2557b86e76
The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
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 WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
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#define WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_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/typedefs.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(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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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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FrameType 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 // WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
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