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 1058 matching lines @@
   rtc::CritScope lock(&acm_crit_sect_);
   return receiver_.LastAudioCodec(current_codec);
 }
 
 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,

[minyue-webrtc, 2017/05/09 21:04:06]

will this affect DTX sync packets?

[hlundin-webrtc, 2017/05/10 08:57:23]

Aren't they 1 or 2 bytes? 
https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/...

[minyue-webrtc, 2017/05/10 09:44:12]

Yes. Thanks

                                           const size_t payload_length,
                                           const WebRtcRTPHeader& rtp_header) {
+  if (payload_length == 0) {
+    // This is an empty packet, which is not a problem, but we will have to
+    // construct an empty ArrayView.
+    return receiver_.InsertPacket(rtp_header, rtc::ArrayView<const uint8_t>());
+  }
   return receiver_.InsertPacket(
       rtp_header,

[minyue-webrtc, 2017/05/09 21:04:06]

reading rtc::ArrayView i think it is safe to just use 1087-1089

[hlundin-webrtc, 2017/05/10 08:57:23]

I don't think so. If we construct an ArrayView with a non-null data pointer, but
a size==0, we will use the ctor at
https://cs.chromium.org/chromium/src/third_party/webrtc/base/array_view.h?typ....
This has a DCHECK at line 149, which says that if size==0, the data must be
nullptr.

[minyue-webrtc, 2017/05/10 09:44:11]

But shouldn't incoming_payload be a null?

[hlundin-webrtc, 2017/05/10 11:19:30]

Done.

       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.");
     return -1;
   }
@@ skipping 268 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