/*
 *  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 "webrtc/video/video_receive_stream.h"

#include <stdlib.h>

#include <string>

#include "webrtc/base/checks.h"
#include "webrtc/call/congestion_controller.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/video/receive_statistics_proxy.h"
#include "webrtc/video_engine/call_stats.h"
#include "webrtc/video_receive_stream.h"

namespace webrtc {

static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) {
  for (const auto& extension : extensions) {
    if (extension.name == RtpExtension::kTransportSequenceNumber)
      return true;
  }
  return false;
}

std::string VideoReceiveStream::Decoder::ToString() const {
  std::stringstream ss;
  ss << "{decoder: " << (decoder != nullptr ? "(VideoDecoder)" : "nullptr");
  ss << ", payload_type: " << payload_type;
  ss << ", payload_name: " << payload_name;
  ss << ", is_renderer: " << (is_renderer ? "yes" : "no");
  ss << ", expected_delay_ms: " << expected_delay_ms;
  ss << '}';

  return ss.str();
}

std::string VideoReceiveStream::Config::ToString() const {
  std::stringstream ss;
  ss << "{decoders: [";
  for (size_t i = 0; i < decoders.size(); ++i) {
    ss << decoders[i].ToString();
    if (i != decoders.size() - 1)
      ss << ", ";
  }
  ss << ']';
  ss << ", rtp: " << rtp.ToString();
  ss << ", renderer: " << (renderer != nullptr ? "(renderer)" : "nullptr");
  ss << ", render_delay_ms: " << render_delay_ms;
  if (!sync_group.empty())
    ss << ", sync_group: " << sync_group;
  ss << ", pre_decode_callback: "
     << (pre_decode_callback != nullptr ? "(EncodedFrameObserver)" : "nullptr");
  ss << ", pre_render_callback: "
     << (pre_render_callback != nullptr ? "(I420FrameCallback)" : "nullptr");
  ss << ", target_delay_ms: " << target_delay_ms;
  ss << '}';

  return ss.str();
}

std::string VideoReceiveStream::Config::Rtp::ToString() const {
  std::stringstream ss;
  ss << "{remote_ssrc: " << remote_ssrc;
  ss << ", local_ssrc: " << local_ssrc;
  ss << ", rtcp_mode: "
     << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
                                          : "RtcpMode::kReducedSize");
  ss << ", rtcp_xr: ";
  ss << "{receiver_reference_time_report: "
     << (rtcp_xr.receiver_reference_time_report ? "on" : "off");
  ss << '}';
  ss << ", remb: " << (remb ? "on" : "off");
  ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
  ss << ", fec: " << fec.ToString();
  ss << ", rtx: {";
  for (auto& kv : rtx) {
    ss << kv.first << " -> ";
    ss << "{ssrc: " << kv.second.ssrc;
    ss << ", payload_type: " << kv.second.payload_type;
    ss << '}';
  }
  ss << '}';
  ss << ", extensions: [";
  for (size_t i = 0; i < extensions.size(); ++i) {
    ss << extensions[i].ToString();
    if (i != extensions.size() - 1)
      ss << ", ";
  }
  ss << ']';
  ss << '}';
  return ss.str();
}

namespace internal {
namespace {

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

  codec.plType = decoder.payload_type;
  strcpy(codec.plName, decoder.payload_name.c_str());
  if (decoder.payload_name == "VP8") {
    codec.codecType = kVideoCodecVP8;
  } else if (decoder.payload_name == "VP9") {
    codec.codecType = kVideoCodecVP9;
  } else if (decoder.payload_name == "H264") {
    codec.codecType = kVideoCodecH264;
  } else {
    codec.codecType = kVideoCodecGeneric;
  }

  if (codec.codecType == kVideoCodecVP8) {
    codec.codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings();
  } else if (codec.codecType == kVideoCodecVP9) {
    codec.codecSpecific.VP9 = VideoEncoder::GetDefaultVp9Settings();
  } else if (codec.codecType == kVideoCodecH264) {
    codec.codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
  }

  codec.width = 320;
  codec.height = 180;
  codec.startBitrate = codec.minBitrate = codec.maxBitrate =
      Call::Config::kDefaultStartBitrateBps / 1000;

  return codec;
}
}  // namespace

VideoReceiveStream::VideoReceiveStream(
    int num_cpu_cores,
    CongestionController* congestion_controller,
    const VideoReceiveStream::Config& config,
    webrtc::VoiceEngine* voice_engine,
    ProcessThread* process_thread,
    CallStats* call_stats)
    : transport_adapter_(config.rtcp_send_transport),
      encoded_frame_proxy_(config.pre_decode_callback),
      config_(config),
      clock_(Clock::GetRealTimeClock()),
      congestion_controller_(congestion_controller),
      call_stats_(call_stats) {
  LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();

  bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions);

  RemoteBitrateEstimator* bitrate_estimator =
      congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe);

  vie_channel_.reset(new ViEChannel(
      num_cpu_cores, &transport_adapter_, process_thread, nullptr,
      congestion_controller_->GetBitrateController()->
          CreateRtcpBandwidthObserver(),
      nullptr, bitrate_estimator, call_stats_->rtcp_rtt_stats(),
      congestion_controller_->pacer(), congestion_controller_->packet_router(),
      1, false));

  RTC_CHECK(vie_channel_->Init() == 0);

  // Register the channel to receive stats updates.
  call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver());

  // TODO(pbos): This is not fine grained enough...
  vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false,
                                  -1, -1);
  RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
      << "A stream should not be configured with RTCP disabled. This value is "
         "reserved for internal usage.";
  vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);

  RTC_DCHECK(config_.rtp.remote_ssrc != 0);
  // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
  RTC_DCHECK(config_.rtp.local_ssrc != 0);
  RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);

  vie_channel_->SetSSRC(config_.rtp.local_ssrc, kViEStreamTypeNormal, 0);
  // TODO(pbos): Support multiple RTX, per video payload.
  Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin();
  for (; it != config_.rtp.rtx.end(); ++it) {
    RTC_DCHECK(it->second.ssrc != 0);
    RTC_DCHECK(it->second.payload_type != 0);

    vie_channel_->SetRemoteSSRCType(kViEStreamTypeRtx, it->second.ssrc);
    vie_channel_->SetRtxReceivePayloadType(it->second.payload_type, it->first);
  }
  // TODO(holmer): When Chrome no longer depends on this being false by default,
  // always use the mapping and remove this whole codepath.
  vie_channel_->SetUseRtxPayloadMappingOnRestore(
      config_.rtp.use_rtx_payload_mapping_on_restore);

  congestion_controller_->SetChannelRembStatus(false, config_.rtp.remb,
                                               vie_channel_->rtp_rtcp());

  for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
    const std::string& extension = config_.rtp.extensions[i].name;
    int id = config_.rtp.extensions[i].id;
    // One-byte-extension local identifiers are in the range 1-14 inclusive.
    RTC_DCHECK_GE(id, 1);
    RTC_DCHECK_LE(id, 14);
    if (extension == RtpExtension::kTOffset) {
      RTC_CHECK_EQ(0, vie_channel_->SetReceiveTimestampOffsetStatus(true, id));
    } else if (extension == RtpExtension::kAbsSendTime) {
      RTC_CHECK_EQ(0, vie_channel_->SetReceiveAbsoluteSendTimeStatus(true, id));
    } else if (extension == RtpExtension::kVideoRotation) {
      RTC_CHECK_EQ(0, vie_channel_->SetReceiveVideoRotationStatus(true, id));
    } else if (extension == RtpExtension::kTransportSequenceNumber) {
      RTC_CHECK_EQ(0,
                   vie_channel_->SetReceiveTransportSequenceNumber(true, id));
    } else {
      RTC_NOTREACHED() << "Unsupported RTP extension.";
    }
  }

  if (config_.rtp.fec.ulpfec_payload_type != -1) {
    // ULPFEC without RED doesn't make sense.
    RTC_DCHECK(config_.rtp.fec.red_payload_type != -1);
    VideoCodec codec;
    memset(&codec, 0, sizeof(codec));
    codec.codecType = kVideoCodecULPFEC;
    strcpy(codec.plName, "ulpfec");
    codec.plType = config_.rtp.fec.ulpfec_payload_type;
    RTC_CHECK_EQ(0, vie_channel_->SetReceiveCodec(codec));
  }
  if (config_.rtp.fec.red_payload_type != -1) {
    VideoCodec codec;
    memset(&codec, 0, sizeof(codec));
    codec.codecType = kVideoCodecRED;
    strcpy(codec.plName, "red");
    codec.plType = config_.rtp.fec.red_payload_type;
    RTC_CHECK_EQ(0, vie_channel_->SetReceiveCodec(codec));
    if (config_.rtp.fec.red_rtx_payload_type != -1) {
      vie_channel_->SetRtxReceivePayloadType(
          config_.rtp.fec.red_rtx_payload_type,
          config_.rtp.fec.red_payload_type);
    }
  }

  if (config.rtp.rtcp_xr.receiver_reference_time_report)
    vie_channel_->SetRtcpXrRrtrStatus(true);

  stats_proxy_.reset(
      new ReceiveStatisticsProxy(config_.rtp.remote_ssrc, clock_));

  vie_channel_->RegisterReceiveStatisticsProxy(stats_proxy_.get());
  vie_channel_->RegisterReceiveChannelRtcpStatisticsCallback(
      stats_proxy_.get());
  vie_channel_->RegisterReceiveChannelRtpStatisticsCallback(stats_proxy_.get());
  vie_channel_->RegisterRtcpPacketTypeCounterObserver(stats_proxy_.get());

  RTC_DCHECK(!config_.decoders.empty());
  for (size_t i = 0; i < config_.decoders.size(); ++i) {
    const Decoder& decoder = config_.decoders[i];
    RTC_CHECK_EQ(0,
                 vie_channel_->RegisterExternalDecoder(
                     decoder.payload_type, decoder.decoder, decoder.is_renderer,
                     decoder.is_renderer ? decoder.expected_delay_ms
                                         : config.render_delay_ms));

    VideoCodec codec = CreateDecoderVideoCodec(decoder);

    RTC_CHECK_EQ(0, vie_channel_->SetReceiveCodec(codec));
  }

  incoming_video_stream_.reset(new IncomingVideoStream(0));
  incoming_video_stream_->SetExpectedRenderDelay(config.render_delay_ms);
  incoming_video_stream_->SetExternalCallback(this);
  vie_channel_->SetIncomingVideoStream(incoming_video_stream_.get());

  vie_channel_->RegisterPreDecodeImageCallback(this);
  vie_channel_->RegisterPreRenderCallback(this);
}

VideoReceiveStream::~VideoReceiveStream() {
  LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
  incoming_video_stream_->Stop();
  vie_channel_->RegisterPreRenderCallback(nullptr);
  vie_channel_->RegisterPreDecodeImageCallback(nullptr);

  for (size_t i = 0; i < config_.decoders.size(); ++i)
    vie_channel_->DeRegisterExternalDecoder(config_.decoders[i].payload_type);

  call_stats_->DeregisterStatsObserver(vie_channel_->GetStatsObserver());
  congestion_controller_->SetChannelRembStatus(false, false,
                                               vie_channel_->rtp_rtcp());

  uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC();
  bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions);
  congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe)->
      RemoveStream(remote_ssrc);
}

void VideoReceiveStream::Start() {
  transport_adapter_.Enable();
  incoming_video_stream_->Start();
  vie_channel_->StartReceive();
}

void VideoReceiveStream::Stop() {
  incoming_video_stream_->Stop();
  vie_channel_->StopReceive();
  transport_adapter_.Disable();
}

void VideoReceiveStream::SetSyncChannel(VoiceEngine* voice_engine,
                                        int audio_channel_id) {
  if (voice_engine != nullptr && audio_channel_id != -1) {
    VoEVideoSync* voe_sync_interface = VoEVideoSync::GetInterface(voice_engine);
    vie_channel_->SetVoiceChannel(audio_channel_id, voe_sync_interface);
    voe_sync_interface->Release();
  } else {
    vie_channel_->SetVoiceChannel(-1, nullptr);
  }
}

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

bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
  return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
}

bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
                                    size_t length,
                                    const PacketTime& packet_time) {
  return vie_channel_->ReceivedRTPPacket(packet, length, packet_time) == 0;
}

void VideoReceiveStream::FrameCallback(VideoFrame* video_frame) {
  stats_proxy_->OnDecodedFrame();

  // Post processing is not supported if the frame is backed by a texture.
  if (video_frame->native_handle() == NULL) {
    if (config_.pre_render_callback)
      config_.pre_render_callback->FrameCallback(video_frame);
  }
}

int VideoReceiveStream::RenderFrame(const uint32_t /*stream_id*/,
                                    const VideoFrame& video_frame) {
  // TODO(pbos): Wire up config_.render->IsTextureSupported() and convert if not
  // supported. Or provide methods for converting a texture frame in
  // VideoFrame.

  if (config_.renderer != nullptr)
    config_.renderer->RenderFrame(
        video_frame,
        video_frame.render_time_ms() - clock_->TimeInMilliseconds());

  stats_proxy_->OnRenderedFrame(video_frame.width(), video_frame.height());

  return 0;
}

// TODO(asapersson): Consider moving callback from video_encoder.h or
// creating a different callback.
int32_t VideoReceiveStream::Encoded(
    const EncodedImage& encoded_image,
    const CodecSpecificInfo* codec_specific_info,
    const RTPFragmentationHeader* fragmentation) {
  stats_proxy_->OnPreDecode(encoded_image, codec_specific_info);
  if (config_.pre_decode_callback) {
    // TODO(asapersson): Remove EncodedFrameCallbackAdapter.
    encoded_frame_proxy_.Encoded(
        encoded_image, codec_specific_info, fragmentation);
  }
  return 0;
}

void VideoReceiveStream::SignalNetworkState(NetworkState state) {
  vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
                                                : RtcpMode::kOff);
}

}  // namespace internal
}  // namespace webrtc