/*
 *  Copyright (c) 2013 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 "video/video_receive_stream.h"

#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include <set>
#include <string>
#include <utility>

#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/video/encoded_image.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "call/rtx_receive_stream.h"
#include "common_video/include/incoming_video_stream.h"
#include "media/base/h264_profile_level_id.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "modules/video_coding/include/video_coding_defines.h"
#include "modules/video_coding/include/video_error_codes.h"
#include "modules/video_coding/jitter_estimator.h"
#include "modules/video_coding/timing.h"
#include "modules/video_coding/utility/vp8_header_parser.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/keyframe_interval_settings.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/platform_file.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/thread_registry.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
#include "video/call_stats.h"
#include "video/frame_dumping_decoder.h"
#include "video/receive_statistics_proxy.h"

namespace webrtc {

namespace {
constexpr int kMinBaseMinimumDelayMs = 0;
constexpr int kMaxBaseMinimumDelayMs = 10000;

constexpr int kMaxWaitForKeyFrameMs = 200;
constexpr int kMaxWaitForFrameMs = 3000;

VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
  VideoCodec codec;
  memset(&codec, 0, sizeof(codec));

  codec.plType = decoder.payload_type;
  codec.codecType = PayloadStringToCodecType(decoder.video_format.name);

  if (codec.codecType == kVideoCodecVP8) {
    *(codec.VP8()) = VideoEncoder::GetDefaultVp8Settings();
  } else if (codec.codecType == kVideoCodecVP9) {
    *(codec.VP9()) = VideoEncoder::GetDefaultVp9Settings();
  } else if (codec.codecType == kVideoCodecH264) {
    *(codec.H264()) = VideoEncoder::GetDefaultH264Settings();
    codec.H264()->profile =
        H264::ParseSdpProfileLevelId(decoder.video_format.parameters)->profile;
  } else if (codec.codecType == kVideoCodecMultiplex) {
    VideoReceiveStream::Decoder associated_decoder = decoder;
    associated_decoder.video_format =
        SdpVideoFormat(CodecTypeToPayloadString(kVideoCodecVP9));
    VideoCodec associated_codec = CreateDecoderVideoCodec(associated_decoder);
    associated_codec.codecType = kVideoCodecMultiplex;
    return associated_codec;
  }

  codec.width = 320;
  codec.height = 180;
  const int kDefaultStartBitrate = 300;
  codec.startBitrate = codec.minBitrate = codec.maxBitrate =
      kDefaultStartBitrate;

  return codec;
}

// Video decoder class to be used for unknown codecs. Doesn't support decoding
// but logs messages to LS_ERROR.
class NullVideoDecoder : public webrtc::VideoDecoder {
 public:
  int32_t InitDecode(const webrtc::VideoCodec* codec_settings,
                     int32_t number_of_cores) override {
    RTC_LOG(LS_ERROR) << "Can't initialize NullVideoDecoder.";
    return WEBRTC_VIDEO_CODEC_OK;
  }

  int32_t Decode(const webrtc::EncodedImage& input_image,
                 bool missing_frames,
                 const webrtc::CodecSpecificInfo* codec_specific_info,
                 int64_t render_time_ms) override {
    RTC_LOG(LS_ERROR) << "The NullVideoDecoder doesn't support decoding.";
    return WEBRTC_VIDEO_CODEC_OK;
  }

  int32_t RegisterDecodeCompleteCallback(
      webrtc::DecodedImageCallback* callback) override {
    RTC_LOG(LS_ERROR)
        << "Can't register decode complete callback on NullVideoDecoder.";
    return WEBRTC_VIDEO_CODEC_OK;
  }

  int32_t Release() override { return WEBRTC_VIDEO_CODEC_OK; }

  const char* ImplementationName() const override { return "NullVideoDecoder"; }
};

// Inherit video_coding::EncodedFrame, which is the class used by
// video_coding::FrameBuffer and other components in the receive pipeline. It's
// a subclass of EncodedImage, and it always owns the buffer.
class EncodedFrameForMediaTransport : public video_coding::EncodedFrame {
 public:
  explicit EncodedFrameForMediaTransport(
      MediaTransportEncodedVideoFrame frame) {
    // TODO(nisse): This is ugly. We copy the EncodedImage (a base class of
    // ours, in several steps), to get all the meta data. We should be using
    // std::move in some way. Then we also need to handle the case of an unowned
    // buffer, in which case we need to make an owned copy.
    *static_cast<class EncodedImage*>(this) = frame.encoded_image();

    // If we don't already own the buffer, make a copy.
    Retain();

    _payloadType = static_cast<uint8_t>(frame.payload_type());

    // TODO(nisse): frame_id and picture_id are probably not the same thing. For
    // a single layer, this should be good enough.
    id.picture_id = frame.frame_id();
    id.spatial_layer = frame.encoded_image().SpatialIndex().value_or(0);
    num_references = std::min(static_cast<size_t>(kMaxFrameReferences),
                              frame.referenced_frame_ids().size());
    for (size_t i = 0; i < num_references; i++) {
      references[i] = frame.referenced_frame_ids()[i];
    }
  }

  // TODO(nisse): Implement. Not sure how they are used.
  int64_t ReceivedTime() const override { return 0; }
  int64_t RenderTime() const override { return 0; }
};

// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
// Maximum time between frames before resetting the FrameBuffer to avoid RTP
// timestamps wraparound to affect FrameBuffer.
constexpr int kInactiveStreamThresholdMs = 600000;  //  10 minutes.

}  // namespace

namespace internal {

VideoReceiveStream::VideoReceiveStream(
    TaskQueueFactory* task_queue_factory,
    RtpStreamReceiverControllerInterface* receiver_controller,
    int num_cpu_cores,
    PacketRouter* packet_router,
    VideoReceiveStream::Config config,
    ProcessThread* process_thread,
    CallStats* call_stats,
    Clock* clock,
    VCMTiming* timing)
    : task_queue_factory_(task_queue_factory),
      transport_adapter_(config.rtcp_send_transport),
      config_(std::move(config)),
      num_cpu_cores_(num_cpu_cores),
      process_thread_(process_thread),
      clock_(clock),
      decode_thread_(&DecodeThreadFunction,
                     this,
                     "DecodingThread",
                     rtc::kHighestPriority),
      call_stats_(call_stats),
      stats_proxy_(&config_, clock_),
      rtp_receive_statistics_(
          ReceiveStatistics::Create(clock_, &stats_proxy_, &stats_proxy_)),
      timing_(timing),
      video_receiver_(clock_, timing_.get()),
      rtp_video_stream_receiver_(clock_,
                                 &transport_adapter_,
                                 call_stats,
                                 packet_router,
                                 &config_,
                                 rtp_receive_statistics_.get(),
                                 &stats_proxy_,
                                 process_thread_,
                                 this,  // NackSender
                                 this,  // KeyFrameRequestSender
                                 this,  // OnCompleteFrameCallback
                                 config_.frame_decryptor),
      rtp_stream_sync_(this),
      max_wait_for_keyframe_ms_(KeyframeIntervalSettings::ParseFromFieldTrials()
                                    .MaxWaitForKeyframeMs()
                                    .value_or(kMaxWaitForKeyFrameMs)),
      max_wait_for_frame_ms_(KeyframeIntervalSettings::ParseFromFieldTrials()
                                 .MaxWaitForFrameMs()
                                 .value_or(kMaxWaitForFrameMs)) {
  RTC_LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();

  RTC_DCHECK(config_.renderer);
  RTC_DCHECK(process_thread_);
  RTC_DCHECK(call_stats_);

  module_process_sequence_checker_.Detach();
  network_sequence_checker_.Detach();

  RTC_DCHECK(!config_.decoders.empty());
  std::set<int> decoder_payload_types;
  for (const Decoder& decoder : config_.decoders) {
    RTC_CHECK(decoder.decoder_factory);
    RTC_CHECK(decoder_payload_types.find(decoder.payload_type) ==
              decoder_payload_types.end())
        << "Duplicate payload type (" << decoder.payload_type
        << ") for different decoders.";
    decoder_payload_types.insert(decoder.payload_type);
  }

  video_receiver_.SetRenderDelay(config_.render_delay_ms);

  jitter_estimator_.reset(new VCMJitterEstimator(clock_));
  frame_buffer_.reset(new video_coding::FrameBuffer(
      clock_, jitter_estimator_.get(), timing_.get(), &stats_proxy_));

  process_thread_->RegisterModule(&rtp_stream_sync_, RTC_FROM_HERE);

  if (config_.media_transport) {
    config_.media_transport->SetReceiveVideoSink(this);
    config_.media_transport->AddRttObserver(this);
  } else {
    // Register with RtpStreamReceiverController.
    media_receiver_ = receiver_controller->CreateReceiver(
        config_.rtp.remote_ssrc, &rtp_video_stream_receiver_);
    if (config_.rtp.rtx_ssrc) {
      rtx_receive_stream_ = absl::make_unique<RtxReceiveStream>(
          &rtp_video_stream_receiver_, config.rtp.rtx_associated_payload_types,
          config_.rtp.remote_ssrc, rtp_receive_statistics_.get());
      rtx_receiver_ = receiver_controller->CreateReceiver(
          config_.rtp.rtx_ssrc, rtx_receive_stream_.get());
    } else {
      rtp_receive_statistics_->EnableRetransmitDetection(config.rtp.remote_ssrc,
                                                         true);
    }
  }
}

VideoReceiveStream::VideoReceiveStream(
    TaskQueueFactory* task_queue_factory,
    RtpStreamReceiverControllerInterface* receiver_controller,
    int num_cpu_cores,
    PacketRouter* packet_router,
    VideoReceiveStream::Config config,
    ProcessThread* process_thread,
    CallStats* call_stats,
    Clock* clock)
    : VideoReceiveStream(task_queue_factory,
                         receiver_controller,
                         num_cpu_cores,
                         packet_router,
                         std::move(config),
                         process_thread,
                         call_stats,
                         clock,
                         new VCMTiming(clock)) {}

VideoReceiveStream::~VideoReceiveStream() {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
  RTC_LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
  Stop();
  if (config_.media_transport) {
    config_.media_transport->SetReceiveVideoSink(nullptr);
    config_.media_transport->RemoveRttObserver(this);
  }
  process_thread_->DeRegisterModule(&rtp_stream_sync_);
}

void VideoReceiveStream::SignalNetworkState(NetworkState state) {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
  rtp_video_stream_receiver_.SignalNetworkState(state);
}

bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
  return rtp_video_stream_receiver_.DeliverRtcp(packet, length);
}

void VideoReceiveStream::SetSync(Syncable* audio_syncable) {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
  rtp_stream_sync_.ConfigureSync(audio_syncable);
}

void VideoReceiveStream::Start() {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);

  if (decode_thread_.IsRunning()) {
    return;
  }

  const bool protected_by_fec = config_.rtp.protected_by_flexfec ||
                                rtp_video_stream_receiver_.IsUlpfecEnabled();

  frame_buffer_->Start();

  if (rtp_video_stream_receiver_.IsRetransmissionsEnabled() &&
      protected_by_fec) {
    frame_buffer_->SetProtectionMode(kProtectionNackFEC);
  }

  transport_adapter_.Enable();
  rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
  if (config_.enable_prerenderer_smoothing) {
    incoming_video_stream_.reset(new IncomingVideoStream(
        task_queue_factory_, config_.render_delay_ms, this));
    renderer = incoming_video_stream_.get();
  } else {
    renderer = this;
  }

  for (const Decoder& decoder : config_.decoders) {
    std::unique_ptr<VideoDecoder> video_decoder =
        decoder.decoder_factory->LegacyCreateVideoDecoder(decoder.video_format,
                                                          config_.stream_id);
    // If we still have no valid decoder, we have to create a "Null" decoder
    // that ignores all calls. The reason we can get into this state is that the
    // old decoder factory interface doesn't have a way to query supported
    // codecs.
    if (!video_decoder) {
      video_decoder = absl::make_unique<NullVideoDecoder>();
    }

    std::string decoded_output_file =
        field_trial::FindFullName("WebRTC-DecoderDataDumpDirectory");
    // Because '/' can't be used inside a field trial parameter, we use ':'
    // instead.
    absl::c_replace(decoded_output_file, ':', '/');
    if (!decoded_output_file.empty()) {
      char filename_buffer[256];
      rtc::SimpleStringBuilder ssb(filename_buffer);
      ssb << decoded_output_file << "/webrtc_receive_stream_"
          << this->config_.rtp.remote_ssrc << "-" << rtc::TimeMicros()
          << ".ivf";
      video_decoder = absl::make_unique<FrameDumpingDecoder>(
          std::move(video_decoder), FileWrapper::OpenWriteOnly(ssb.str()));
    }

    video_decoders_.push_back(std::move(video_decoder));

    video_receiver_.RegisterExternalDecoder(video_decoders_.back().get(),
                                            decoder.payload_type);
    VideoCodec codec = CreateDecoderVideoCodec(decoder);
    rtp_video_stream_receiver_.AddReceiveCodec(codec,
                                               decoder.video_format.parameters);
    RTC_CHECK_EQ(VCM_OK, video_receiver_.RegisterReceiveCodec(
                             &codec, num_cpu_cores_, false));
  }

  RTC_DCHECK(renderer != nullptr);
  video_stream_decoder_.reset(new VideoStreamDecoder(
      &video_receiver_, &rtp_video_stream_receiver_,
      rtp_video_stream_receiver_.IsRetransmissionsEnabled(), protected_by_fec,
      &stats_proxy_, renderer));

  // Make sure we register as a stats observer *after* we've prepared the
  // |video_stream_decoder_|.
  call_stats_->RegisterStatsObserver(this);

  // NOTE: *Not* registering video_receiver_ on process_thread_. Its Process
  // method does nothing that is useful for us, since we no longer use the old
  // jitter buffer.

  // Start the decode thread
  video_receiver_.DecoderThreadStarting();
  stats_proxy_.DecoderThreadStarting();
  decode_thread_.Start();
  rtp_video_stream_receiver_.StartReceive();
}

void VideoReceiveStream::Stop() {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
  rtp_video_stream_receiver_.StopReceive();

  stats_proxy_.OnUniqueFramesCounted(
      rtp_video_stream_receiver_.GetUniqueFramesSeen());

  frame_buffer_->Stop();
  call_stats_->DeregisterStatsObserver(this);

  if (decode_thread_.IsRunning()) {
    // TriggerDecoderShutdown will release any waiting decoder thread and make
    // it stop immediately, instead of waiting for a timeout. Needs to be called
    // before joining the decoder thread.
    video_receiver_.TriggerDecoderShutdown();

    decode_thread_.Stop();
    video_receiver_.DecoderThreadStopped();
    stats_proxy_.DecoderThreadStopped();
    // Deregister external decoders so they are no longer running during
    // destruction. This effectively stops the VCM since the decoder thread is
    // stopped, the VCM is deregistered and no asynchronous decoder threads are
    // running.
    for (const Decoder& decoder : config_.decoders)
      video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type);
  }

  video_stream_decoder_.reset();
  incoming_video_stream_.reset();
  transport_adapter_.Disable();
}

VideoReceiveStream::Stats VideoReceiveStream::GetStats() const {
  return stats_proxy_.GetStats();
}

void VideoReceiveStream::AddSecondarySink(RtpPacketSinkInterface* sink) {
  rtp_video_stream_receiver_.AddSecondarySink(sink);
}

void VideoReceiveStream::RemoveSecondarySink(
    const RtpPacketSinkInterface* sink) {
  rtp_video_stream_receiver_.RemoveSecondarySink(sink);
}

bool VideoReceiveStream::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
  if (delay_ms < kMinBaseMinimumDelayMs || delay_ms > kMaxBaseMinimumDelayMs) {
    return false;
  }

  rtc::CritScope cs(&playout_delay_lock_);
  base_minimum_playout_delay_ms_ = delay_ms;
  UpdatePlayoutDelays();
  return true;
}

int VideoReceiveStream::GetBaseMinimumPlayoutDelayMs() const {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);

  rtc::CritScope cs(&playout_delay_lock_);
  return base_minimum_playout_delay_ms_;
}

// TODO(tommi): This method grabs a lock 6 times.
void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) {
  int64_t sync_offset_ms;
  double estimated_freq_khz;
  // TODO(tommi): GetStreamSyncOffsetInMs grabs three locks.  One inside the
  // function itself, another in GetChannel() and a third in
  // GetPlayoutTimestamp.  Seems excessive.  Anyhow, I'm assuming the function
  // succeeds most of the time, which leads to grabbing a fourth lock.
  if (rtp_stream_sync_.GetStreamSyncOffsetInMs(
          video_frame.timestamp(), video_frame.render_time_ms(),
          &sync_offset_ms, &estimated_freq_khz)) {
    // TODO(tommi): OnSyncOffsetUpdated grabs a lock.
    stats_proxy_.OnSyncOffsetUpdated(sync_offset_ms, estimated_freq_khz);
  }
  config_.renderer->OnFrame(video_frame);

  // TODO(tommi): OnRenderFrame grabs a lock too.
  stats_proxy_.OnRenderedFrame(video_frame);
}

void VideoReceiveStream::SendNack(
    const std::vector<uint16_t>& sequence_numbers) {
  rtp_video_stream_receiver_.RequestPacketRetransmit(sequence_numbers);
}

void VideoReceiveStream::RequestKeyFrame() {
  if (config_.media_transport) {
    config_.media_transport->RequestKeyFrame(config_.rtp.remote_ssrc);
  } else {
    rtp_video_stream_receiver_.RequestKeyFrame();
  }
}

void VideoReceiveStream::OnCompleteFrame(
    std::unique_ptr<video_coding::EncodedFrame> frame) {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&network_sequence_checker_);
  // TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
  int64_t time_now_ms = rtc::TimeMillis();
  if (last_complete_frame_time_ms_ > 0 &&
      time_now_ms - last_complete_frame_time_ms_ > kInactiveStreamThresholdMs) {
    frame_buffer_->Clear();
  }
  last_complete_frame_time_ms_ = time_now_ms;

  const PlayoutDelay& playout_delay = frame->EncodedImage().playout_delay_;
  if (playout_delay.min_ms >= 0) {
    rtc::CritScope cs(&playout_delay_lock_);
    frame_minimum_playout_delay_ms_ = playout_delay.min_ms;
    UpdatePlayoutDelays();
  }

  if (playout_delay.max_ms >= 0) {
    rtc::CritScope cs(&playout_delay_lock_);
    frame_maximum_playout_delay_ms_ = playout_delay.max_ms;
    UpdatePlayoutDelays();
  }

  int64_t last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame));
  if (last_continuous_pid != -1)
    rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
}

void VideoReceiveStream::OnData(uint64_t channel_id,
                                MediaTransportEncodedVideoFrame frame) {
  OnCompleteFrame(
      absl::make_unique<EncodedFrameForMediaTransport>(std::move(frame)));
}

void VideoReceiveStream::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&module_process_sequence_checker_);
  frame_buffer_->UpdateRtt(max_rtt_ms);
  rtp_video_stream_receiver_.UpdateRtt(max_rtt_ms);
}

void VideoReceiveStream::OnRttUpdated(int64_t rtt_ms) {
  frame_buffer_->UpdateRtt(rtt_ms);
}

int VideoReceiveStream::id() const {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
  return config_.rtp.remote_ssrc;
}

absl::optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&module_process_sequence_checker_);
  absl::optional<Syncable::Info> info =
      rtp_video_stream_receiver_.GetSyncInfo();

  if (!info)
    return absl::nullopt;

  info->current_delay_ms = video_receiver_.Delay();
  return info;
}

uint32_t VideoReceiveStream::GetPlayoutTimestamp() const {
  RTC_NOTREACHED();
  return 0;
}

void VideoReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
  RTC_DCHECK_CALLED_SEQUENTIALLY(&module_process_sequence_checker_);
  rtc::CritScope cs(&playout_delay_lock_);
  syncable_minimum_playout_delay_ms_ = delay_ms;
  UpdatePlayoutDelays();
}

void VideoReceiveStream::DecodeThreadFunction(void* ptr) {
  ScopedRegisterThreadForDebugging thread_dbg(RTC_FROM_HERE);
  while (static_cast<VideoReceiveStream*>(ptr)->Decode()) {
  }
}

bool VideoReceiveStream::Decode() {
  TRACE_EVENT0("webrtc", "VideoReceiveStream::Decode");

  const int wait_ms =
      keyframe_required_ ? max_wait_for_keyframe_ms_ : max_wait_for_frame_ms_;
  std::unique_ptr<video_coding::EncodedFrame> frame;
  // TODO(philipel): Call NextFrame with |keyframe_required| argument when
  //                 downstream project has been fixed.
  video_coding::FrameBuffer::ReturnReason res =
      frame_buffer_->NextFrame(wait_ms, &frame);

  if (res == video_coding::FrameBuffer::ReturnReason::kStopped) {
    return false;
  }

  if (frame) {
    int64_t now_ms = clock_->TimeInMilliseconds();
    RTC_DCHECK_EQ(res, video_coding::FrameBuffer::ReturnReason::kFrameFound);

    // Current OnPreDecode only cares about QP for VP8.
    int qp = -1;
    if (frame->CodecSpecific()->codecType == kVideoCodecVP8) {
      if (!vp8::GetQp(frame->data(), frame->size(), &qp)) {
        RTC_LOG(LS_WARNING) << "Failed to extract QP from VP8 video frame";
      }
    }
    stats_proxy_.OnPreDecode(frame->CodecSpecific()->codecType, qp);

    int decode_result = video_receiver_.Decode(frame.get());
    if (decode_result == WEBRTC_VIDEO_CODEC_OK ||
        decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) {
      keyframe_required_ = false;
      frame_decoded_ = true;
      rtp_video_stream_receiver_.FrameDecoded(frame->id.picture_id);

      if (decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME)
        RequestKeyFrame();
    } else if (!frame_decoded_ || !keyframe_required_ ||
               (last_keyframe_request_ms_ + max_wait_for_keyframe_ms_ <
                now_ms)) {
      keyframe_required_ = true;
      // TODO(philipel): Remove this keyframe request when downstream project
      //                 has been fixed.
      RequestKeyFrame();
      last_keyframe_request_ms_ = now_ms;
    }
  } else {
    RTC_DCHECK_EQ(res, video_coding::FrameBuffer::ReturnReason::kTimeout);
    int64_t now_ms = clock_->TimeInMilliseconds();
    absl::optional<int64_t> last_packet_ms =
        rtp_video_stream_receiver_.LastReceivedPacketMs();
    absl::optional<int64_t> last_keyframe_packet_ms =
        rtp_video_stream_receiver_.LastReceivedKeyframePacketMs();

    // To avoid spamming keyframe requests for a stream that is not active we
    // check if we have received a packet within the last 5 seconds.
    bool stream_is_active = last_packet_ms && now_ms - *last_packet_ms < 5000;
    if (!stream_is_active)
      stats_proxy_.OnStreamInactive();

    // If we recently have been receiving packets belonging to a keyframe then
    // we assume a keyframe is currently being received.
    bool receiving_keyframe =
        last_keyframe_packet_ms &&
        now_ms - *last_keyframe_packet_ms < max_wait_for_keyframe_ms_;

    if (stream_is_active && !receiving_keyframe &&
        (!config_.crypto_options.sframe.require_frame_encryption ||
         rtp_video_stream_receiver_.IsDecryptable())) {
      RTC_LOG(LS_WARNING) << "No decodable frame in " << wait_ms
                          << " ms, requesting keyframe.";
      RequestKeyFrame();
    }
  }
  return true;
}

void VideoReceiveStream::UpdatePlayoutDelays() const {
  const int minimum_delay_ms =
      std::max({frame_minimum_playout_delay_ms_, base_minimum_playout_delay_ms_,
                syncable_minimum_playout_delay_ms_});
  if (minimum_delay_ms >= 0) {
    timing_->set_min_playout_delay(minimum_delay_ms);
  }

  const int maximum_delay_ms = frame_maximum_playout_delay_ms_;
  if (maximum_delay_ms >= 0) {
    timing_->set_max_playout_delay(maximum_delay_ms);
  }
}

std::vector<webrtc::RtpSource> VideoReceiveStream::GetSources() const {
  return rtp_video_stream_receiver_.GetSources();
}

}  // namespace internal
}  // namespace webrtc