// Copyright 2014 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "media/cast/test/receiver/frame_receiver.h"

#include <algorithm>
#include <string>
#include <utility>

#include "base/big_endian.h"
#include "base/bind.h"
#include "base/logging.h"
#include "base/numerics/safe_conversions.h"
#include "media/cast/cast_config.h"
#include "media/cast/cast_environment.h"
#include "media/cast/constants.h"
#include "media/cast/net/rtcp/rtcp_utility.h"

namespace {

const int kMinSchedulingDelayMs = 1;

media::cast::RtcpTimeData CreateRtcpTimeData(base::TimeTicks now) {
  media::cast::RtcpTimeData ret;
  ret.timestamp = now;
  media::cast::ConvertTimeTicksToNtp(now, &ret.ntp_seconds, &ret.ntp_fraction);
  return ret;
}

}  // namespace

namespace media {
namespace cast {

FrameReceiver::FrameReceiver(
    const scoped_refptr<CastEnvironment>& cast_environment,
    const FrameReceiverConfig& config,
    EventMediaType event_media_type,
    CastTransport* const transport)
    : cast_environment_(cast_environment),
      transport_(transport),
      packet_parser_(
          config.sender_ssrc,
          config.rtp_payload_type <= RtpPayloadType::AUDIO_LAST ? 127 : 96),
      stats_(cast_environment->Clock()),
      event_media_type_(event_media_type),
      event_subscriber_(kReceiverRtcpEventHistorySize, event_media_type),
      rtp_timebase_(config.rtp_timebase),
      target_playout_delay_(base::Milliseconds(config.rtp_max_delay_ms)),
      expected_frame_duration_(base::Seconds(1.0 / config.target_frame_rate)),
      reports_are_scheduled_(false),
      framer_(cast_environment->Clock(),
              this,
              config.sender_ssrc,
              true,
              static_cast<int>(config.rtp_max_delay_ms *
                               config.target_frame_rate / 1000)),
      rtcp_(cast_environment_->Clock(),
            config.receiver_ssrc,
            config.sender_ssrc),
      is_waiting_for_consecutive_frame_(false),
      lip_sync_drift_(ClockDriftSmoother::GetDefaultTimeConstant()) {
  transport_->AddValidRtpReceiver(config.sender_ssrc, config.receiver_ssrc);
  DCHECK_GT(config.rtp_max_delay_ms, 0);
  DCHECK_GT(config.target_frame_rate, 0);
  decryptor_.Initialize(config.aes_key, config.aes_iv_mask);
  cast_environment_->logger()->Subscribe(&event_subscriber_);
}

FrameReceiver::~FrameReceiver() {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
  cast_environment_->logger()->Unsubscribe(&event_subscriber_);
}

void FrameReceiver::RequestEncodedFrame(ReceiveEncodedFrameCallback callback) {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
  frame_request_queue_.push_back(std::move(callback));
  EmitAvailableEncodedFrames();
}

bool FrameReceiver::ProcessPacket(std::unique_ptr<Packet> packet) {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

  if (IsRtcpPacket(&packet->front(), packet->size())) {
    rtcp_.IncomingRtcpPacket(&packet->front(), packet->size());
  } else {
    RtpCastHeader rtp_header;
    const uint8_t* payload_data;
    size_t payload_size;
    if (!packet_parser_.ParsePacket(&packet->front(), packet->size(),
                                    &rtp_header, &payload_data,
                                    &payload_size)) {
      return false;
    }

    ProcessParsedPacket(rtp_header, payload_data, payload_size);
    stats_.UpdateStatistics(rtp_header, rtp_timebase_);
  }

  if (!reports_are_scheduled_) {
    ScheduleNextRtcpReport();
    ScheduleNextCastMessage();
    reports_are_scheduled_ = true;
  }

  return true;
}

base::WeakPtr<FrameReceiver> FrameReceiver::AsWeakPtr() {
  return weak_factory_.GetWeakPtr();
}

void FrameReceiver::ProcessParsedPacket(const RtpCastHeader& rtp_header,
                                        const uint8_t* payload_data,
                                        size_t payload_size) {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

  const base::TimeTicks now = cast_environment_->Clock()->NowTicks();

  frame_id_to_rtp_timestamp_[rtp_header.frame_id.lower_8_bits()] =
      rtp_header.rtp_timestamp;

  std::unique_ptr<PacketEvent> receive_event(new PacketEvent());
  receive_event->timestamp = now;
  receive_event->type = PACKET_RECEIVED;
  receive_event->media_type = event_media_type_;
  receive_event->rtp_timestamp = rtp_header.rtp_timestamp;
  receive_event->frame_id = rtp_header.frame_id;
  receive_event->packet_id = rtp_header.packet_id;
  receive_event->max_packet_id = rtp_header.max_packet_id;
  receive_event->size = base::checked_cast<uint32_t>(payload_size);
  cast_environment_->logger()->DispatchPacketEvent(std::move(receive_event));

  bool duplicate = false;
  const bool complete =
      framer_.InsertPacket(payload_data, payload_size, rtp_header, &duplicate);

  // Duplicate packets are ignored.
  if (duplicate)
    return;

  // Update lip-sync values upon receiving the first packet of each frame, or if
  // they have never been set yet.
  if (rtp_header.packet_id == 0 || lip_sync_reference_time_.is_null()) {
    RtpTimeTicks fresh_sync_rtp;
    base::TimeTicks fresh_sync_reference;
    if (!rtcp_.GetLatestLipSyncTimes(&fresh_sync_rtp, &fresh_sync_reference)) {
      // HACK: The sender should have provided Sender Reports before the first
      // frame was sent.  However, the spec does not currently require this.
      // Therefore, when the data is missing, the local clock is used to
      // generate reference timestamps.
      VLOG(2) << "Lip sync info missing.  Falling-back to local clock.";
      fresh_sync_rtp = rtp_header.rtp_timestamp;
      fresh_sync_reference = now;
    }
    // |lip_sync_reference_time_| is always incremented according to the time
    // delta computed from the difference in RTP timestamps.  Then,
    // |lip_sync_drift_| accounts for clock drift and also smoothes-out any
    // sudden/discontinuous shifts in the series of reference time values.
    if (lip_sync_reference_time_.is_null()) {
      lip_sync_reference_time_ = fresh_sync_reference;
    } else {
      // Note: It's okay for the conversion ToTimeDelta() to be approximate
      // because |lip_sync_drift_| will account for accumulated errors.
      lip_sync_reference_time_ +=
          (fresh_sync_rtp - lip_sync_rtp_timestamp_).ToTimeDelta(rtp_timebase_);
    }
    lip_sync_rtp_timestamp_ = fresh_sync_rtp;
    lip_sync_drift_.Update(now,
                           fresh_sync_reference - lip_sync_reference_time_);
  }

  // Another frame is complete from a non-duplicate packet.  Attempt to emit
  // more frames to satisfy enqueued requests.
  if (complete)
    EmitAvailableEncodedFrames();
}

void FrameReceiver::CastFeedback(const RtcpCastMessage& cast_message) {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

  base::TimeTicks now = cast_environment_->Clock()->NowTicks();
  RtpTimeTicks rtp_timestamp =
      frame_id_to_rtp_timestamp_[cast_message.ack_frame_id.lower_8_bits()];

  std::unique_ptr<FrameEvent> ack_sent_event(new FrameEvent());
  ack_sent_event->timestamp = now;
  ack_sent_event->type = FRAME_ACK_SENT;
  ack_sent_event->media_type = event_media_type_;
  ack_sent_event->rtp_timestamp = rtp_timestamp;
  ack_sent_event->frame_id = cast_message.ack_frame_id;
  cast_environment_->logger()->DispatchFrameEvent(std::move(ack_sent_event));

  ReceiverRtcpEventSubscriber::RtcpEvents rtcp_events;
  event_subscriber_.GetRtcpEventsWithRedundancy(&rtcp_events);
  SendRtcpReport(rtcp_.local_ssrc(), rtcp_.remote_ssrc(),
                 CreateRtcpTimeData(now), &cast_message, nullptr,
                 target_playout_delay_, &rtcp_events, nullptr);
}

void FrameReceiver::EmitAvailableEncodedFrames() {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

  while (!frame_request_queue_.empty()) {
    // Attempt to peek at the next completed frame from the |framer_|.
    // TODO(miu): We should only be peeking at the metadata, and not copying the
    // payload yet!  Or, at least, peek using a StringPiece instead of a copy.
    std::unique_ptr<EncodedFrame> encoded_frame(new EncodedFrame());
    bool is_consecutively_next_frame = false;
    bool have_multiple_complete_frames = false;
    if (!framer_.GetEncodedFrame(encoded_frame.get(),
                                 &is_consecutively_next_frame,
                                 &have_multiple_complete_frames)) {
      VLOG(1) << "Wait for more packets to produce a completed frame.";
      return;  // ProcessParsedPacket() will invoke this method in the future.
    }

    const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
    const base::TimeTicks playout_time = GetPlayoutTime(*encoded_frame);

    // If we have multiple decodable frames, and the current frame is
    // too old, then skip it and decode the next frame instead.
    if (have_multiple_complete_frames && now > playout_time) {
      framer_.ReleaseFrame(encoded_frame->frame_id);
      continue;
    }

    // If |framer_| has a frame ready that is out of sequence, examine the
    // playout time to determine whether it's acceptable to continue, thereby
    // skipping one or more frames.  Skip if the missing frame wouldn't complete
    // playing before the start of playback of the available frame.
    if (!is_consecutively_next_frame) {
      // This assumes that decoding takes as long as playing, which might
      // not be true.
      const base::TimeTicks earliest_possible_end_time_of_missing_frame =
          now + expected_frame_duration_ * 2;
      if (earliest_possible_end_time_of_missing_frame < playout_time) {
        VLOG(1) << "Wait for next consecutive frame instead of skipping.";
        if (!is_waiting_for_consecutive_frame_) {
          is_waiting_for_consecutive_frame_ = true;
          cast_environment_->PostDelayedTask(
              CastEnvironment::MAIN, FROM_HERE,
              base::BindOnce(
                  &FrameReceiver::EmitAvailableEncodedFramesAfterWaiting,
                  AsWeakPtr()),
              playout_time - now);
        }
        return;
      }
    }

    // At this point, we have the complete next frame, or a decodable
    // frame from somewhere later in the stream, AND we have given up
    // on waiting for any frames in between, so now we can ACK the frame.
    framer_.AckFrame(encoded_frame->frame_id);

    // Decrypt the payload data in the frame, if crypto is being used.
    if (decryptor_.is_activated()) {
      std::string decrypted_data;
      if (!decryptor_.Decrypt(encoded_frame->frame_id, encoded_frame->data,
                              &decrypted_data)) {
        // Decryption failed.  Give up on this frame.
        framer_.ReleaseFrame(encoded_frame->frame_id);
        continue;
      }
      encoded_frame->data.swap(decrypted_data);
    }

    // At this point, we have a decrypted EncodedFrame ready to be emitted.
    encoded_frame->reference_time = playout_time;
    framer_.ReleaseFrame(encoded_frame->frame_id);
    if (encoded_frame->new_playout_delay_ms) {
      target_playout_delay_ =
          base::Milliseconds(encoded_frame->new_playout_delay_ms);
    }
    cast_environment_->PostTask(
        CastEnvironment::MAIN, FROM_HERE,
        base::BindOnce(&FrameReceiver::EmitOneFrame, AsWeakPtr(),
                       std::move(*frame_request_queue_.begin()),
                       std::move(encoded_frame)));
    frame_request_queue_.pop_front();
  }
}

void FrameReceiver::EmitAvailableEncodedFramesAfterWaiting() {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
  DCHECK(is_waiting_for_consecutive_frame_);
  is_waiting_for_consecutive_frame_ = false;
  EmitAvailableEncodedFrames();
}

void FrameReceiver::EmitOneFrame(
    ReceiveEncodedFrameCallback callback,
    std::unique_ptr<EncodedFrame> encoded_frame) const {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
  if (!callback.is_null())
    std::move(callback).Run(std::move(encoded_frame));
}

base::TimeTicks FrameReceiver::GetPlayoutTime(const EncodedFrame& frame) const {
  base::TimeDelta target_playout_delay = target_playout_delay_;
  if (frame.new_playout_delay_ms) {
    target_playout_delay = base::Milliseconds(frame.new_playout_delay_ms);
  }
  return lip_sync_reference_time_ + lip_sync_drift_.Current() +
         (frame.rtp_timestamp - lip_sync_rtp_timestamp_)
             .ToTimeDelta(rtp_timebase_) +
         target_playout_delay;
}

void FrameReceiver::ScheduleNextCastMessage() {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
  base::TimeTicks send_time;
  framer_.TimeToSendNextCastMessage(&send_time);
  base::TimeDelta time_to_send =
      send_time - cast_environment_->Clock()->NowTicks();
  time_to_send =
      std::max(time_to_send, base::Milliseconds(kMinSchedulingDelayMs));
  cast_environment_->PostDelayedTask(
      CastEnvironment::MAIN, FROM_HERE,
      base::BindOnce(&FrameReceiver::SendNextCastMessage, AsWeakPtr()),
      time_to_send);
}

void FrameReceiver::SendNextCastMessage() {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
  framer_.SendCastMessage();  // Will only send a message if it is time.
  ScheduleNextCastMessage();
}

void FrameReceiver::ScheduleNextRtcpReport() {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));

  cast_environment_->PostDelayedTask(
      CastEnvironment::MAIN, FROM_HERE,
      base::BindOnce(&FrameReceiver::SendNextRtcpReport, AsWeakPtr()),
      base::Milliseconds(kRtcpReportIntervalMs));
}

void FrameReceiver::SendNextRtcpReport() {
  DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
  const base::TimeTicks now = cast_environment_->Clock()->NowTicks();
  RtpReceiverStatistics stats = stats_.GetStatistics();
  SendRtcpReport(rtcp_.local_ssrc(), rtcp_.remote_ssrc(),
                 CreateRtcpTimeData(now), nullptr, nullptr, base::TimeDelta(),
                 nullptr, &stats);
  ScheduleNextRtcpReport();
}

void FrameReceiver::SendRtcpReport(
    uint32_t rtp_receiver_ssrc,
    uint32_t rtp_sender_ssrc,
    const RtcpTimeData& time_data,
    const RtcpCastMessage* cast_message,
    const RtcpPliMessage* pli_message,
    base::TimeDelta target_delay,
    const ReceiverRtcpEventSubscriber::RtcpEvents* rtcp_events,
    const RtpReceiverStatistics* rtp_receiver_statistics) {
  transport_->InitializeRtpReceiverRtcpBuilder(rtp_receiver_ssrc, time_data);
  RtcpReportBlock report_block;
  if (rtp_receiver_statistics) {
    report_block.remote_ssrc = 0;  // Not needed to set send side.
    report_block.media_ssrc =
        rtp_sender_ssrc;  // SSRC of the RTP packet sender.
    report_block.fraction_lost = rtp_receiver_statistics->fraction_lost;
    report_block.cumulative_lost = rtp_receiver_statistics->cumulative_lost;
    report_block.extended_high_sequence_number =
        rtp_receiver_statistics->extended_high_sequence_number;
    report_block.jitter = rtp_receiver_statistics->jitter;
    report_block.last_sr = rtcp_.last_report_truncated_ntp();
    base::TimeTicks last_report_received_time =
        rtcp_.time_last_report_received();
    if (!last_report_received_time.is_null()) {
      uint32_t delay_seconds = 0;
      uint32_t delay_fraction = 0;
      base::TimeDelta delta = time_data.timestamp - last_report_received_time;
      ConvertTimeToFractions(delta.InMicroseconds(), &delay_seconds,
                             &delay_fraction);
      report_block.delay_since_last_sr =
          ConvertToNtpDiff(delay_seconds, delay_fraction);
    } else {
      report_block.delay_since_last_sr = 0;
    }
    transport_->AddRtpReceiverReport(report_block);
  }
  if (cast_message)
    transport_->AddCastFeedback(*cast_message, target_delay);
  if (pli_message)
    transport_->AddPli(*pli_message);
  if (rtcp_events)
    transport_->AddRtcpEvents(*rtcp_events);
  transport_->SendRtcpFromRtpReceiver();
}

}  // namespace cast
}  // namespace media