Index: webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
diff --git a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
index fb5adb6cd8e55ca236138a00e762d75b5151cde5..88362c145c3651d42d8e27eba744b333aae802e7 100644 |
--- a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
+++ b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
@@ -11,14 +11,39 @@ |
#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h" |
#include "webrtc/base/checks.h" |
+#include "webrtc/base/optional.h" |
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
namespace webrtc { |
namespace { |
+rtc::Optional<uint64_t> ReadNBytes(const uint8_t* data, size_t size, size_t N) { |
pbos-webrtc
2016/01/20 13:16:36
This sounds like memcpy, can you rename this to "P
hlundin-webrtc
2016/01/21 15:18:51
Sure can do. But it is not really memcpy, dependin
|
+ RTC_CHECK_LE(N, sizeof(uint64_t)); |
+ RTC_CHECK_GT(N, static_cast<size_t>(0)); |
+ if (size < N) |
+ return rtc::Optional<uint64_t>(); |
+ uint64_t val = 0; |
+ for (size_t i = 0; i < N; ++i) { |
+ val = (val << 8) + data[i]; |
+ } |
+ return rtc::Optional<uint64_t>(val); |
+} |
+ |
size_t PacketSizeFromTwoBytes(const uint8_t* data, size_t size) { |
- if (size < 2) |
- return 0; |
- return static_cast<size_t>((data[0] << 8) + data[1]); |
+ rtc::Optional<uint64_t> val = ReadNBytes(data, size, 2); |
+ return val ? static_cast<size_t>(*val) : 0; |
+} |
+ |
+template <typename T> |
+T ReadTCheckedAndIncrement(const uint8_t** data, size_t* size) { |
pbos-webrtc
2016/01/20 13:16:36
Can we do something like ReadUint? this sounds lik
terelius
2016/01/20 13:47:27
You could do a static_assert on std::numeric_limit
terelius
2016/01/20 13:50:06
Hm, actually it won't. std::numeric_limits<T>::is_
hlundin-webrtc
2016/01/21 15:18:51
Done.
|
+ static_assert(sizeof(T) <= sizeof(uint64_t), |
+ "Cannot read wider than uint64_t."); |
+ const size_t N = sizeof(T); |
+ rtc::Optional<uint64_t> val = ReadNBytes(*data, *size, N); |
+ RTC_CHECK(val); |
+ *data += N; |
+ RTC_DCHECK_GE(*size, N); // Make extra sure we don't wrap *size. |
+ *size -= N; |
+ return static_cast<T>(*val); |
} |
} // namespace |
@@ -35,7 +60,7 @@ void FuzzAudioDecoder(const uint8_t* data, |
const uint8_t* data_ptr = data; |
size_t remaining_size = size; |
size_t packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size); |
- while (packet_len != 0 && packet_len <= remaining_size - 2) { |
+ while (packet_len != 0 && packet_len + 2 <= remaining_size) { |
hlundin-webrtc
2016/01/20 12:24:59
(Turned out to be a bad idea to subtract from a si
pbos-webrtc
2016/01/20 13:16:36
:D
|
data_ptr += 2; |
remaining_size -= 2; |
AudioDecoder::SpeechType speech_type; |
@@ -46,4 +71,55 @@ void FuzzAudioDecoder(const uint8_t* data, |
packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size); |
} |
} |
+ |
+// This function is identical to FuzzAudioDecoder above, with the distinction |
+// that it call DecodeRedundant instead of Decode. |
+void FuzzAudioDecoderRedundant(const uint8_t* data, |
+ size_t size, |
+ AudioDecoder* decoder, |
+ int sample_rate_hz, |
+ size_t max_decoded_bytes, |
+ int16_t* decoded) { |
+ const uint8_t* data_ptr = data; |
+ size_t remaining_size = size; |
+ size_t packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size); |
+ while (packet_len != 0 && packet_len + 2 <= remaining_size) { |
+ data_ptr += 2; |
+ remaining_size -= 2; |
+ AudioDecoder::SpeechType speech_type; |
+ decoder->DecodeRedundant(data_ptr, packet_len, sample_rate_hz, |
+ max_decoded_bytes, decoded, &speech_type); |
+ data_ptr += packet_len; |
+ remaining_size -= packet_len; |
+ packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size); |
+ } |
+} |
+ |
+// This function is similar to FuzzAudioDecoder, but also reads fuzzed data into |
+// RTP header values. The fuzzed data and values are sent to the decoder's |
+// IncomingPacket method. |
+void FuzzAudioDecoderIncomingPacket(const uint8_t* data, |
+ size_t size, |
+ AudioDecoder* decoder) { |
+ const uint8_t* data_ptr = data; |
+ size_t remaining_size = size; |
+ size_t packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size); |
+ // Sum length of rtp_sequence_number, rtp_timestamp, and arrival_timestamp. |
+ const size_t header_len = sizeof(uint16_t) + 2 * sizeof(uint32_t); |
+ while (packet_len != 0 && packet_len + 2 + header_len <= remaining_size) { |
+ data_ptr += 2; |
+ remaining_size -= 2; |
+ const uint16_t rtp_sequence_number = |
+ ReadTCheckedAndIncrement<uint16_t>(&data_ptr, &remaining_size); |
+ const uint32_t rtp_timestamp = |
+ ReadTCheckedAndIncrement<uint32_t>(&data_ptr, &remaining_size); |
+ const uint32_t arrival_timestamp = |
+ ReadTCheckedAndIncrement<uint32_t>(&data_ptr, &remaining_size); |
+ decoder->IncomingPacket(data_ptr, packet_len, rtp_sequence_number, |
+ rtp_timestamp, arrival_timestamp); |
+ data_ptr += packet_len; |
+ remaining_size -= packet_len; |
+ packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size); |
+ } |
+} |
} // namespace webrtc |