Hooking up audio network adaptor to VoE.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 582bde5f26216f645340ef2102d87e485ff2fafb..6543789e8b5917be13b11b97717c4eb2abe95283 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1285,19 +1285,27 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) {
 void Channel::SetBitRate(int bitrate_bps) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
-  audio_coding_->SetBitRate(bitrate_bps);
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (!*encoder) {
+      // There is no existing encoder.
+      return;
+    }
+    (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps);
+  });

[kwiberg-webrtc, 2016/10/06 09:46:13]

You can probably just do

  if (*encoder) {
    (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps);
  }

That !*encoder means we have no encoder is probably not that hard to guess.

[minyue-webrtc, 2016/10/11 09:00:22]

Done.

   retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
 }
 
 void Channel::OnIncomingFractionLoss(int fraction_lost) {
   network_predictor_->UpdatePacketLossRate(fraction_lost);
   uint8_t average_fraction_loss = network_predictor_->GetLossRate();
-
-  // Normalizes rate to 0 - 100.
-  if (audio_coding_->SetPacketLossRate(100 * average_fraction_loss / 255) !=
-      0) {
-    assert(false);  // This should not happen.
-  }
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (!*encoder) {
+      // There is no existing encoder.
+      return;
+    }
+    (*encoder)->OnReceivedUplinkPacketLossFraction(average_fraction_loss /
+                                                   255.0f);
+  });
 }
 
 int32_t Channel::SetVADStatus(bool enableVAD,
@@ -1494,6 +1502,41 @@ int Channel::GetOpusDtx(bool* enabled) {
   return success;
 }
 
+bool Channel::EnableAudioNetworkAdaptor(const std::string& config_string) {
+  bool success = false;
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (!*encoder) {
+      // There is no existing encoder.
+      return;
+    }
+    success = (*encoder)->EnableAudioNetworkAdaptor(config_string,
+                                                    Clock::GetRealTimeClock());
+  });
+  return success;
+}
+
+void Channel::DisableAudioNetworkAdaptor() {
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (!*encoder) {
+      // There is no existing encoder.
+      return;
+    }
+    (*encoder)->DisableAudioNetworkAdaptor();
+  });
+}
+
+void Channel::SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                          int max_frame_length_ms) {
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (!*encoder) {
+      // There is no existing encoder.
+      return;
+    }
+    (*encoder)->SetReceiverFrameLengthRange(min_frame_length_ms,
+                                            max_frame_length_ms);
+  });
+}
+
 int32_t Channel::RegisterExternalTransport(Transport* transport) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterExternalTransport()");
@@ -1654,6 +1697,15 @@ int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
   }
   retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
 
+  // Invoke audio encoders OnReceivedRtt().
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (!*encoder) {
+      // There is no existing encoder.
+      return;
+    }
+    (*encoder)->OnReceivedRtt(rtt);
+  });
+
   uint32_t ntp_secs = 0;
   uint32_t ntp_frac = 0;
   uint32_t rtp_timestamp = 0;