/*
 *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/absolute_capture_time_sender.h"

#include <algorithm>
#include <cstdint>
#include <optional>
#include <span>

#include "api/rtp_headers.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/ntp_time.h"

namespace webrtc {

static_assert(
    AbsoluteCaptureTimeInterpolator::kInterpolationMaxInterval >=
        AbsoluteCaptureTimeSender::kInterpolationMaxInterval,
    "Receivers should be as willing to interpolate timestamps as senders.");

AbsoluteCaptureTimeSender::AbsoluteCaptureTimeSender(Clock* clock)
    : clock_(clock) {}

uint32_t AbsoluteCaptureTimeSender::GetSource(uint32_t ssrc,
                                              std::span<const uint32_t> csrcs) {
  return AbsoluteCaptureTimeInterpolator::GetSource(ssrc, csrcs);
}

std::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
    uint32_t source,
    uint32_t rtp_timestamp,
    int rtp_clock_frequency_hz,
    NtpTime absolute_capture_time,
    std::optional<int64_t> estimated_capture_clock_offset,
    bool force) {
  Timestamp send_time = clock_->CurrentTime();
  if (!(force || ShouldSendExtension(
                     send_time, source, rtp_timestamp, rtp_clock_frequency_hz,
                     absolute_capture_time, estimated_capture_clock_offset))) {
    return std::nullopt;
  }

  last_source_ = source;
  last_rtp_timestamp_ = rtp_timestamp;
  last_rtp_clock_frequency_hz_ = rtp_clock_frequency_hz;
  last_absolute_capture_time_ = absolute_capture_time;
  last_estimated_capture_clock_offset_ = estimated_capture_clock_offset;
  last_send_time_ = send_time;

  return AbsoluteCaptureTime{
      .absolute_capture_timestamp = uint64_t{absolute_capture_time},
      .estimated_capture_clock_offset = estimated_capture_clock_offset,
  };
}

bool AbsoluteCaptureTimeSender::ShouldSendExtension(
    Timestamp send_time,
    uint32_t source,
    uint32_t rtp_timestamp,
    int rtp_clock_frequency_hz,
    NtpTime absolute_capture_time,
    std::optional<int64_t> estimated_capture_clock_offset) const {
  // Should if the last sent extension is too old, in particular if we've never
  // sent anything before.
  if (send_time - last_send_time_ > kInterpolationMaxInterval) {
    return true;
  }

  // Should if the source has changed.
  if (last_source_ != source) {
    return true;
  }

  // Should if the RTP clock frequency has changed.
  if (last_rtp_clock_frequency_hz_ != rtp_clock_frequency_hz) {
    return true;
  }

  // Should if the RTP clock frequency is invalid.
  if (rtp_clock_frequency_hz <= 0) {
    return true;
  }

  // Should if the estimated capture clock offset has changed.
  if (last_estimated_capture_clock_offset_ != estimated_capture_clock_offset) {
    return true;
  }

  // Should if interpolation would introduce too much error.
  const uint64_t interpolated_absolute_capture_timestamp =
      AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
          rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_,
          uint64_t{last_absolute_capture_time_});
  const uint64_t absolute_capture_timestamp = uint64_t{absolute_capture_time};
  const int64_t interpolation_error_ms = UQ32x32ToInt64Ms(std::min(
      interpolated_absolute_capture_timestamp - absolute_capture_timestamp,
      absolute_capture_timestamp - interpolated_absolute_capture_timestamp));
  if (interpolation_error_ms > kInterpolationMaxError.ms()) {
    return true;
  }

  return false;
}

}  // namespace webrtc