iLBC: Handle a case of bad input data

webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.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.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+
+namespace webrtc {
+
+TEST(IlbcTest, BadPacket) {
+  // Get a good packet.
+  AudioEncoderIlbc::Config config;
+  config.frame_size_ms = 20;  // We need 20 ms rather than the default 30 ms;
+                              // otherwise, all possible values of cb_index[2]
+                              // are valid.
+  AudioEncoderIlbc encoder(config);
+  std::vector<int16_t> samples(encoder.SampleRateHz() / 100, 4711);
+  rtc::Buffer packet;
+  int num_10ms_chunks = 0;
+  while (packet.size() == 0) {
+    encoder.Encode(0, samples, &packet);
+    num_10ms_chunks += 1;
+  }
+
+  // Break the packet by setting all bits of the unsigned 7-bit number
+  // cb_index[2] to 1, giving it a value of 127. For a 20 ms packet, this is
+  // too large.
+  EXPECT_EQ(38u, packet.size());
+  rtc::Buffer bad_packet(packet.data(), packet.size());
+  bad_packet[29] |= 0x3f;  // Bits 1-6.
+  bad_packet[30] |= 0x80;  // Bit 0.
+
+  // Decode the bad packet. We expect the decoder to respond by returning -1.
+  AudioDecoderIlbc decoder;
+  std::vector<int16_t> decoded_samples(num_10ms_chunks * samples.size());
+  AudioDecoder::SpeechType speech_type;
+  EXPECT_EQ(-1, decoder.Decode(bad_packet.data(), bad_packet.size(),
+                               encoder.SampleRateHz(),
+                               sizeof(int16_t) * decoded_samples.size(),
+                               decoded_samples.data(), &speech_type));
+
+  // Decode the good packet. This should work, because the failed decoding
+  // should not have left the decoder in a broken state.
+  EXPECT_EQ(static_cast<int>(decoded_samples.size()),
+            decoder.Decode(packet.data(), packet.size(), encoder.SampleRateHz(),
+                           sizeof(int16_t) * decoded_samples.size(),
+                           decoded_samples.data(), &speech_type));
+}
+
+}  // namespace webrtc