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}

modules/video_coding/test/stream_generator.cc

@@ -0,0 +1,129 @@
+/*
+ *  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 "webrtc/modules/video_coding/test/stream_generator.h"
+
+#include <string.h>
+
+#include <list>
+
+#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/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(FrameType 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,
+                                          FrameType type) {
+  EXPECT_LT(size, kMaxPacketSize);
+  VCMPacket packet;
+  packet.seqNum = sequence_number;
+  packet.timestamp = timestamp;
+  packet.frameType = type;
+  packet.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

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_

modules/video_coding/test/test_util.h

@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2012 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_TEST_UTIL_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
+
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+
+class NullEventFactory : public webrtc::EventFactory {
+ public:
+  virtual ~NullEventFactory() {}
+
+  webrtc::EventWrapper* CreateEvent() override { return new NullEvent; }
+ private:
+  // Private class to avoid more dependencies on it in tests.
+  class NullEvent : public webrtc::EventWrapper {
+   public:
+    ~NullEvent() override {}
+    bool Set() override { return true; }
+    webrtc::EventTypeWrapper Wait(unsigned long max_time) override {  // NOLINT
+      return webrtc::kEventTimeout;
+    }
+  };
+};
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_