0d1b2b6880
Add RTC_DEPRACATed anonymous unions to not break downstream projects.
Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date: Mon Dec 19 09:36:33 2016 -0800
>
> Revert of Rename RTPVideoHeader.isFirstPacket to
> .is_first_packet_in_frame. (patchset #1 id:1 of
> https://codereview.webrtc.org/2574943003/ )
>
> Reason for revert:
> breaks downstream project.
>
> Can you make this change in a compatible way using anonymous
> union:
> union {
> bool is_first_packet_in_frame;
> RTC_DEPRECATED bool isFirstPacket;
> };
> (unfortunetly this this treak breaks braced initialization in
> rtp_rtcp_impl_unittest.cc,
> so that should be rewritting in a more classic way)
>
> Original issue's description:
> > Rename RTPVideoHeader.isFirstPacket to
> > .is_first_packet_in_frame.
> >
> > Name should represent the actual meaning.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2574943003
> > Cr-Commit-Position: refs/heads/master@{#15684}
> > Committed:
> > efde908380
>
> TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days
> ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2589783003
> Cr-Commit-Position: refs/heads/master@{#15686}
>
BUG=None
Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
131 lines
4.0 KiB
C++
131 lines
4.0 KiB
C++
/*
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* Copyright (c) 2013 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 "webrtc/modules/video_coding/test/stream_generator.h"
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#include <string.h>
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#include <list>
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#include "webrtc/modules/video_coding/packet.h"
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#include "webrtc/modules/video_coding/test/test_util.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/test/gtest.h"
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namespace webrtc {
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StreamGenerator::StreamGenerator(uint16_t start_seq_num, int64_t current_time)
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: packets_(), sequence_number_(start_seq_num), start_time_(current_time) {}
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void StreamGenerator::Init(uint16_t start_seq_num, int64_t current_time) {
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packets_.clear();
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sequence_number_ = start_seq_num;
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start_time_ = current_time;
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memset(packet_buffer_, 0, sizeof(packet_buffer_));
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}
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void StreamGenerator::GenerateFrame(FrameType type,
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int num_media_packets,
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int num_empty_packets,
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int64_t time_ms) {
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uint32_t timestamp = 90 * (time_ms - start_time_);
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for (int i = 0; i < num_media_packets; ++i) {
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const int packet_size =
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(kFrameSize + num_media_packets / 2) / num_media_packets;
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bool marker_bit = (i == num_media_packets - 1);
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packets_.push_back(GeneratePacket(sequence_number_, timestamp, packet_size,
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(i == 0), marker_bit, type));
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++sequence_number_;
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}
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for (int i = 0; i < num_empty_packets; ++i) {
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packets_.push_back(GeneratePacket(sequence_number_, timestamp, 0, false,
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false, kEmptyFrame));
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++sequence_number_;
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}
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}
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VCMPacket StreamGenerator::GeneratePacket(uint16_t sequence_number,
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uint32_t timestamp,
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unsigned int size,
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bool first_packet,
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bool marker_bit,
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FrameType type) {
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EXPECT_LT(size, kMaxPacketSize);
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VCMPacket packet;
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packet.seqNum = sequence_number;
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packet.timestamp = timestamp;
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packet.frameType = type;
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packet.is_first_packet_in_frame = first_packet;
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packet.markerBit = marker_bit;
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packet.sizeBytes = size;
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packet.dataPtr = packet_buffer_;
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if (packet.is_first_packet_in_frame)
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packet.completeNALU = kNaluStart;
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else if (packet.markerBit)
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packet.completeNALU = kNaluEnd;
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else
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packet.completeNALU = kNaluIncomplete;
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return packet;
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}
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bool StreamGenerator::PopPacket(VCMPacket* packet, int index) {
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std::list<VCMPacket>::iterator it = GetPacketIterator(index);
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if (it == packets_.end())
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return false;
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if (packet)
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*packet = (*it);
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packets_.erase(it);
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return true;
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}
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bool StreamGenerator::GetPacket(VCMPacket* packet, int index) {
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std::list<VCMPacket>::iterator it = GetPacketIterator(index);
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if (it == packets_.end())
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return false;
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if (packet)
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*packet = (*it);
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return true;
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}
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bool StreamGenerator::NextPacket(VCMPacket* packet) {
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if (packets_.empty())
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return false;
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if (packet != NULL)
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*packet = packets_.front();
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packets_.pop_front();
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return true;
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}
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void StreamGenerator::DropLastPacket() {
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packets_.pop_back();
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}
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uint16_t StreamGenerator::NextSequenceNumber() const {
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if (packets_.empty())
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return sequence_number_;
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return packets_.front().seqNum;
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}
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int StreamGenerator::PacketsRemaining() const {
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return packets_.size();
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}
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std::list<VCMPacket>::iterator StreamGenerator::GetPacketIterator(int index) {
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std::list<VCMPacket>::iterator it = packets_.begin();
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for (int i = 0; i < index; ++i) {
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++it;
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if (it == packets_.end())
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break;
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}
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return it;
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}
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} // namespace webrtc
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