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Add functions to interact with ASan and MSan, and some sample uses

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The sample uses are from when I debugged bug 617124. The change in neteq_network_stats_unittest.cc is a fix for a minor unrelated bug found by the try bots when I tried to land this CL (a test was passing uninitialized packet data to NetEq).

BUG=chromium:617124

Review-Url: https://codereview.webrtc.org/2293893002
Cr-Commit-Position: refs/heads/master@{#14034}
This commit is contained in:
kwiberg
2016-09-02 00:39:33 -07:00
committed by Commit bot
parent 97d2dacf9f
commit ac554eebb9
10 changed files with 145 additions and 2 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/audio_coding/codecs/audio_decoder.cc
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@ -12,7 +12,9 @@
#include <assert.h>
#include "webrtc/base/array_view.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/sanitizer.h"
#include "webrtc/base/trace_event.h"
namespace webrtc {
@ -21,6 +23,7 @@ int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
int sample_rate_hz, size_t max_decoded_bytes,
int16_t* decoded, SpeechType* speech_type) {
TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
int duration = PacketDuration(encoded, encoded_len);
if (duration >= 0 &&
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
@ -34,6 +37,7 @@ int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
int sample_rate_hz, size_t max_decoded_bytes,
int16_t* decoded, SpeechType* speech_type) {
TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
int duration = PacketDurationRedundant(encoded, encoded_len);
if (duration >= 0 &&
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {

4
webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
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@ -18,6 +18,7 @@
#include "defines.h"
#include "constants.h"
#include "webrtc/base/sanitizer.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
/*----------------------------------------------------------------*
@ -40,6 +41,9 @@ void WebRtcIlbcfix_CreateAugmentedVec(
*/
size_t interp_len = WEBRTC_SPL_MIN(index, 4);
rtc_MsanCheckInitialized(buffer - index - interp_len, sizeof(buffer[0]),
index + interp_len);
ilow = index - interp_len;
/* copy the first noninterpolated part */

2
webrtc/modules/audio_coding/neteq/neteq_impl.cc
View File

@ -19,6 +19,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/sanitizer.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
@ -132,6 +133,7 @@ NetEqImpl::~NetEqImpl() = default;
int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) {
rtc::MsanCheckInitialized(payload);
TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
rtc::CritScope lock(&crit_sect_);
int error =

4
webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View File

@ -201,7 +201,8 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
frame_size_samples_,
&rtp_header_);
if (!Lost(next_send_time)) {
InsertPacket(rtp_header_, payload_, next_send_time);
static const uint8_t payload[kPayloadSizeByte] = {0};
InsertPacket(rtp_header_, payload, next_send_time);
}
}
GetOutputAudio(&output_frame_);
@ -277,7 +278,6 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
WebRtcRTPHeader rtp_header_;
uint32_t last_lost_time_;
uint32_t packet_loss_interval_;
uint8_t payload_[kPayloadSizeByte];
AudioFrame output_frame_;
};