Making FakeNetworkPipe demux audio and video packets.

webrtc/test/fake_network_pipe.cc

 /*
  *  Copyright (c) 2012 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/test/fake_network_pipe.h"
 
 #include <assert.h>
 #include <math.h>
 #include <string.h>
 
 #include <algorithm>
 #include <cmath>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/call/call.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 namespace {
 constexpr int64_t kDefaultProcessIntervalMs = 5;
 }
 
+DemuxerImpl::DemuxerImpl(const std::map<uint8_t, MediaType>& payload_type_map)
+    : packet_receiver_(nullptr), payload_type_map_(payload_type_map) {}
+
+void DemuxerImpl::SetReceiver(PacketReceiver* receiver) {
+  packet_receiver_ = receiver;
+}
+
+void DemuxerImpl::DeliverPacket(const NetworkPacket* packet,
+                                const PacketTime& packet_time) {
+  if (!packet_receiver_)
+    return;
+  const uint8_t* const packet_data = packet->data();
+  const size_t packet_length = packet->data_length();
+  MediaType media_type = MediaType::ANY;
+  if (!RtpHeaderParser::IsRtcp(packet_data, packet_length)) {
+    RTC_CHECK_GE(packet_length, 2);
+    const uint8_t payload_type = packet_data[1] & 0x7f;
+    std::map<uint8_t, MediaType>::const_iterator it =
+        payload_type_map_.find(payload_type);
+    RTC_CHECK(it != payload_type_map_.end())
+        << "payload type " << static_cast<int>(payload_type) << " unknown.";
+    media_type = it->second;
+  }
+  packet_receiver_->DeliverPacket(media_type, packet_data, packet_length,
+                                  packet_time);
+}
+
 FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
                                  const FakeNetworkPipe::Config& config,
-                                 MediaType media_type)
-    : FakeNetworkPipe(clock, config, media_type, 1) {}
+                                 std::unique_ptr<Demuxer> demuxer)
+    : FakeNetworkPipe(clock, config, std::move(demuxer), 1) {}
 
 FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
                                  const FakeNetworkPipe::Config& config,
-                                 MediaType media_type,
+                                 std::unique_ptr<Demuxer> demuxer,
                                  uint64_t seed)
     : clock_(clock),
-      media_type_(media_type),
-      packet_receiver_(NULL),
+      demuxer_(std::move(demuxer)),
       random_(seed),
       config_(),
       dropped_packets_(0),
       sent_packets_(0),
       total_packet_delay_(0),
       bursting_(false),
       next_process_time_(clock_->TimeInMilliseconds()),
       last_log_time_(clock_->TimeInMilliseconds()) {
   SetConfig(config);
 }
 
 FakeNetworkPipe::~FakeNetworkPipe() {
   while (!capacity_link_.empty()) {
     delete capacity_link_.front();
     capacity_link_.pop();
   }
   while (!delay_link_.empty()) {
     delete *delay_link_.begin();
     delay_link_.erase(delay_link_.begin());
   }
 }
 
 void FakeNetworkPipe::SetReceiver(PacketReceiver* receiver) {
-  packet_receiver_ = receiver;
+  if (!demuxer_)

[perkj_webrtc, 2017/04/07 14:24:15]

demuxer always exists right? It is moved into the ctor and the unique_ptr can be
made const and CHECK(demuxer) instead of if...

[minyue-webrtc, 2017/04/10 09:56:10]

Yes, that would be nice.

+    return;
+  demuxer_->SetReceiver(receiver);
 }
 
 void FakeNetworkPipe::SetConfig(const FakeNetworkPipe::Config& config) {
   rtc::CritScope crit(&lock_);
   config_ = config;  // Shallow copy of the struct.
   double prob_loss = config.loss_percent / 100.0;
   if (config_.avg_burst_loss_length == -1) {
     // Uniform loss
     prob_loss_bursting_ = prob_loss;
     prob_start_bursting_ = prob_loss;
   } else {
     // Lose packets according to a gilbert-elliot model.
     int avg_burst_loss_length = config.avg_burst_loss_length;
     int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss));
 
     RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length)
         << "For a total packet loss of " << config.loss_percent << "%% then"
         << " avg_burst_loss_length must be " << min_avg_burst_loss_length + 1
         << " or higher.";
 
     prob_loss_bursting_ = (1.0 - 1.0 / avg_burst_loss_length);
     prob_start_bursting_ = prob_loss / (1 - prob_loss) / avg_burst_loss_length;
   }
 }
 
 void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) {
-  // A NULL packet_receiver_ means that this pipe will terminate the flow of
-  // packets.
-  if (packet_receiver_ == NULL)
+  // No demuxer means that this pipe will terminate the flow of packets.
+  if (!demuxer_)

[stefan-webrtc, 2017/04/10 07:45:26]

As Per pointed out above, this can also be removed.

[minyue-webrtc, 2017/04/10 09:56:10]

Done.

     return;
   rtc::CritScope crit(&lock_);
   if (config_.queue_length_packets > 0 &&
       capacity_link_.size() >= config_.queue_length_packets) {
     // Too many packet on the link, drop this one.
     ++dropped_packets_;
     return;
   }
 
   int64_t time_now = clock_->TimeInMilliseconds();
@@ skipping 90 matching lines @@
       // |time_now| might be later than when the packet should have arrived, due
       // to NetworkProcess being called too late. For stats, use the time it
       // should have been on the link.
       total_packet_delay_ += packet->arrival_time() - packet->send_time();
     }
     sent_packets_ += packets_to_deliver.size();
   }
   while (!packets_to_deliver.empty()) {
     NetworkPacket* packet = packets_to_deliver.front();
     packets_to_deliver.pop();
-    packet_receiver_->DeliverPacket(media_type_, packet->data(),
-                                    packet->data_length(), PacketTime());
+    demuxer_->DeliverPacket(packet, PacketTime());
     delete packet;
   }
 
   next_process_time_ = !delay_link_.empty()
                            ? (*delay_link_.begin())->arrival_time()
                            : time_now + kDefaultProcessIntervalMs;
 }
 
 int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
   rtc::CritScope crit(&lock_);
   return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
                            0);
 }
 
 }  // namespace webrtc