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 ce9770e8a461b68fa67c61ae9e17979b8199035c..03d2ccb21efdab99e955161e51fbbc5388bbcea9 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1285,19 +1285,22 @@ 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) {

[michaelt, 2016/10/11 11:38:40]

Have you done a real call with this ?
ModifyEncoder will destroy the encoder_factory. So we have to be sure that all
actions with the encoder_factory are done before we call ModifyEncoder.

[minyue-webrtc, 2016/10/11 11:59:21]

I think this should work. But thank you for bring it up. I will try it.

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

[michaelt, 2016/10/11 11:38:40]

shouldn't we remove this double smoothing for ANA ?

[minyue-webrtc, 2016/10/11 11:59:21]

Good question. Now the forking between using old SetPacketLossRate or
AudioNetworkAdaptor is done within OnReceivedUplinkPacketLossFraction, which
makes much better sense.

I think OnReceivedUplinkPacketLossFraction is supposed to receive non-filtered
packet loss fractions. Any filtering is encoder's responsibility.

I will add another patch set to move |network_predictor_| to AudioEncoderOpus.
WDYT? 

[michaelt, 2016/10/11 12:27:23]

Moving |network_predictor_| to the encoder and smooth PL in the case we don't
use ANA seams OK for the moment. We me move in future to make ANA not optional.
In this case we could remove |network_predictor_| and let ANA care of the
smoothing. 

-
-  // 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) {
+      (*encoder)->OnReceivedUplinkPacketLossFraction(average_fraction_loss /
+                                                     255.0f);
+    }
+  });
 }
 
 int32_t Channel::SetVADStatus(bool enableVAD,
@@ -1494,6 +1497,34 @@ 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) {
+      success = (*encoder)->EnableAudioNetworkAdaptor(
+          config_string, Clock::GetRealTimeClock());
+    }
+  });
+  return success;
+}
+
+void Channel::DisableAudioNetworkAdaptor() {
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (*encoder)
+      (*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) {
+      (*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 +1685,12 @@ 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)
+      (*encoder)->OnReceivedRtt(rtt);
+  });
+
   uint32_t ntp_secs = 0;
   uint32_t ntp_frac = 0;
   uint32_t rtp_timestamp = 0;