_GE(limits.last_packet_reduction_len, 0);
std::vector<int> result;
if (limits.max_payload_len >=
limits.single_packet_reduction_len + payload_len) {
result.push_back(payload_len);
return result;
}
if (limits.max_payload_len - limits.first_packet_reduction_len < 1 ||
limits.max_payload_len - limits.last_packet_reduction_len < 1) {
// Capacity is not enough to put a single byte into one of the packets.
return result;
}
// First and last packet of the frame can be smaller. Pretend that it's
// the same size, but we must write more payload to it.
// Assume frame fits in single packet if packet has extra space for sum
// of first and last packets reductions.
int total_bytes = payload_len + limits.first_packet_reduction_len +
limits.last_packet_reduction_len;
// Integer divisions with rounding up.
int num_packets_left =
(total_bytes + limits.max_payload_len - 1) / limits.max_payload_len;
if (num_packets_left == 1) {
// Single packet is a special case handled above.
num_packets_left = 2;
}
if (payload_len < num_packets_left) {
// Edge case where limits force to have more packets than there are payload
// bytes. This may happen when there is single byte of payload that can't be
// put into single packet if
// first_packet_reduction + last_packet_reduction >= max_payload_len.
return result;
}
int bytes_per_packet = total_bytes / num_packets_left;
int num_larger_packets = total_bytes % num_packets_left;
int remaining_data = payload_len;
result.reserve(num_packets_left);
bool first_packet = true;
while (remaining_data > 0) {
// Last num_larger_packets are 1 byte wider than the rest. Increase
// per-packet payload size when needed.
if (num_packets_left == num_larger_packets)
++bytes_per_packet;
int current_packet_bytes = bytes_per_packet;
if (first_packet) {
if (current_packet_bytes > limits.first_packet_reduction_len + 1)
current_packet_bytes -= limits.first_packet_reduction_len;
else
current_packet_bytes = 1;
}
if (current_packet_bytes > remaining_data) {
current_packet_bytes = remaining_data;
}
// This is not the last packet in the whole payload, but there's no data
// left for the last packet. Leave at least one byte for the last packet.
if (num_packets_left == 2 && current_packet_bytes == remaining_data) {
--current_packet_bytes;
}
result.push_back(current_packet_bytes);
remaining_data -= current_packet_bytes;
--num_packets_left;
first_packet = false;
}
return result;
}
} // namespace webrtc
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