/*
 *  Copyright (c) 2015 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/base/checks.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/test/layer_filtering_transport.h"

namespace webrtc {
namespace test {

LayerFilteringTransport::LayerFilteringTransport(
    const FakeNetworkPipe::Config& config,
    uint8_t vp8_video_payload_type,
    uint8_t vp9_video_payload_type,
    uint8_t tl_discard_threshold,
    uint8_t sl_discard_threshold)
    : test::DirectTransport(config),
      vp8_video_payload_type_(vp8_video_payload_type),
      vp9_video_payload_type_(vp9_video_payload_type),
      tl_discard_threshold_(tl_discard_threshold),
      sl_discard_threshold_(sl_discard_threshold),
      current_seq_num_(10000) {
}  // TODO(ivica): random seq num?

bool LayerFilteringTransport::SendRtp(const uint8_t* packet, size_t length) {
  if (tl_discard_threshold_ == 0 && sl_discard_threshold_ == 0) {
    // Nothing to change, forward the packet immediately.
    return test::DirectTransport::SendRtp(packet, length);
  }

  bool set_marker_bit = false;
  rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
  RTPHeader header;
  parser->Parse(packet, length, &header);

  if (header.payloadType == vp8_video_payload_type_ ||
      header.payloadType == vp9_video_payload_type_) {
    const uint8_t* payload = packet + header.headerLength;
    RTC_DCHECK_GT(length, header.headerLength);
    const size_t payload_length = length - header.headerLength;
    RTC_DCHECK_GT(payload_length, header.paddingLength);
    const size_t payload_data_length = payload_length - header.paddingLength;

    const bool is_vp8 = header.payloadType == vp8_video_payload_type_;
    rtc::scoped_ptr<RtpDepacketizer> depacketizer(
        RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
    RtpDepacketizer::ParsedPayload parsed_payload;
    if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) {
      const uint8_t temporalIdx =
          is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
                 : parsed_payload.type.Video.codecHeader.VP9.temporal_idx;
      const uint8_t spatialIdx =
          is_vp8 ? kNoSpatialIdx
                 : parsed_payload.type.Video.codecHeader.VP9.spatial_idx;
      if (sl_discard_threshold_ > 0 &&
          spatialIdx == sl_discard_threshold_ - 1 &&
          parsed_payload.type.Video.codecHeader.VP9.end_of_frame) {
        // This layer is now the last in the superframe.
        set_marker_bit = true;
      }
      if ((tl_discard_threshold_ > 0 && temporalIdx != kNoTemporalIdx &&
           temporalIdx >= tl_discard_threshold_) ||
          (sl_discard_threshold_ > 0 && spatialIdx != kNoSpatialIdx &&
           spatialIdx >= sl_discard_threshold_)) {
        return true;  // Discard the packet.
      }
    } else {
      RTC_NOTREACHED() << "Parse error";
    }
  }

  uint8_t temp_buffer[IP_PACKET_SIZE];
  memcpy(temp_buffer, packet, length);

  // We are discarding some of the packets (specifically, whole layers), so
  // make sure the marker bit is set properly, and that sequence numbers are
  // continuous.
  if (set_marker_bit) {
    temp_buffer[1] |= kRtpMarkerBitMask;
  }
  ByteWriter<uint16_t>::WriteBigEndian(&temp_buffer[2], current_seq_num_);

  ++current_seq_num_;  // Increase only if packet not discarded.

  return test::DirectTransport::SendRtp(temp_buffer, length);
}

}  // namespace test
}  // namespace webrtc