generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Protect ACM decoder buffer in stereo.

Browse Source 
In https://code.google.com/p/webrtc/source/detail?r=8730, I did a protection on ACM decoder buffer from being overflow.

However, the I misunderstood the return unit for PacketDuration(), and therefore, stereo decoders are not well protected.

This CL fixed this.

BUG=4361
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47289004

Cr-Commit-Position: refs/heads/master@{#9275}
This commit is contained in:
Minyue 2015-05-25 13:49:37 +02:00
parent 57e5fd2e60
commit 323b132f5e
4 changed files with 28 additions and 24 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/audio_coding/codecs/audio_decoder.cc
View File

@ -20,7 +20,8 @@ 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) {
int duration = PacketDuration(encoded, encoded_len);
if (duration >= 0 && duration * sizeof(int16_t) > max_decoded_bytes) {
if (duration >= 0 &&
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
return -1;
}
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
@ -31,7 +32,8 @@ 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) {
int duration = PacketDurationRedundant(encoded, encoded_len);
if (duration >= 0 && duration * sizeof(int16_t) > max_decoded_bytes) {
if (duration >= 0 &&
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
return -1;
}
return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,

16
webrtc/modules/audio_coding/codecs/audio_decoder.h
View File

@ -36,8 +36,8 @@ class AudioDecoder {
// Decodes |encode_len| bytes from |encoded| and writes the result in
// |decoded|. The maximum bytes allowed to be written into |decoded| is
// |max_decoded_bytes|. The number of samples from all channels produced is
// in the return value. If the decoder produced comfort noise, |speech_type|
// |max_decoded_bytes|. Returns the total number of samples across all
// channels. If the decoder produced comfort noise, |speech_type|
// is set to kComfortNoise, otherwise it is kSpeech. The desired output
// sample rate is provided in |sample_rate_hz|, which must be valid for the
// codec at hand.
@ -77,14 +77,14 @@ class AudioDecoder {
// Returns the last error code from the decoder.
virtual int ErrorCode();
// Returns the duration in samples of the payload in |encoded| which is
// |encoded_len| bytes long. Returns kNotImplemented if no duration estimate
// is available, or -1 in case of an error.
// Returns the duration in samples-per-channel of the payload in |encoded|
// which is |encoded_len| bytes long. Returns kNotImplemented if no duration
// estimate is available, or -1 in case of an error.
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
// Returns the duration in samples of the redandant payload in |encoded| which
// is |encoded_len| bytes long. Returns kNotImplemented if no duration
// estimate is available, or -1 in case of an error.
// Returns the duration in samples-per-channel of the redandant payload in
// |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
// duration estimate is available, or -1 in case of an error.
virtual int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const;

5
webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
View File

@ -288,7 +288,8 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
* - payload : Encoded data pointer
* - payload_length_bytes : Bytes of encoded data
*
* Return value : The duration of the packet, in samples.
* Return value : The duration of the packet, in samples per
* channel.
*/
int WebRtcOpus_DurationEst(OpusDecInst* inst,
const uint8_t* payload,
@ -308,7 +309,7 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
* - payload_length_bytes : Bytes of encoded data
*
* Return value : >0 - The duration of the FEC data in the
* packet in samples.
* packet in samples per channel.
* 0 - No FEC data in the packet.
*/
int WebRtcOpus_FecDurationEst(const uint8_t* payload,

25
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -805,16 +805,16 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
UseNoMocks();
CreateInstance();
static const size_t kNetEqMaxFrameSize = 2880; // 60 ms @ 48 kHz.
static const int kChannels = 2;
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const int kChannles = 1;
const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000; // 10 ms.
const size_t kPayloadLengthBytes = 1;
uint8_t payload[kPayloadLengthBytes]= {0};
int16_t dummy_output[kPayloadLengthSamples] = {0};
int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
WebRtcRTPHeader rtp_header;
rtp_header.header.payloadType = kPayloadType;
rtp_header.header.sequenceNumber = 0x1234;
@ -829,7 +829,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
MOCK_METHOD5(DecodeInternal, int(const uint8_t*, size_t, int, int16_t*,
SpeechType*));
size_t Channels() const override { return 1; }
size_t Channels() const override { return kChannels; }
} decoder_;
const uint8_t kFirstPayloadValue = 1;
@ -838,7 +838,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
EXPECT_CALL(decoder_, PacketDuration(Pointee(kFirstPayloadValue),
kPayloadLengthBytes))
.Times(AtLeast(1))
.WillRepeatedly(Return(kNetEqMaxFrameSize * kChannles + 1));
.WillRepeatedly(Return(kNetEqMaxFrameSize + 1));
EXPECT_CALL(decoder_,
DecodeInternal(Pointee(kFirstPayloadValue), _, _, _, _))
@ -849,14 +849,15 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
kSampleRateHz, _, _))
.Times(1)
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
dummy_output +
kPayloadLengthSamples * kChannels),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples)));
Return(kPayloadLengthSamples * kChannels)));
EXPECT_CALL(decoder_, PacketDuration(Pointee(kSecondPayloadValue),
kPayloadLengthBytes))
.Times(AtLeast(1))
.WillRepeatedly(Return(kNetEqMaxFrameSize * kChannles));
.WillRepeatedly(Return(kNetEqMaxFrameSize));
EXPECT_EQ(NetEq::kOK,
neteq_->RegisterExternalDecoder(
@ -878,7 +879,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
const int kMaxOutputSize = 10 * kSampleRateHz / 1000 * kChannels;
int16_t output[kMaxOutputSize];
int samples_per_channel;
int num_channels;
@ -888,14 +889,14 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
&samples_per_channel, &num_channels,
&type));
EXPECT_EQ(NetEq::kOtherDecoderError, neteq_->LastError());
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
EXPECT_EQ(kChannles, num_channels);
EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels);
EXPECT_EQ(kChannels, num_channels);
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(kMaxOutputSize, output,
&samples_per_channel, &num_channels,
&type));
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
EXPECT_EQ(kChannles, num_channels);
EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels);
EXPECT_EQ(kChannels, num_channels);
}
} // namespace webrtc