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190244bb59
This CL completes the removal of assert() and relative headers from the codebase (excluded //examples/objc/AppRTCMobile/third_party/SocketRocket which is in a third_party sub-directory). Bug: webrtc:6779 Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34528}
100 lines
3.3 KiB
C++
100 lines
3.3 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
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#include <string.h>
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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static const size_t kGenericHeaderLength = 1;
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static const size_t kExtendedHeaderLength = 2;
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RtpPacketizerGeneric::RtpPacketizerGeneric(
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rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits,
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const RTPVideoHeader& rtp_video_header)
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: remaining_payload_(payload) {
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BuildHeader(rtp_video_header);
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limits.max_payload_len -= header_size_;
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payload_sizes_ = SplitAboutEqually(payload.size(), limits);
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current_packet_ = payload_sizes_.begin();
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}
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RtpPacketizerGeneric::RtpPacketizerGeneric(
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rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits)
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: header_size_(0), remaining_payload_(payload) {
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payload_sizes_ = SplitAboutEqually(payload.size(), limits);
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current_packet_ = payload_sizes_.begin();
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}
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RtpPacketizerGeneric::~RtpPacketizerGeneric() = default;
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size_t RtpPacketizerGeneric::NumPackets() const {
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return payload_sizes_.end() - current_packet_;
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}
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bool RtpPacketizerGeneric::NextPacket(RtpPacketToSend* packet) {
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RTC_DCHECK(packet);
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if (current_packet_ == payload_sizes_.end())
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return false;
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size_t next_packet_payload_len = *current_packet_;
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uint8_t* out_ptr =
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packet->AllocatePayload(header_size_ + next_packet_payload_len);
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RTC_CHECK(out_ptr);
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if (header_size_ > 0) {
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memcpy(out_ptr, header_, header_size_);
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// Remove first-packet bit, following packets are intermediate.
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header_[0] &= ~RtpFormatVideoGeneric::kFirstPacketBit;
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}
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memcpy(out_ptr + header_size_, remaining_payload_.data(),
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next_packet_payload_len);
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remaining_payload_ = remaining_payload_.subview(next_packet_payload_len);
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++current_packet_;
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// Packets left to produce and data left to split should end at the same time.
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RTC_DCHECK_EQ(current_packet_ == payload_sizes_.end(),
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remaining_payload_.empty());
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packet->SetMarker(remaining_payload_.empty());
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return true;
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}
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void RtpPacketizerGeneric::BuildHeader(const RTPVideoHeader& rtp_video_header) {
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header_size_ = kGenericHeaderLength;
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header_[0] = RtpFormatVideoGeneric::kFirstPacketBit;
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if (rtp_video_header.frame_type == VideoFrameType::kVideoFrameKey) {
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header_[0] |= RtpFormatVideoGeneric::kKeyFrameBit;
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}
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if (const auto* generic_header = absl::get_if<RTPVideoHeaderLegacyGeneric>(
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&rtp_video_header.video_type_header)) {
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// Store bottom 15 bits of the picture id. Only 15 bits are used for
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// compatibility with other packetizer implemenetations.
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uint16_t picture_id = generic_header->picture_id;
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header_[0] |= RtpFormatVideoGeneric::kExtendedHeaderBit;
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header_[1] = (picture_id >> 8) & 0x7F;
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header_[2] = picture_id & 0xFF;
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header_size_ += kExtendedHeaderLength;
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}
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}
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} // namespace webrtc
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