Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 06:15:48 +02:00
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
--- a/modules/video_coding/test/stream_generator.h
+++ b/modules/video_coding/test/stream_generator.h
@ -0,0 +1,72 @@
+/*
+ *  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 WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
+
+#include <list>
+
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/typedefs.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  // WEBRTC_MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_