Adding support for simulcast and spatial layers into VideoQualityTest

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/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)
+    int selected_tl,
+    int selected_sl)
     : 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) {
+      selected_tl_(selected_tl),
+      selected_sl_(selected_sl),
+      discarded_last_packet_(false) {
+}
+
+bool LayerFilteringTransport::DiscardedLastPacket() const {
+  return discarded_last_packet_;
 }
 
 uint16_t LayerFilteringTransport::NextSequenceNumber(uint32_t ssrc) {
   auto it = current_seq_nums_.find(ssrc);
   if (it == current_seq_nums_.end())
     return current_seq_nums_[ssrc] = 10000;
   return ++it->second;
 }
 
 bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
                                       size_t length,
                                       const PacketOptions& options) {
-  if (tl_discard_threshold_ == 0 && sl_discard_threshold_ == 0) {
+  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;
   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 =
+      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 uint8_t spatialIdx =
+                 : 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 (sl_discard_threshold_ > 0 &&
-          spatialIdx == sl_discard_threshold_ - 1 &&
+                 : 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;
       }
-      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.
+      if ((selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
+           temporal_idx > selected_tl_) ||
+          (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
+           spatial_idx > selected_sl_)) {
+        discarded_last_packet_ = true;
+        return true;
       }
     } 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;
 
   uint16_t seq_num = NextSequenceNumber(header.ssrc);
   ByteWriter<uint16_t>::WriteBigEndian(&temp_buffer[2], seq_num);
   return test::DirectTransport::SendRtp(temp_buffer, length, options);
 }
 
 }  // namespace test
 }  // namespace webrtc