Handle padded audio packets correctly

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

Index: webrtc/modules/audio_coding/acm2/audio_coding_module.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index e22eb5a5914cc4d5ba565659dc4e205eda41672d..551ae057b46b67a27b07406b9991d515c64e9a44 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -1079,6 +1079,7 @@ rtc::Optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
 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));