xref: /external/webrtc/webrtc/test/layer_filtering_transport.cc

/*
 *  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/include/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,
    Call* send_call,
    uint8_t vp8_video_payload_type,
    uint8_t vp9_video_payload_type,
    int selected_tl,
    int selected_sl)
    : test::DirectTransport(config, send_call),
      vp8_video_payload_type_(vp8_video_payload_type),
      vp9_video_payload_type_(vp9_video_payload_type),
      selected_tl_(selected_tl),
      selected_sl_(selected_sl),
      discarded_last_packet_(false) {}

bool LayerFilteringTransport::DiscardedLastPacket() const {
  return discarded_last_packet_;
}

bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
                                      size_t length,
                                      const PacketOptions& options) {
  if (selected_tl_ == -1 && selected_sl_ == -1) {
    // Nothing to change, forward the packet immediately.
    return test::DirectTransport::SendRtp(packet, length, options);
  }

  bool set_marker_bit = false;
  RtpUtility::RtpHeaderParser parser(packet, length);
  RTPHeader header;
  parser.Parse(&header);

  RTC_DCHECK_LE(length, static_cast<size_t>(IP_PACKET_SIZE));
  uint8_t temp_buffer[IP_PACKET_SIZE];
  memcpy(temp_buffer, packet, length);

  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 int temporal_idx = static_cast<int>(
          is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
                 : parsed_payload.type.Video.codecHeader.VP9.temporal_idx);
      const int spatial_idx = static_cast<int>(
          is_vp8 ? kNoSpatialIdx
                 : parsed_payload.type.Video.codecHeader.VP9.spatial_idx);
      if (selected_sl_ >= 0 && spatial_idx == selected_sl_ &&
          parsed_payload.type.Video.codecHeader.VP9.end_of_frame) {
        // This layer is now the last in the superframe.
        set_marker_bit = true;
      } else if ((selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
                  temporal_idx > selected_tl_) ||
                 (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
                  spatial_idx > selected_sl_)) {
        // Truncate packet to a padding packet.
        length = header.headerLength + 1;
        temp_buffer[0] |= (1 << 5);  // P = 1.
        temp_buffer[1] &= 0x7F;      // M = 0.
        discarded_last_packet_ = true;
        temp_buffer[header.headerLength] = 1;  // One byte of padding.
      }
    } else {
      RTC_NOTREACHED() << "Parse error";
    }
  }

  // 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;

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

}  // namespace test
}  // namespace webrtc