/*
 *  Copyright (c) 2011 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 "receiver_tests.h"
#include "video_coding.h"
#include "trace.h"
#include "../source/event.h"
#include "../source/internal_defines.h"
#include "timing.h"
#include "test_macros.h"
#include "test_util.h"

#include <cstdio>
#include <cstdlib>
#include <cmath>

using namespace webrtc;

float vcmFloatMax(float a, float b)
{
    return a > b ? a : b;
}

float vcmFloatMin(float a, float b)
{
    return a < b ? a : b;
}

double const pi = 4*std::atan(1.0);

class GaussDist
{
public:
    static float RandValue(float m, float stdDev) // returns a single normally distributed number
    {
        float r1 = static_cast<float>((std::rand() + 1.0)/(RAND_MAX + 1.0)); // gives equal distribution in (0, 1]
        float r2 = static_cast<float>((std::rand() + 1.0)/(RAND_MAX + 1.0));
        return m + stdDev * static_cast<float>(std::sqrt(-2*std::log(r1))*std::cos(2*pi*r2));
    }
};

int ReceiverTimingTests(CmdArgs& args)
{
    // Make sure this test is never executed with simulated events.
#if defined(EVENT_DEBUG)
    return -1;
#endif

    // Set up trace
    Trace::CreateTrace();
    Trace::SetTraceFile((test::OutputPath() + "receiverTestTrace.txt").c_str());
    Trace::SetLevelFilter(webrtc::kTraceAll);

    // A static random seed
    srand(0);

    TickTimeBase clock;
    VCMTiming timing(&clock);
    float clockInMs = 0.0;
    WebRtc_UWord32 waitTime = 0;
    WebRtc_UWord32 jitterDelayMs = 0;
    WebRtc_UWord32 maxDecodeTimeMs = 0;
    WebRtc_UWord32 timeStamp = 0;

    timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));

    timing.UpdateCurrentDelay(timeStamp);

    timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));

    timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
    jitterDelayMs = 20;
    timing.SetRequiredDelay(jitterDelayMs);
    timing.UpdateCurrentDelay(timeStamp);
    waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
        static_cast<WebRtc_Word64>(clockInMs + 0.5));
    // First update initializes the render time. Since we have no decode delay
    // we get waitTime = renderTime - now - renderDelay = jitter
    TEST(waitTime == jitterDelayMs);

    jitterDelayMs += VCMTiming::kDelayMaxChangeMsPerS + 10;
    timeStamp += 90000;
    clockInMs += 1000.0f;
    timing.SetRequiredDelay(jitterDelayMs);
    timing.UpdateCurrentDelay(timeStamp);
    waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
        static_cast<WebRtc_Word64>(clockInMs + 0.5));
    // Since we gradually increase the delay we only get
    // 100 ms every second.
    TEST(waitTime == jitterDelayMs - 10);

    timeStamp += 90000;
    clockInMs += 1000.0;
    timing.UpdateCurrentDelay(timeStamp);
    waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
        static_cast<WebRtc_Word64>(clockInMs + 0.5));
    TEST(waitTime == jitterDelayMs);

    // 300 incoming frames without jitter, verify that this gives the exact wait time
    for (int i=0; i < 300; i++)
    {
        clockInMs += 1000.0f/30.0f;
        timeStamp += 3000;
        timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
    }
    timing.UpdateCurrentDelay(timeStamp);
    waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
        static_cast<WebRtc_Word64>(clockInMs + 0.5));
    TEST(waitTime == jitterDelayMs);

    // Add decode time estimates
    for (int i=0; i < 10; i++)
    {
        WebRtc_Word64 startTimeMs = static_cast<WebRtc_Word64>(clockInMs + 0.5);
        clockInMs += 10.0f;
        timing.StopDecodeTimer(timeStamp, startTimeMs, static_cast<WebRtc_Word64>(clockInMs + 0.5));
        timeStamp += 3000;
        clockInMs += 1000.0f/30.0f - 10.0f;
        timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
    }
    maxDecodeTimeMs = 10;
    timing.SetRequiredDelay(jitterDelayMs);
    clockInMs += 1000.0f;
    timeStamp += 90000;
    timing.UpdateCurrentDelay(timeStamp);
    waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
        static_cast<WebRtc_Word64>(clockInMs + 0.5));
    TEST(waitTime == jitterDelayMs);

    WebRtc_UWord32 totalDelay1 = timing.TargetVideoDelay();
    WebRtc_UWord32 minTotalDelayMs = 200;
    timing.SetMinimumTotalDelay(minTotalDelayMs);
    clockInMs += 5000.0f;
    timeStamp += 5*90000;
    timing.UpdateCurrentDelay(timeStamp);
    waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
        static_cast<WebRtc_Word64>(clockInMs + 0.5));
    WebRtc_UWord32 totalDelay2 = timing.TargetVideoDelay();
    // We should at least have minTotalDelayMs - decodeTime (10) - renderTime (10) to wait
    TEST(waitTime == minTotalDelayMs - maxDecodeTimeMs - 10);
    // The total video delay should not increase with the extra delay,
    // the extra delay should be independent.
    TEST(totalDelay1 == totalDelay2);

    // Reset min total delay
    timing.SetMinimumTotalDelay(0);
    clockInMs += 5000.0f;
    timeStamp += 5*90000;
    timing.UpdateCurrentDelay(timeStamp);

    // A sudden increase in timestamp of 2.1 seconds
    clockInMs += 1000.0f/30.0f;
    timeStamp += static_cast<WebRtc_UWord32>(2.1*90000 + 0.5);
    WebRtc_Word64 ret = timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
    TEST(ret == -1);
    timing.Reset();

    // This test produces a trace which can be parsed with plotTimingTest.m. The plot
    // can be used to see that the timing is reasonable under noise, and that the
    // gradual transition between delays works as expected.
    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,  "Stochastic test 1");

    jitterDelayMs = 60;
    maxDecodeTimeMs = 10;

    timeStamp = static_cast<WebRtc_UWord32>(-10000); // To produce a wrap
    clockInMs = 10000.0f;
    timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));

    float noise = 0.0f;
    for (int i=0; i < 1400; i++)
    {
        if (i == 400)
        {
            jitterDelayMs = 30;
        }
        else if (i == 700)
        {
            jitterDelayMs = 100;
        }
        else if (i == 1000)
        {
            minTotalDelayMs = 200;
            timing.SetMinimumTotalDelay(minTotalDelayMs);
        }
        else if (i == 1200)
        {
            minTotalDelayMs = 0;
            timing.SetMinimumTotalDelay(minTotalDelayMs);
        }
        WebRtc_Word64 startTimeMs = static_cast<WebRtc_Word64>(clockInMs + 0.5);
        noise = vcmFloatMin(vcmFloatMax(GaussDist::RandValue(0, 2), -10.0f), 30.0f);
        clockInMs += 10.0f;
        timing.StopDecodeTimer(timeStamp, startTimeMs, static_cast<WebRtc_Word64>(clockInMs + noise + 0.5));
        timeStamp += 3000;
        clockInMs += 1000.0f/30.0f - 10.0f;
        noise = vcmFloatMin(vcmFloatMax(GaussDist::RandValue(0, 8), -15.0f), 15.0f);
        timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + noise + 0.5));
        timing.SetRequiredDelay(jitterDelayMs);
        timing.UpdateCurrentDelay(timeStamp);
        waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
            static_cast<WebRtc_Word64>(clockInMs + 0.5));

        WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,  "timeStamp=%u clock=%u maxWaitTime=%u", timeStamp,
            static_cast<WebRtc_UWord32>(clockInMs + 0.5), waitTime);

        WebRtc_Word64 renderTimeMs = timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));

        WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,
                   "timeStamp=%u renderTime=%u",
                   timeStamp,
                   MaskWord64ToUWord32(renderTimeMs));
    }
    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,  "End Stochastic test 1");

    printf("\nVCM Timing Test: \n\n%i tests completed\n", vcmMacrosTests);
    if (vcmMacrosErrors > 0)
    {
        printf("%i FAILED\n\n", vcmMacrosErrors);
    }
    else
    {
        printf("ALL PASSED\n\n");
    }

    Trace::ReturnTrace();
    return 0;
}