Move rtp dump writer from quality test to test transport

webrtc/test/layer_filtering_transport.cc

Index: webrtc/test/layer_filtering_transport.cc
diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc
index 7d7288a0b56addc655fa62d21e58812dddb4af53..e180be827103aab93925d076d67b59a280028e3e 100644
--- a/webrtc/test/layer_filtering_transport.cc
+++ b/webrtc/test/layer_filtering_transport.cc
@@ -16,6 +16,7 @@
 #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_writer.h"
 
 namespace webrtc {
 namespace test {
@@ -28,13 +29,19 @@ LayerFilteringTransport::LayerFilteringTransport(
     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),
+      receiver_(nullptr),
+      start_ms_(rtc::TimeMillis()),
+      rtp_file_writer_(std::move(rtp_file_writer)) {
+  DirectTransport::SetReceiver(this);
+}
 
 bool LayerFilteringTransport::DiscardedLastPacket() const {
   return discarded_last_packet_;
@@ -43,6 +50,7 @@ bool LayerFilteringTransport::DiscardedLastPacket() const {
 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);
@@ -52,7 +60,6 @@ bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
   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);
@@ -105,5 +112,29 @@ bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
   return test::DirectTransport::SendRtp(temp_buffer, length, options);
 }
 
+void LayerFilteringTransport::SetReceiver(PacketReceiver* receiver) {
+  receiver_ = receiver;
+}
+
+PacketReceiver::DeliveryStatus LayerFilteringTransport::DeliverPacket(

[stefan-webrtc, 2017/08/24 12:52:18]

Maybe we should make this a feature of DirectTransport instead?

+    MediaType media_type,
+    const uint8_t* packet,
+    size_t length,
+    const PacketTime& packet_time) {
+  if (rtp_file_writer_.get()) {
+    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_;
+    rtp_file_writer_->WritePacket(&rtp_packet);
+  }
+
+  if (!receiver_)
+    return PacketReceiver::DELIVERY_PACKET_ERROR;
+  return receiver_->DeliverPacket(media_type, packet, length, packet_time);
+}
+
 }  // namespace test
 }  // namespace webrtc