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}

1 files changed, 160 insertions, 0 deletions

diff --git a/call/rampup_tests.h b/call/rampup_tests.h
new file mode 100644
index 0000000000..83588221f1
--- /dev/null
+++ b/call/rampup_tests.h
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2014 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_CALL_RAMPUP_TESTS_H_
+#define WEBRTC_CALL_RAMPUP_TESTS_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "webrtc/call/call.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/rtc_base/event.h"
+#include "webrtc/test/call_test.h"
+
+namespace webrtc {
+
+static const int kTransmissionTimeOffsetExtensionId = 6;
+static const int kAbsSendTimeExtensionId = 7;
+static const int kTransportSequenceNumberExtensionId = 8;
+static const unsigned int kSingleStreamTargetBps = 1000000;
+
+class Clock;
+
+class RampUpTester : public test::EndToEndTest {
+ public:
+  RampUpTester(size_t num_video_streams,
+               size_t num_audio_streams,
+               size_t num_flexfec_streams,
+               unsigned int start_bitrate_bps,
+               int64_t min_run_time_ms,
+               const std::string& extension_type,
+               bool rtx,
+               bool red,
+               bool report_perf_stats);
+  ~RampUpTester() override;
+
+  size_t GetNumVideoStreams() const override;
+  size_t GetNumAudioStreams() const override;
+  size_t GetNumFlexfecStreams() const override;
+
+  void PerformTest() override;
+
+ protected:
+  virtual void PollStats();
+
+  void AccumulateStats(const VideoSendStream::StreamStats& stream,
+                       size_t* total_packets_sent,
+                       size_t* total_sent,
+                       size_t* padding_sent,
+                       size_t* media_sent) const;
+
+  void ReportResult(const std::string& measurement,
+                    size_t value,
+                    const std::string& units) const;
+  void TriggerTestDone();
+
+  webrtc::RtcEventLogNullImpl event_log_;
+  rtc::Event stop_event_;
+  Clock* const clock_;
+  FakeNetworkPipe::Config forward_transport_config_;
+  const size_t num_video_streams_;
+  const size_t num_audio_streams_;
+  const size_t num_flexfec_streams_;
+  const bool rtx_;
+  const bool red_;
+  const bool report_perf_stats_;
+  Call* sender_call_;
+  VideoSendStream* send_stream_;
+  test::PacketTransport* send_transport_;
+
+ private:
+  typedef std::map<uint32_t, uint32_t> SsrcMap;
+  class VideoStreamFactory;
+
+  Call::Config GetSenderCallConfig() override;
+  void OnVideoStreamsCreated(
+      VideoSendStream* send_stream,
+      const std::vector<VideoReceiveStream*>& receive_streams) override;
+  test::PacketTransport* CreateSendTransport(
+      test::SingleThreadedTaskQueueForTesting* task_queue,
+      Call* sender_call) override;
+  void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) override;
+  void ModifyAudioConfigs(
+      AudioSendStream::Config* send_config,
+      std::vector<AudioReceiveStream::Config>* receive_configs) override;
+  void ModifyFlexfecConfigs(
+      std::vector<FlexfecReceiveStream::Config>* receive_configs) override;
+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
+
+  static void BitrateStatsPollingThread(void* obj);
+
+  const int start_bitrate_bps_;
+  const int64_t min_run_time_ms_;
+  int expected_bitrate_bps_;
+  int64_t test_start_ms_;
+  int64_t ramp_up_finished_ms_;
+
+  const std::string extension_type_;
+  std::vector<uint32_t> video_ssrcs_;
+  std::vector<uint32_t> video_rtx_ssrcs_;
+  std::vector<uint32_t> audio_ssrcs_;
+
+  rtc::PlatformThread poller_thread_;
+};
+
+class RampUpDownUpTester : public RampUpTester {
+ public:
+  RampUpDownUpTester(size_t num_video_streams,
+                     size_t num_audio_streams,
+                     size_t num_flexfec_streams,
+                     unsigned int start_bitrate_bps,
+                     const std::string& extension_type,
+                     bool rtx,
+                     bool red,
+                     const std::vector<int>& loss_rates,
+                     bool report_perf_stats);
+  ~RampUpDownUpTester() override;
+
+ protected:
+  void PollStats() override;
+
+ private:
+  enum TestStates {
+    kFirstRampup = 0,
+    kLowRate,
+    kSecondRampup,
+    kTestEnd,
+    kTransitionToNextState,
+  };
+
+  Call::Config GetReceiverCallConfig() override;
+
+  std::string GetModifierString() const;
+  int GetExpectedHighBitrate() const;
+  int GetHighLinkCapacity() const;
+  size_t GetFecBytes() const;
+  bool ExpectingFec() const;
+  void EvolveTestState(int bitrate_bps, bool suspended);
+
+  const std::vector<int> link_rates_;
+  TestStates test_state_;
+  TestStates next_state_;
+  int64_t state_start_ms_;
+  int64_t interval_start_ms_;
+  int sent_bytes_;
+  std::vector<int> loss_rates_;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_CALL_RAMPUP_TESTS_H_