Revert of AcmReceiver: Ask NetEq to delete all decoders at once instead of one by one

webrtc/modules/audio_coding/acm2/acm_receiver.cc

 /*
  *  Copyright (c) 2013 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 227 matching lines @@
   decoder.channels = channels;
   decoder.sample_rate_hz = sample_rate_hz;
   decoders_[payload_type] = decoder;
   return 0;
 }
 
 void AcmReceiver::FlushBuffers() {
   neteq_->FlushBuffers();
 }
 
-void AcmReceiver::RemoveAllCodecs() {
+// If failed in removing one of the codecs, this method continues to remove as
+// many as it can.
+int AcmReceiver::RemoveAllCodecs() {
+  int ret_val = 0;
   rtc::CritScope lock(&crit_sect_);
-  neteq_->RemoveAllPayloadTypes();
-  decoders_.clear();
+  for (auto it = decoders_.begin(); it != decoders_.end(); ) {
+    auto cur = it;
+    ++it;  // it will be valid even if we erase cur
+    if (neteq_->RemovePayloadType(cur->second.payload_type) == 0) {
+      decoders_.erase(cur);
+    } else {
+      LOG_F(LS_ERROR) << "Cannot remove payload "
+                      << static_cast<int>(cur->second.payload_type);
+      ret_val = -1;
+    }
+  }
+
+  // No codec is registered, invalidate last audio decoder.
   last_audio_decoder_ = rtc::Optional<CodecInst>();
   last_packet_sample_rate_hz_ = rtc::Optional<int>();
+  return ret_val;
 }
 
 int AcmReceiver::RemoveCodec(uint8_t payload_type) {
   rtc::CritScope lock(&crit_sect_);
   auto it = decoders_.find(payload_type);
   if (it == decoders_.end()) {  // Such a payload-type is not registered.
     return 0;
   }
   if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
@@ skipping 110 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc