Add LoudestFilter in ConferenceTransport

webrtc/voice_engine/test/auto_test/fakes/conference_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/voice_engine/test/auto_test/fakes/conference_transport.h"
 
 #include <string>
 
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 
 namespace {
   static const unsigned int kReflectorSsrc = 0x0000;
   static const unsigned int kLocalSsrc = 0x0001;
   static const unsigned int kFirstRemoteSsrc = 0x0002;
   static const webrtc::CodecInst kCodecInst =
       {120, "opus", 48000, 960, 2, 64000};
+  static const int kInvalidAudioLevel = 128;
+  static const int kEnergyHeaderId = 1;

[hlundin-webrtc, 2015/08/10 10:50:17]

Why call it energy header? I suggest kAudioLevelId, or something else including
"audio level".

[minyue-webrtc, 2015/08/12 14:53:23]

Now called kAudioLevelHeaderId

 
-  static unsigned int ParseSsrc(const void* data, size_t len, bool rtcp) {
-    const size_t ssrc_pos = (!rtcp) ? 8 : 4;
+  static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
+    const size_t ssrc_pos = 4;
     unsigned int ssrc = 0;
     if (len >= (ssrc_pos + sizeof(ssrc))) {
       ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
     }
     return ssrc;
   }
 }  // namespace
 
 namespace voetest {
 
 ConferenceTransport::ConferenceTransport()
     : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
       stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
       packet_event_(webrtc::EventWrapper::Create()),
       thread_(webrtc::ThreadWrapper::CreateThread(Run,
                                                   this,
                                                   "ConferenceTransport")),
       rtt_ms_(0),
-      stream_count_(0) {
+      stream_count_(0),
+      rtp_header_parser_(webrtc::RtpHeaderParser::Create()) {
+  rtp_header_parser_->
+      RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
+                                 kEnergyHeaderId);
+
   local_voe_ = webrtc::VoiceEngine::Create();
   local_base_ = webrtc::VoEBase::GetInterface(local_voe_);
   local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
   local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);
 
   // In principle, we can use one VoiceEngine to achieve the same goal. Well, in
   // here, we use two engines to make it more like reality.
   remote_voe_ = webrtc::VoiceEngine::Create();
   remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_);
   remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_);
   remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_);
   remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
   remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);
 
   EXPECT_EQ(0, local_base_->Init());
   local_sender_ = local_base_->CreateChannel();
   EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
   EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
+  EXPECT_EQ(0, local_rtp_rtcp_->
+      SetSendAudioLevelIndicationStatus(local_sender_, true, kEnergyHeaderId));
+
   EXPECT_EQ(0, local_base_->StartSend(local_sender_));
 
   EXPECT_EQ(0, remote_base_->Init());
   reflector_ = remote_base_->CreateChannel();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
   EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
 
   thread_->Start();
   thread_->SetPriority(webrtc::kHighPriority);
 }
@@ skipping 50 matching lines @@
     webrtc::CriticalSectionScoped lock(pq_crit_.get());
     packet_queue_.push_back(Packet(type, channel, data, len, rtc::Time()));
   }
   packet_event_->Set();
 }
 
 // This simulates the flow of RTP and RTCP packets. Complications like that
 // a packet is first sent to the reflector, and then forwarded to the receiver
 // are simplified, in this particular case, to a direct link between the sender
 // and the receiver.
-void ConferenceTransport::SendPacket(const Packet& packet) const {
-  unsigned int sender_ssrc;
+void ConferenceTransport::SendPacket(const Packet& packet) {
   int destination = -1;
+
   switch (packet.type_) {
-    case Packet::Rtp:
-      sender_ssrc = ParseSsrc(packet.data_, packet.len_, false);
-      if (sender_ssrc == kLocalSsrc) {
+    case Packet::Rtp: {
+      webrtc::RTPHeader rtp_header;
+      rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header);
+      if (rtp_header.ssrc == kLocalSsrc) {
         remote_network_->ReceivedRTPPacket(reflector_, packet.data_,
                                            packet.len_, webrtc::PacketTime());
       } else {
-        destination = GetReceiverChannelForSsrc(sender_ssrc);
+        int audio_level = rtp_header.extension.hasAudioLevel ?
+                rtp_header.extension.audioLevel & 0x7f : kInvalidAudioLevel;

[phoglund, 2015/08/10 08:43:27]

Nit: indent

[minyue-webrtc, 2015/08/10 09:30:22]

Yes. and I expect here a rebase after
https://codereview.chromium.org/1272343003/

+        if (!loudest_filter_.DecideForward(rtp_header.ssrc, audio_level)) {

[hlundin-webrtc, 2015/08/10 10:50:17]

I think the code will be more readable if you reverse the logic of this
if-statement. That is:
if (loudest_filter_.DecideForward(...)) {
  destination = GetReceiver...
  ...
}

[hlundin-webrtc, 2015/08/10 10:50:17]

Why parse out two components of the webrtc::RTPHeader struct and then pass them
to the same method? I think DecideForward should take a const webrtc::RTPHeader
reference as input.

[phoglund, 2015/08/10 11:34:04]

+1

[minyue-webrtc, 2015/08/12 14:53:23]

Done.

[minyue-webrtc, 2015/08/12 14:53:23]

Done.

[minyue-webrtc, 2015/08/12 14:53:23]

Done.

+          break;
+        }
+        destination = GetReceiverChannelForSsrc(rtp_header.ssrc);
         if (destination != -1) {
           local_network_->ReceivedRTPPacket(destination, packet.data_,
                                             packet.len_,
                                             webrtc::PacketTime());
         }
       }
       break;
-    case Packet::Rtcp:
-      sender_ssrc = ParseSsrc(packet.data_, packet.len_, true);
+    }
+    case Packet::Rtcp: {
+      unsigned int sender_ssrc = ParseRtcpSsrc(packet.data_, packet.len_);
       if (sender_ssrc == kLocalSsrc) {
         remote_network_->ReceivedRTCPPacket(reflector_, packet.data_,
                                             packet.len_);
       } else if (sender_ssrc == kReflectorSsrc) {
         local_network_->ReceivedRTCPPacket(local_sender_, packet.data_,
                                            packet.len_);
       } else {
         destination = GetReceiverChannelForSsrc(sender_ssrc);
         if (destination != -1) {
           local_network_->ReceivedRTCPPacket(destination, packet.data_,
                                              packet.len_);
         }
       }
       break;
+    }
   }
 }
 
 bool ConferenceTransport::DispatchPackets() {
   switch (packet_event_->Wait(1000)) {
     case webrtc::kEventSignaled:
       break;
     case webrtc::kEventTimeout:
       return true;
     case webrtc::kEventError:
@@ skipping 20 matching lines @@
 
     SendPacket(packet);
   }
   return true;
 }
 
 void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
   rtt_ms_ = rtt_ms;
 }
 
-unsigned int ConferenceTransport::AddStream() {
-  const std::string kInputFileName =
-      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
-
+unsigned int ConferenceTransport::AddStream(std::string file_name,
+                                            webrtc::FileFormats format) {
   const int new_sender = remote_base_->CreateChannel();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
 
   const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
   EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc));
+  EXPECT_EQ(0, remote_rtp_rtcp_->
+      SetSendAudioLevelIndicationStatus(new_sender, true, kEnergyHeaderId));
 
   EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst));
   EXPECT_EQ(0, remote_base_->StartSend(new_sender));
   EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone(
-      new_sender, kInputFileName.c_str(), true, false,
-      webrtc::kFileFormatPcm32kHzFile, 1.0));
+      new_sender, file_name.c_str(), true, false, format, 1.0));
 
   const int new_receiver = local_base_->CreateChannel();
   EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
 
   EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
   // Receive channels have to have the same SSRC in order to send receiver
   // reports with this SSRC.
   EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc));
 
   {
@@ skipping 31 matching lines @@
 bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
                                                 webrtc::CallStatistics* stats) {
   int dst = GetReceiverChannelForSsrc(id);
   if (dst == -1) {
     return false;
   }
   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
   return true;
 }
 }  // namespace voetest