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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
73 lines
2.3 KiB
C++
73 lines
2.3 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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#ifndef MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
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#define MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
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#include <stdint.h>
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#include <list>
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#include "modules/video_coding/packet.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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const unsigned int kDefaultBitrateKbps = 1000;
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const unsigned int kDefaultFrameRate = 25;
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const unsigned int kMaxPacketSize = 1500;
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const unsigned int kFrameSize =
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(kDefaultBitrateKbps + kDefaultFrameRate * 4) / (kDefaultFrameRate * 8);
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const int kDefaultFramePeriodMs = 1000 / kDefaultFrameRate;
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class StreamGenerator {
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public:
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StreamGenerator(uint16_t start_seq_num, int64_t current_time);
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void Init(uint16_t start_seq_num, int64_t current_time);
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// |time_ms| denotes the timestamp you want to put on the frame, and the unit
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// is millisecond. GenerateFrame will translate |time_ms| into a 90kHz
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// timestamp and put it on the frame.
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void GenerateFrame(VideoFrameType 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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bool PopPacket(VCMPacket* packet, int index);
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void DropLastPacket();
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bool GetPacket(VCMPacket* packet, int index);
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bool NextPacket(VCMPacket* packet);
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uint16_t NextSequenceNumber() const;
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int PacketsRemaining() const;
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private:
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VCMPacket 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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VideoFrameType type);
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std::list<VCMPacket>::iterator GetPacketIterator(int index);
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std::list<VCMPacket> packets_;
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uint16_t sequence_number_;
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int64_t start_time_;
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uint8_t packet_buffer_[kMaxPacketSize];
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RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
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};
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
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