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;
+  static const int kPaddingHeaderId = 0;
 
   static unsigned int ParseSsrc(const void* data, size_t len, bool rtcp) {

[minyue-webrtc, 2015/08/06 13:31:22]

It would be good to get rid of this parser too. I'd like to remove the RTP part
in this function and leave the RTCP parsing to a separate CL. 

     const size_t ssrc_pos = (!rtcp) ? 8 : 4;
     unsigned int ssrc = 0;
     if (len >= (ssrc_pos + sizeof(ssrc))) {
       ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
     }
     return ssrc;
   }
+
+  // Parses audio level in RTP header extension according to

[hlundin-webrtc, 2015/08/05 14:04:24]

I suggest you use RtpHeaderParser from
webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h.

[minyue-webrtc, 2015/08/05 14:15:16]

I actually tried it, but it did not fit here very well. I forgot what the
problem was. I'll reconsider it.

[hlundin-webrtc, 2015/08/05 15:41:19]

OK. I used it in webrtc::test::RtpFileSource. Look there and see if our use
cases match.

[Andrew MacDonald, 2015/08/05 16:37:07]

In any case, you really don't want to be doing this header parsing in a test
file. The existing one should be refactored to meet your needs, but if that
really doesn't fit, you should lift this out and give it its own unit test.

[minyue-webrtc, 2015/08/06 13:31:22]

Thanks. I agree that it is better to use existing tool.

and I found the reason that the existing tool did not fit, and made it a
separate CL.

+  // https://tools.ietf.org/html/rfc6464
+  static int ParseAudioLevelFromRtp(const uint8_t* data, size_t len) {
+    if (len < 4) {
+      return kInvalidAudioLevel;

[tlegrand-webrtc, 2015/08/05 13:32:44]

If we end up here, the RTP header is too short to contain an audio level value,
but we report invalid value?

[minyue-webrtc, 2015/08/06 13:31:22]

Sorry Tina, other reviewers want these function to be got rid of. I agree with
them. So I will skip your comments in this function. Sorry for the
inconvenience.

+    }
+
+    const uint32_t first_word = rtc::GetBE32(data);

[tlegrand-webrtc, 2015/08/05 13:32:44]

Can we add a more descriptive variable name? What does the first 32 bits(?)
contain?

+    if ((first_word & 0x10000000) == 0) {  // Check if RTP header has extension
+      printf("No extension %x.\n", first_word);

[tlegrand-webrtc, 2015/08/05 13:32:44]

Is the printf needed?

+      return kInvalidAudioLevel;
+    }
+
+    int cc = (first_word >> 24) & 0x0f;  // CSRC count, to calculate header size

[tlegrand-webrtc, 2015/08/05 13:32:44]

cc -> csrc_count, perhaps?

+    size_t ext_pos = 12 + cc * 4;

[tlegrand-webrtc, 2015/08/05 13:32:43]

The variable names are a bit confusing, since they seem to be abreviations.
Consider changing to make them more descriptive, or add comments that explains
what the code does.

Example: ext_pos, ext_type, ext_length.

+    if (len < ext_pos + 4) {
+      return kInvalidAudioLevel;
+    }
+
+    int ext_type = rtc::GetBE16(data + ext_pos) & 0xffff;
+    size_t ext_length = rtc::GetBE16(data + ext_pos + 2) & 0xffff;
+    if (len < ext_pos + ext_length * 4) {
+      return kInvalidAudioLevel;
+    }
+
+    size_t offset = 0;
+    if (ext_type == 0xbede) {
+      // one-byte format
+      // 0                   1
+      // 0 1 2 3 4 5  7 8 9 0 1 2 3 4 5
+      // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      // |  ID   | len=0 |V| level       |
+      // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      while (offset < ext_length * 4) {
+        uint8_t b = rtc::Get8(data, ext_pos + 4 + offset);

[tlegrand-webrtc, 2015/08/05 13:32:44]

Change "b" to a descriptive name.

+        int id = (b >> 4) & 0x0f;
+        if (id == kEnergyHeaderId) {
+          return rtc::Get8(data, ext_pos + 5 + offset) & 0x7f;
+        } else if (id == kPaddingHeaderId) {
+          offset++;
+        } else {
+          int len = b & 0x0f;
+          // The len field is the length minus one
+          // http://tools.ietf.org/html/rfc5285#section-4.2
+          offset += len + 2;
+        }
+      }
+    } else if ((ext_type & 0xfff0) == 0x1000) {
+      // two-byte format

[tlegrand-webrtc, 2015/08/05 13:32:44]

Maybe you can save some code lines by only using the if-statement to decide
where to read the data. I think it is possible to combine the code on line 74-83
and 94-101

+      // 0                   1                   2                   3
+      // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+      // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      // |      ID       |     len=1     |V|    level    |    0 (pad)    |
+      // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      while (offset < ext_length * 4) {
+        int id = rtc::Get8(data, ext_pos + 4 + offset) & 0xff;
+        if (id == kEnergyHeaderId) {
+          return rtc::Get8(data, ext_pos + 6 + offset) & 0x7f;
+        } else if (id == kPaddingHeaderId) {
+          offset++;
+        } else {
+          int len = rtc::Get8(data, ext_pos + 5 + offset) & 0xff;
+          offset += len + 2;
+        }
+      }
+    }
+    return kInvalidAudioLevel;
+  }
 }  // 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,
@@ skipping 11 matching lines @@
   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 {
+void ConferenceTransport::SendPacket(const Packet& packet) {
   unsigned int sender_ssrc;
   int destination = -1;
   switch (packet.type_) {
-    case Packet::Rtp:
+    case Packet::Rtp: {
       sender_ssrc = ParseSsrc(packet.data_, packet.len_, false);
       if (sender_ssrc == kLocalSsrc) {
         remote_network_->ReceivedRTPPacket(reflector_, packet.data_,
                                            packet.len_, webrtc::PacketTime());
       } else {
+        int audio_level = ParseAudioLevelFromRtp(packet.data_, packet.len_);
+        if (!loudest_filter_.DecideForward(sender_ssrc, audio_level)) {
+          break;
+        }
         destination = GetReceiverChannelForSsrc(sender_ssrc);
         if (destination != -1) {
           local_network_->ReceivedRTPPacket(destination, packet.data_,
                                             packet.len_,
                                             webrtc::PacketTime());
         }
       }
       break;
-    case Packet::Rtcp:
+    }
+    case Packet::Rtcp: {
       sender_ssrc = ParseSsrc(packet.data_, packet.len_, true);
       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