/*
 *  Copyright (c) 2015 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 <stdio.h>

#include "webrtc/rtc_base/flags.h"
#include "webrtc/rtc_base/stringencode.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/run_test.h"
#include "webrtc/video/video_quality_test.h"

namespace webrtc {
namespace flags {

// Flags common with video loopback, with different default values.
DEFINE_int(width, 1850, "Video width (crops source).");
size_t Width() {
  return static_cast<size_t>(FLAG_width);
}

DEFINE_int(height, 1110, "Video height (crops source).");
size_t Height() {
  return static_cast<size_t>(FLAG_height);
}

DEFINE_int(fps, 5, "Frames per second.");
int Fps() {
  return static_cast<int>(FLAG_fps);
}

DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps.");
int MinBitrateKbps() {
  return static_cast<int>(FLAG_min_bitrate);
}

DEFINE_int(start_bitrate,
           Call::Config::kDefaultStartBitrateBps / 1000,
           "Call start bitrate in kbps.");
int StartBitrateKbps() {
  return static_cast<int>(FLAG_start_bitrate);
}

DEFINE_int(target_bitrate, 200, "Stream target bitrate in kbps.");
int TargetBitrateKbps() {
  return static_cast<int>(FLAG_target_bitrate);
}

DEFINE_int(max_bitrate, 2000, "Call and stream max bitrate in kbps.");
int MaxBitrateKbps() {
  return static_cast<int>(FLAG_max_bitrate);
}

DEFINE_int(num_temporal_layers, 2, "Number of temporal layers to use.");
int NumTemporalLayers() {
  return static_cast<int>(FLAG_num_temporal_layers);
}

// Flags common with video loopback, with equal default values.
DEFINE_string(codec, "VP8", "Video codec to use.");
std::string Codec() {
  return static_cast<std::string>(FLAG_codec);
}

DEFINE_string(rtc_event_log_name, "", "Filename for rtc event log.");
std::string RtcEventLogName() {
  return static_cast<std::string>(FLAG_rtc_event_log_name);
}

DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream.");
std::string RtpDumpName() {
  return static_cast<std::string>(FLAG_rtp_dump_name);
}

DEFINE_int(selected_tl,
           -1,
           "Temporal layer to show or analyze. -1 to disable filtering.");
int SelectedTL() {
  return static_cast<int>(FLAG_selected_tl);
}

DEFINE_int(
    duration,
    0,
    "Duration of the test in seconds. If 0, rendered will be shown instead.");
int DurationSecs() {
  return static_cast<int>(FLAG_duration);
}

DEFINE_string(output_filename, "", "Target graph data filename.");
std::string OutputFilename() {
  return static_cast<std::string>(FLAG_output_filename);
}

DEFINE_string(graph_title,
              "",
              "If empty, title will be generated automatically.");
std::string GraphTitle() {
  return static_cast<std::string>(FLAG_graph_title);
}

DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost.");
int LossPercent() {
  return static_cast<int>(FLAG_loss_percent);
}

DEFINE_int(link_capacity,
           0,
           "Capacity (kbps) of the fake link. 0 means infinite.");
int LinkCapacityKbps() {
  return static_cast<int>(FLAG_link_capacity);
}

DEFINE_int(queue_size, 0, "Size of the bottleneck link queue in packets.");
int QueueSize() {
  return static_cast<int>(FLAG_queue_size);
}

DEFINE_int(avg_propagation_delay_ms,
           0,
           "Average link propagation delay in ms.");
int AvgPropagationDelayMs() {
  return static_cast<int>(FLAG_avg_propagation_delay_ms);
}

DEFINE_int(std_propagation_delay_ms,
           0,
           "Link propagation delay standard deviation in ms.");
int StdPropagationDelayMs() {
  return static_cast<int>(FLAG_std_propagation_delay_ms);
}

DEFINE_int(num_streams, 0, "Number of streams to show or analyze.");
int NumStreams() {
  return static_cast<int>(FLAG_num_streams);
}

DEFINE_int(selected_stream,
           0,
           "ID of the stream to show or analyze. "
           "Set to the number of streams to show them all.");
int SelectedStream() {
  return static_cast<int>(FLAG_selected_stream);
}

DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use.");
int NumSpatialLayers() {
  return static_cast<int>(FLAG_num_spatial_layers);
}

DEFINE_int(selected_sl,
           -1,
           "Spatial layer to show or analyze. -1 to disable filtering.");
int SelectedSL() {
  return static_cast<int>(FLAG_selected_sl);
}

DEFINE_string(stream0,
              "",
              "Comma separated values describing VideoStream for stream #0.");
std::string Stream0() {
  return static_cast<std::string>(FLAG_stream0);
}

DEFINE_string(stream1,
              "",
              "Comma separated values describing VideoStream for stream #1.");
std::string Stream1() {
  return static_cast<std::string>(FLAG_stream1);
}

DEFINE_string(sl0,
              "",
              "Comma separated values describing SpatialLayer for layer #0.");
std::string SL0() {
  return static_cast<std::string>(FLAG_sl0);
}

DEFINE_string(sl1,
              "",
              "Comma separated values describing SpatialLayer for layer #1.");
std::string SL1() {
  return static_cast<std::string>(FLAG_sl1);
}

DEFINE_string(encoded_frame_path,
              "",
              "The base path for encoded frame logs. Created files will have "
              "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
std::string EncodedFramePath() {
  return static_cast<std::string>(FLAG_encoded_frame_path);
}

DEFINE_bool(logs, false, "print logs to stderr");

DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");

DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");

DEFINE_string(
    force_fieldtrials,
    "",
    "Field trials control experimental feature code which can be forced. "
    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
    " will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
    "trials are separated by \"/\"");

// Screenshare-specific flags.
DEFINE_int(min_transmit_bitrate, 400, "Min transmit bitrate incl. padding.");
int MinTransmitBitrateKbps() {
  return FLAG_min_transmit_bitrate;
}

DEFINE_int(slide_change_interval,
           10,
           "Interval (in seconds) between simulated slide changes.");
int SlideChangeInterval() {
  return static_cast<int>(FLAG_slide_change_interval);
}

DEFINE_int(
    scroll_duration,
    0,
    "Duration (in seconds) during which a slide will be scrolled into place.");
int ScrollDuration() {
  return static_cast<int>(FLAG_scroll_duration);
}

DEFINE_string(slides,
              "",
              "Comma-separated list of *.yuv files to display as slides.");
std::vector<std::string> Slides() {
  std::vector<std::string> slides;
  std::string slides_list = FLAG_slides;
  rtc::tokenize(slides_list, ',', &slides);
  return slides;
}

DEFINE_bool(help, false, "prints this message");

}  // namespace flags

void Loopback() {
  FakeNetworkPipe::Config pipe_config;
  pipe_config.loss_percent = flags::LossPercent();
  pipe_config.link_capacity_kbps = flags::LinkCapacityKbps();
  pipe_config.queue_length_packets = flags::QueueSize();
  pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs();
  pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
  pipe_config.allow_reordering = flags::FLAG_allow_reordering;

  Call::Config::BitrateConfig call_bitrate_config;
  call_bitrate_config.min_bitrate_bps = flags::MinBitrateKbps() * 1000;
  call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
  call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;

  VideoQualityTest::Params params;
  params.call = {flags::FLAG_send_side_bwe, call_bitrate_config};
  params.video = {true,
                  flags::Width(),
                  flags::Height(),
                  flags::Fps(),
                  flags::MinBitrateKbps() * 1000,
                  flags::TargetBitrateKbps() * 1000,
                  flags::MaxBitrateKbps() * 1000,
                  false,
                  flags::Codec(),
                  flags::NumTemporalLayers(),
                  flags::SelectedTL(),
                  flags::MinTransmitBitrateKbps() * 1000,
                  false,  // ULPFEC disabled.
                  false,  // FlexFEC disabled.
                  ""};
  params.screenshare = {true, flags::SlideChangeInterval(),
                        flags::ScrollDuration(), flags::Slides()};
  params.analyzer = {"screenshare", 0.0, 0.0, flags::DurationSecs(),
      flags::OutputFilename(), flags::GraphTitle()};
  params.pipe = pipe_config;
  params.logging = {flags::FLAG_logs, flags::RtcEventLogName(),
                    flags::RtpDumpName(), flags::EncodedFramePath()};

  if (flags::NumStreams() > 1 && flags::Stream0().empty() &&
      flags::Stream1().empty()) {
    params.ss.infer_streams = true;
  }

  std::vector<std::string> stream_descriptors;
  stream_descriptors.push_back(flags::Stream0());
  stream_descriptors.push_back(flags::Stream1());
  std::vector<std::string> SL_descriptors;
  SL_descriptors.push_back(flags::SL0());
  SL_descriptors.push_back(flags::SL1());
  VideoQualityTest::FillScalabilitySettings(
      &params, stream_descriptors, flags::NumStreams(), flags::SelectedStream(),
      flags::NumSpatialLayers(), flags::SelectedSL(), SL_descriptors);

  VideoQualityTest test;
  if (flags::DurationSecs()) {
    test.RunWithAnalyzer(params);
  } else {
    test.RunWithRenderers(params);
  }
}
}  // namespace webrtc

int main(int argc, char* argv[]) {
  ::testing::InitGoogleTest(&argc, argv);
  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
  if (webrtc::flags::FLAG_help) {
    rtc::FlagList::Print(nullptr, false);
    return 0;
  }

  // InitFieldTrialsFromString needs a reference to an std::string instance,
  // with a scope that outlives the test.
  std::string field_trials = webrtc::flags::FLAG_force_fieldtrials;
  webrtc::test::InitFieldTrialsFromString(field_trials);

  webrtc::test::RunTest(webrtc::Loopback);
  return 0;
}