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..1c4d9d5660720b740ffec28d86e4c5a9be870e06 100644 |
--- a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
+++ b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
@@ -10,15 +10,28 @@ |
#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h" |
+#include <limits> |
+ |
#include "webrtc/base/checks.h" |
+#include "webrtc/base/optional.h" |
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
namespace webrtc { |
namespace { |
-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]); |
+template <typename T, unsigned int B = sizeof(T)> |
+bool ParseInt(const uint8_t** data, size_t* remaining_size, T* value) { |
+ static_assert(std::numeric_limits<T>::is_integer, "Type must be an integer."); |
+ static_assert(sizeof(T) <= sizeof(uint64_t), |
+ "Cannot read wider than uint64_t."); |
+ static_assert(B <= sizeof(T), "T must be at least B bytes wide."); |
+ if (B > *remaining_size) |
+ return false; |
+ uint64_t val = ByteReader<uint64_t, B>::ReadBigEndian(*data); |
+ *data += B; |
+ *remaining_size -= B; |
+ *value = static_cast<T>(val); |
+ return true; |
} |
} // namespace |
@@ -34,16 +47,63 @@ void FuzzAudioDecoder(const uint8_t* data, |
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 <= remaining_size - 2) { |
- data_ptr += 2; |
- remaining_size -= 2; |
+ size_t packet_len; |
+ while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) && |
+ packet_len <= remaining_size) { |
AudioDecoder::SpeechType speech_type; |
decoder->Decode(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 identical to FuzzAudioDecoder above, with the distinction |
+// that it call DecodeRedundant instead of Decode. |
pbos-webrtc
2016/01/26 14:18:12
Can you refactor this into one method with an enum
hlundin-webrtc
2016/01/26 22:32:31
Done.
|
+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; |
+ while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) && |
+ packet_len <= remaining_size) { |
+ 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; |
+ } |
+} |
+ |
+// This function is similar to FuzzAudioDecoder, but also reads fuzzed data into |
pbos-webrtc
2016/01/26 14:18:12
Same as above if possible.
pbos-webrtc
2016/01/26 14:19:08
Not very possible, ignore this. :)
hlundin-webrtc
2016/01/26 22:32:31
Acknowledged.
|
+// 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; |
+ while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len)) { |
+ uint16_t rtp_sequence_number; |
+ if (!ParseInt(&data_ptr, &remaining_size, &rtp_sequence_number)) |
+ break; |
+ uint32_t rtp_timestamp; |
+ if (!ParseInt(&data_ptr, &remaining_size, &rtp_timestamp)) |
+ break; |
+ uint32_t arrival_timestamp; |
+ if (!ParseInt(&data_ptr, &remaining_size, &arrival_timestamp)) |
+ break; |
+ if (remaining_size < packet_len) |
+ break; |
+ decoder->IncomingPacket(data_ptr, packet_len, rtp_sequence_number, |
+ rtp_timestamp, arrival_timestamp); |
+ data_ptr += packet_len; |
+ remaining_size -= packet_len; |
} |
} |
} // namespace webrtc |