AcmReceiver: Look up last decoder in NetEq's table of decoders

webrtc/modules/audio_coding/neteq/neteq_impl.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.
  */
 
@@ skipping 413 matching lines @@
   }
   return rtc::Optional<uint32_t>(
       timestamp_scaler_->ToExternal(playout_timestamp_));
 }
 
 int NetEqImpl::last_output_sample_rate_hz() const {
   rtc::CritScope lock(&crit_sect_);
   return last_output_sample_rate_hz_;
 }
 
+rtc::Optional<CodecInst> NetEqImpl::GetDecoder(int payload_type) const {
+  rtc::CritScope lock(&crit_sect_);
+  const DecoderDatabase::DecoderInfo* di =
+      decoder_database_->GetDecoderInfo(payload_type);
+  if (!di) {
+    return rtc::Optional<CodecInst>();
+  }
+
+  // Create a CodecInst with some fields set. The remaining fields are zeroed,
+  // but we tell MSan to consider them uninitialized.
+  CodecInst ci = {0};
+  rtc::MsanMarkUninitialized(rtc::MakeArrayView(&ci, 1));
+  ci.pltype = payload_type;
+  std::strncpy(ci.plname, di->name.c_str(), sizeof(ci.plname));
+  ci.plname[sizeof(ci.plname) - 1] = '\0';
+  ci.plfreq = di->IsRed() || di->IsDtmf() ? 8000 : di->SampleRateHz();

[ossu, 2016/09/16 11:39:32]

This looks like a place where something might go wrong in the confluence of my
(upcoming) changes to DecoderDatabase, these changes, and solenberg's change
adding DTMF support at other clockrates. I'm not certain what, if anything, can
be done about this right here and now, though.

[kwiberg-webrtc, 2016/09/16 11:50:38]

It sounds like my best strategy is to land this quickly, then, so that I won't
be the one sitting with a broken CL when the dust settles. :-)

Seriously though, adding support for multiple DTMF clockrates would probably
involve letting NetEq keep track of the DTMF clockrate, meaning we could
actually get rid of one of the two special cases here. ...see, a perfectly
rational-sounding reason for why I should land this CL as-is. :-)

[ossu, 2016/09/16 11:55:59]

Hmm. I think di->GetFormat().clockrate_hz instead of 8000 should keep you safe
including upcoming changes.

[kwiberg-webrtc, 2016/09/16 12:04:20]

No, because (1) that's the RTP timestamp rate, not the sample rate (since it's
taken directly from the SdpAudioFormat), and (2) GetFormat() only works if the
DecoderInfo was created from a NetEqDecoder that makes
RentACodec::NetEqDecoderToSdpAudioFormat return a non-empty value.

[ossu, 2016/09/16 12:10:06]

Yes, but for DTMF the clockrate == sample rate, so that would be correct. For
RED, though, that would probably not work. :/ Should this method really return a
non-empty value for RED?

+  AudioDecoder* const decoder = di->GetDecoder();
+  ci.channels = decoder ? decoder->Channels() : 1;
+  return rtc::Optional<CodecInst>(ci);
+}
+
 int NetEqImpl::SetTargetNumberOfChannels() {
   return kNotImplemented;
 }
 
 int NetEqImpl::SetTargetSampleRate() {
   return kNotImplemented;
 }
 
 int NetEqImpl::LastError() const {
   rtc::CritScope lock(&crit_sect_);
@@ skipping 1631 matching lines @@
   }
 }
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(
       fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
       *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
       tick_timer_.get()));
 }
 }  // namespace webrtc