webrtc/modules/video_coding/test/stream_generator.cc
Niels Möller 87e2d785a0 Prepare for splitting FrameType into AudioFrameType and VideoFrameType
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.

After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.

Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
2019-03-07 10:12:57 +00:00

128 lines
3.9 KiB
C++

/*
* Copyright (c) 2013 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/video_coding/test/stream_generator.h"
#include <string.h>
#include <list>
#include "modules/include/module_common_types.h"
#include "modules/video_coding/packet.h"
#include "test/gtest.h"
namespace webrtc {
StreamGenerator::StreamGenerator(uint16_t start_seq_num, int64_t current_time)
: packets_(), sequence_number_(start_seq_num), start_time_(current_time) {}
void StreamGenerator::Init(uint16_t start_seq_num, int64_t current_time) {
packets_.clear();
sequence_number_ = start_seq_num;
start_time_ = current_time;
memset(packet_buffer_, 0, sizeof(packet_buffer_));
}
void StreamGenerator::GenerateFrame(VideoFrameType type,
int num_media_packets,
int num_empty_packets,
int64_t time_ms) {
uint32_t timestamp = 90 * (time_ms - start_time_);
for (int i = 0; i < num_media_packets; ++i) {
const int packet_size =
(kFrameSize + num_media_packets / 2) / num_media_packets;
bool marker_bit = (i == num_media_packets - 1);
packets_.push_back(GeneratePacket(sequence_number_, timestamp, packet_size,
(i == 0), marker_bit, type));
++sequence_number_;
}
for (int i = 0; i < num_empty_packets; ++i) {
packets_.push_back(GeneratePacket(sequence_number_, timestamp, 0, false,
false, kEmptyFrame));
++sequence_number_;
}
}
VCMPacket StreamGenerator::GeneratePacket(uint16_t sequence_number,
uint32_t timestamp,
unsigned int size,
bool first_packet,
bool marker_bit,
VideoFrameType type) {
EXPECT_LT(size, kMaxPacketSize);
VCMPacket packet;
packet.seqNum = sequence_number;
packet.timestamp = timestamp;
packet.frameType = type;
packet.video_header.is_first_packet_in_frame = first_packet;
packet.markerBit = marker_bit;
packet.sizeBytes = size;
packet.dataPtr = packet_buffer_;
if (packet.is_first_packet_in_frame())
packet.completeNALU = kNaluStart;
else if (packet.markerBit)
packet.completeNALU = kNaluEnd;
else
packet.completeNALU = kNaluIncomplete;
return packet;
}
bool StreamGenerator::PopPacket(VCMPacket* packet, int index) {
std::list<VCMPacket>::iterator it = GetPacketIterator(index);
if (it == packets_.end())
return false;
if (packet)
*packet = (*it);
packets_.erase(it);
return true;
}
bool StreamGenerator::GetPacket(VCMPacket* packet, int index) {
std::list<VCMPacket>::iterator it = GetPacketIterator(index);
if (it == packets_.end())
return false;
if (packet)
*packet = (*it);
return true;
}
bool StreamGenerator::NextPacket(VCMPacket* packet) {
if (packets_.empty())
return false;
if (packet != NULL)
*packet = packets_.front();
packets_.pop_front();
return true;
}
void StreamGenerator::DropLastPacket() {
packets_.pop_back();
}
uint16_t StreamGenerator::NextSequenceNumber() const {
if (packets_.empty())
return sequence_number_;
return packets_.front().seqNum;
}
int StreamGenerator::PacketsRemaining() const {
return packets_.size();
}
std::list<VCMPacket>::iterator StreamGenerator::GetPacketIterator(int index) {
std::list<VCMPacket>::iterator it = packets_.begin();
for (int i = 0; i < index; ++i) {
++it;
if (it == packets_.end())
break;
}
return it;
}
} // namespace webrtc