Address sanitizer out of bounds read in iSAC

BUG=issue1770
TBR=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/1472006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4030 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/codecs/isac/main/source/isac.c

@@ -272,6 +272,7 @@ int16_t WebRtcIsac_Assign(ISACStruct** ISAC_main_inst,
 int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst) {
   ISACMainStruct* instISAC;
 
+  if (ISAC_main_inst != NULL) {
     instISAC = (ISACMainStruct*)WEBRTC_SPL_VNEW(ISACMainStruct, 1);
     *ISAC_main_inst = (ISACStruct*)instISAC;
     if (*ISAC_main_inst != NULL) {
@@ -286,6 +287,9 @@ int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst) {
     } else {
       return -1;
     }
+  } else {
+    return -1;
+  }
 }
 
 
@@ -1015,7 +1019,9 @@ int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
     return -1;
   }
 
-  if (packet_size <= 0) {
+  /* Check that the size of the packet is valid, and if not return without
+   * updating the bandwidth estimate. A valid size is at least 10 bytes. */
+  if (packet_size < 10) {
     /* Return error code if the packet length is null. */
     instISAC->errorCode = ISAC_EMPTY_PACKET;
     return -1;

webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc

@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2013 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 <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+struct WebRtcISACStruct;
+
+namespace webrtc {
+
+// Number of samples in a 60 ms, sampled at 32 kHz.
+const int kIsacNumberOfSamples = 320 * 6;
+// Maximum number of bytes in output bitstream.
+const size_t kMaxBytes = 1000;
+
+class IsacTest : public ::testing::Test {
+ protected:
+  IsacTest();
+  virtual void SetUp();
+
+  WebRtcISACStruct* isac_codec_;
+
+  int16_t speech_data_[kIsacNumberOfSamples];
+  int16_t output_data_[kIsacNumberOfSamples];
+  int16_t bitstream_[kMaxBytes / 2];
+  uint8_t bitstream_small_[7];  // Simulate sync packets.
+};
+
+IsacTest::IsacTest()
+    : isac_codec_(NULL) {
+}
+
+void IsacTest::SetUp() {
+  // Read some samples from a speech file, to be used in the encode test.
+  FILE* input_file;
+  const std::string file_name =
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  input_file = fopen(file_name.c_str(), "rb");
+  ASSERT_TRUE(input_file != NULL);
+  ASSERT_EQ(kIsacNumberOfSamples,
+            static_cast<int32_t>(fread(speech_data_, sizeof(int16_t),
+                                       kIsacNumberOfSamples, input_file)));
+  fclose(input_file);
+  input_file = NULL;
+}
+
+// Test failing Create.
+TEST_F(IsacTest, IsacCreateFail) {
+  // Test to see that an invalid pointer is caught.
+  EXPECT_EQ(-1, WebRtcIsac_Create(NULL));
+}
+
+// Test failing Free.
+TEST_F(IsacTest, IsacFreeFail) {
+  // Test to see that free function doesn't crash.
+  EXPECT_EQ(0, WebRtcIsac_Free(NULL));
+}
+
+// Test normal Create and Free.
+TEST_F(IsacTest, IsacCreateFree) {
+  EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
+  EXPECT_TRUE(isac_codec_ != NULL);
+  EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));}
+
+TEST_F(IsacTest, IsacUpdateBWE) {
+  // Create encoder memory.
+  EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
+
+  // Init encoder (adaptive mode) and decoder.
+  WebRtcIsac_EncoderInit(isac_codec_, 0);
+  WebRtcIsac_DecoderInit(isac_codec_);
+
+  // Encode & decode.
+  int16_t encoded_bytes;
+  uint16_t* coded = reinterpret_cast<uint16_t*>(bitstream_);
+  uint16_t* coded_small = reinterpret_cast<uint16_t*>(bitstream_small_);
+
+  // Test with call with a small packet (sync packet).
+  EXPECT_EQ(-1, WebRtcIsac_UpdateBwEstimate(isac_codec_, coded_small, 7, 1,
+                                            12345, 56789));
+
+  // Encode 60 ms of data (needed to create a first packet).
+  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
+  EXPECT_EQ(0, encoded_bytes);
+  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
+  EXPECT_EQ(0, encoded_bytes);
+  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
+  EXPECT_EQ(0, encoded_bytes);
+  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
+  EXPECT_EQ(0, encoded_bytes);
+  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
+  EXPECT_EQ(0, encoded_bytes);
+  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
+
+  // Call to update bandwidth estimator with real data.
+  EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, coded, encoded_bytes, 1,
+                                           12345, 56789));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/main/source/audio_coding_module.gypi

@@ -188,6 +188,7 @@
              '../../codecs/isac/fix/source/filterbanks_unittest.cc',
              '../../codecs/isac/fix/source/lpc_masking_model_unittest.cc',
              '../../codecs/isac/fix/source/transform_unittest.cc',
+            '../../codecs/isac/main/source/isac_unittest.cc',
              '../../codecs/opus/opus_unittest.cc',
              # Test for NetEq 4.
              '../../neteq4/audio_multi_vector_unittest.cc',