Handle padded audio packets correctly

RTP packets can be padded with extra data at the end of the payload. The usable
payload length of the packet should then be reduced with the padding length,
since the padding must be discarded. This was not the case; instead, the entire
payload, including padding data, was forwarded to the audio channel and in the
end to the decoder.

A special case of padding is packets which are empty except for the padding.
That is, they carry no usable payload. These packets are sometimes used for
probing the network and were discarded in
RTPReceiverAudio::ParseAudioCodecSpecific. The result is that NetEq never sees
those empty packets, just the holes in the sequence number series; this can
throw off the target buffer calculations.

With this change, the empty (after removing the padding) packets are let through,
all the way down to NetEq, to a new method called NetEq::InsertEmptyPacket. This
method notifies the DelayManager that an empty packet was received.

BUG=webrtc:7610, webrtc:7625

Review-Url: https://codereview.webrtc.org/2870043003
Cr-Commit-Position: refs/heads/master@{#18083}

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));