webrtc/modules/video_coding/test/stream_generator.h
Steve Anton 10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00

73 lines
2.3 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.
*/
#ifndef MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
#define MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
#include <stdint.h>
#include <list>
#include "common_types.h" // NOLINT(build/include)
#include "modules/video_coding/packet.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
const unsigned int kDefaultBitrateKbps = 1000;
const unsigned int kDefaultFrameRate = 25;
const unsigned int kMaxPacketSize = 1500;
const unsigned int kFrameSize =
(kDefaultBitrateKbps + kDefaultFrameRate * 4) / (kDefaultFrameRate * 8);
const int kDefaultFramePeriodMs = 1000 / kDefaultFrameRate;
class StreamGenerator {
public:
StreamGenerator(uint16_t start_seq_num, int64_t current_time);
void Init(uint16_t start_seq_num, int64_t current_time);
// |time_ms| denotes the timestamp you want to put on the frame, and the unit
// is millisecond. GenerateFrame will translate |time_ms| into a 90kHz
// timestamp and put it on the frame.
void GenerateFrame(FrameType type,
int num_media_packets,
int num_empty_packets,
int64_t time_ms);
bool PopPacket(VCMPacket* packet, int index);
void DropLastPacket();
bool GetPacket(VCMPacket* packet, int index);
bool NextPacket(VCMPacket* packet);
uint16_t NextSequenceNumber() const;
int PacketsRemaining() const;
private:
VCMPacket GeneratePacket(uint16_t sequence_number,
uint32_t timestamp,
unsigned int size,
bool first_packet,
bool marker_bit,
FrameType type);
std::list<VCMPacket>::iterator GetPacketIterator(int index);
std::list<VCMPacket> packets_;
uint16_t sequence_number_;
int64_t start_time_;
uint8_t packet_buffer_[kMaxPacketSize];
RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
};
} // namespace webrtc
#endif // MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_