Handle padded audio packets correctly

webrtc/modules/audio_coding/acm2/audio_coding_module.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 1061 matching lines @@
 
 rtc::Optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
   rtc::CritScope lock(&acm_crit_sect_);
   return receiver_.LastAudioFormat();
 }
 
 // Incoming packet from network parsed and ready for decode.
 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
                                           const size_t payload_length,
                                           const WebRtcRTPHeader& rtp_header) {
+  RTC_DCHECK_EQ(payload_length == 0, incoming_payload == nullptr);
   return receiver_.InsertPacket(
       rtp_header,
       rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
 }
 
 // Minimum playout delay (Used for lip-sync).
 int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
   if ((time_ms < 0) || (time_ms > 10000)) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Delay must be in the range of 0-1000 milliseconds.");
@@ skipping 270 matching lines @@
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   bool valid = acm2::RentACodec::IsCodecValid(codec);
   if (!valid)
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
                  "Invalid codec setting");
   return valid;
 }
 
 }  // namespace webrtc