NetEq: Limit payload size for replacement audio input

webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc

 /*
  *  Copyright (c) 2016 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 68 matching lines @@
     // If CNG, simply insert a zero-energy one-byte payload.
     uint8_t cng_payload[1] = {127};  // Max attenuation of CNG.
     packet_->payload.SetData(cng_payload);
     return;
   }
 
   rtc::Optional<RTPHeader> next_hdr = source_->NextHeader();
   RTC_DCHECK(next_hdr);
   uint8_t payload[12];
   uint32_t input_frame_size_timestamps = last_frame_size_timestamps_;
-  if (next_hdr->sequenceNumber == packet_->header.sequenceNumber + 1) {
-    // Packets are in order.
-    input_frame_size_timestamps =
-        next_hdr->timestamp - packet_->header.timestamp;
+  const uint32_t timestamp_diff =
+      next_hdr->timestamp - packet_->header.timestamp;
+  if (next_hdr->sequenceNumber == packet_->header.sequenceNumber + 1 &&
+      timestamp_diff <= 120 * 48) {
+    // Packets are in order and the timestamp diff is less than 5760 samples.
+    // Accept the timestamp diff as a valid frame size.
+    input_frame_size_timestamps = timestamp_diff;
     last_frame_size_timestamps_ = input_frame_size_timestamps;
   }
+  RTC_DCHECK_LE(input_frame_size_timestamps, 120 * 48);

[AleBzk, 2017/04/26 14:26:48]

I'd rather do this before line 88 and on last_frame_size_timestamps_, not
input_frame_size_timestamps.

Let me make an example.
Imagine that last_frame_size_timestamps_ is initialized to 300 * 48 (or any
other value > 120 * 48).
Then, until you don't come across a timestamp_diff <= 120 * 48,
last_frame_size_timestamps_ is not updated and
FakeDecodeFromFile::PrepareEncoded() is called using 300 * 48.

If your intention is to always limit to 120ms, then you need to move and change
the DCHECK as I proposed above. In this way, if someone changes the init value
for last_frame_size_timestamps_ in the .h file, and the value is > 120 * 48, the
code fails.

[hlundin-webrtc, 2017/04/26 14:32:32]

I see your point. But I think that we are not as much documenting what the init
should be as asserting that at the end of the day PrepareEncoded will be called
with a sane value.

I did both now. :)

   FakeDecodeFromFile::PrepareEncoded(packet_->header.timestamp,
                                      input_frame_size_timestamps,
                                      packet_->payload.size(), payload);
   packet_->payload.SetData(payload);
   packet_->header.payloadType = replacement_payload_type_;
   return;
 }
 
 }  // namespace test
 }  // namespace webrtc