Move rtp dump writer from quality test to test transport

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 <memory>
 
 #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/rtc_base/checks.h"
 #include "webrtc/test/layer_filtering_transport.h"
+#include "webrtc/test/rtp_file_reader.h"

[ilnik, 2017/08/23 08:24:25]

Why reader is needed here?

[sprang_webrtc, 2017/08/23 10:32:05]

Oops. Autocomplete. rtp_file_writer was included from header, so it worked
anyway. change this and used a forward declaration in the header instead.

 
 namespace webrtc {
 namespace test {
 
 LayerFilteringTransport::LayerFilteringTransport(
     SingleThreadedTaskQueueForTesting* task_queue,
     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,
-    const std::map<uint8_t, MediaType>& payload_type_map)
+    const std::map<uint8_t, MediaType>& payload_type_map,
+    std::unique_ptr<test::RtpFileWriter> rtp_file_writer)
     : DirectTransport(task_queue, config, send_call, payload_type_map),
       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) {}
+      discarded_last_packet_(false),
+      start_ms_(rtc::TimeMillis()),
+      rtp_file_writer_(std::move(rtp_file_writer)) {}
 
 bool LayerFilteringTransport::DiscardedLastPacket() const {
   return discarded_last_packet_;
 }
 
 bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
                                       size_t length,
                                       const PacketOptions& options) {
+  RtpPacket rtp_packet;
+  RTC_DCHECK_LE(length, IP_PACKET_SIZE);
+  memcpy(rtp_packet.data, packet, length);
+  rtp_packet.length = length;
+  rtp_packet.original_length = length;
+  rtp_packet.time_ms = rtc::TimeMillis() - start_ms_;
+
   if (selected_tl_ == -1 && selected_sl_ == -1) {
     // Nothing to change, forward the packet immediately.
+    if (rtp_file_writer_.get())
+      rtp_file_writer_->WritePacket(&rtp_packet);
     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, 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_;
     std::unique_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.
+        rtp_packet.data[0] |= (1 << 5);  // P = 1.
+        rtp_packet.data[1] &= 0x7F;      // M = 0.
         discarded_last_packet_ = true;
-        temp_buffer[header.headerLength] = 1;  // One byte of padding.
+        rtp_packet.data[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;
+    rtp_packet.data[1] |= kRtpMarkerBitMask;
 
-  return test::DirectTransport::SendRtp(temp_buffer, length, options);
+  if (rtp_file_writer_.get())
+    rtp_file_writer_->WritePacket(&rtp_packet);
+
+  return test::DirectTransport::SendRtp(rtp_packet.data, length, options);
 }
 
 }  // namespace test
 }  // namespace webrtc