Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 114 matching lines @@
 }
 
 // Dumps an AudioCodec in RFC 2327-ish format.
 std::string ToString(const AudioCodec& codec) {
   std::stringstream ss;
   ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
      << " (" << codec.id << ")";
   return ss.str();
 }
 
+std::string ToString(const webrtc::CodecInst& codec) {
+  std::stringstream ss;
+  ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
+     << " (" << codec.pltype << ")";
+  return ss.str();
+}
+
 bool IsCodec(const AudioCodec& codec, const char* ref_name) {
   return (_stricmp(codec.name.c_str(), ref_name) == 0);
 }
 
 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
   return (_stricmp(codec.plname, ref_name) == 0);
 }
 
 bool FindCodec(const std::vector<AudioCodec>& codecs,
                const AudioCodec& codec,
@@ skipping 1307 matching lines @@
   WebRtcAudioReceiveStream(
       int ch,
       uint32_t remote_ssrc,
       uint32_t local_ssrc,
       bool use_transport_cc,
       bool use_nack,
       const std::string& sync_group,
       const std::vector<webrtc::RtpExtension>& extensions,
       webrtc::Call* call,
       webrtc::Transport* rtcp_send_transport,
-      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
-      const std::map<int, webrtc::SdpAudioFormat>& decoder_map)
+      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
       : call_(call), config_() {
     RTC_DCHECK_GE(ch, 0);
     RTC_DCHECK(call);
     config_.rtp.remote_ssrc = remote_ssrc;
     config_.rtp.local_ssrc = local_ssrc;
     config_.rtp.transport_cc = use_transport_cc;
     config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
     config_.rtp.extensions = extensions;
     config_.rtcp_send_transport = rtcp_send_transport;
     config_.voe_channel_id = ch;
     config_.sync_group = sync_group;
     config_.decoder_factory = decoder_factory;
-    config_.decoder_map = decoder_map;
     RecreateAudioReceiveStream();
   }
 
   ~WebRtcAudioReceiveStream() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     call_->DestroyAudioReceiveStream(stream_);
   }
 
   void RecreateAudioReceiveStream(uint32_t local_ssrc) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 368 matching lines @@
     }
     decoder_map.insert({codec.id, std::move(format)});
   }
 
   if (playout_) {
     // Receive codecs can not be changed while playing. So we temporarily
     // pause playout.
     ChangePlayout(false);
   }
 
-  decoder_map_ = std::move(decoder_map);
   for (auto& kv : recv_streams_) {
-    kv.second->RecreateAudioReceiveStream(decoder_map_);
+    kv.second->RecreateAudioReceiveStream(decoder_map);
   }
   recv_codecs_ = codecs;
 
   if (desired_playout_ && !playout_) {
     ChangePlayout(desired_playout_);
   }
   return true;
 }
 
 // Utility function called from SetSendParameters() to extract current send
@@ skipping 336 matching lines @@
     LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
     return false;
   }
 
   // Create a new channel for receiving audio data.
   const int channel = CreateVoEChannel();
   if (channel == -1) {
     return false;
   }
 
+  // Turn off all supported codecs.
+  // TODO(solenberg): Remove once "no codecs" is the default state of a stream.
+  for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
+    voe_codec.pltype = -1;
+    if (engine()->voe()->codec()->SetRecPayloadType(channel, voe_codec) == -1) {
+      LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+      DeleteVoEChannel(channel);
+      return false;
+    }
+  }
+
+  // Only enable those configured for this channel.
+  for (const auto& codec : recv_codecs_) {
+    webrtc::CodecInst voe_codec = {0};
+    if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
+      voe_codec.pltype = codec.id;
+      if (engine()->voe()->codec()->SetRecPayloadType(
+          channel, voe_codec) == -1) {
+        LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+        DeleteVoEChannel(channel);
+        return false;
+      }
+    }
+  }
+
   recv_streams_.insert(std::make_pair(
-      ssrc,
-      new WebRtcAudioReceiveStream(
-          channel, ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
-          recv_nack_enabled_, sp.sync_label, recv_rtp_extensions_, call_, this,
-          engine()->decoder_factory_, decoder_map_)));
+      ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
+                                         recv_transport_cc_enabled_,
+                                         recv_nack_enabled_,
+                                         sp.sync_label, recv_rtp_extensions_,
+                                         call_, this,
+                                         engine()->decoder_factory_)));
   recv_streams_[ssrc]->SetPlayout(playout_);
 
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
 
@@ skipping 384 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE