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Prevent crash in NetEQ when decoder overflow.

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NetEQ can crash when decoder gives too many output samples than it can handle. A practical case this happens is when multiple opus packets are combined.

The best solution is to pass the max size to the ACM decode function and let it return a failure if the max size if too small.

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

Review URL: https://webrtc-codereview.appspot.com/45619004

Cr-Commit-Position: refs/heads/master@{#8730}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8730 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org
2015-03-16 12:30:37 +00:00
parent 4b89aa03bb
commit 7f7d7e3427
19 changed files with 453 additions and 188 deletions
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84
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
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@ -38,11 +38,11 @@
namespace webrtc {
// PCMu
int AudioDecoderPcmU::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeU(encoded, static_cast<int16_t>(encoded_len),
@ -58,11 +58,11 @@ int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
}
// PCMa
int AudioDecoderPcmA::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeA(encoded, static_cast<int16_t>(encoded_len),
@ -81,11 +81,11 @@ int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
#ifdef WEBRTC_CODEC_PCM16
AudioDecoderPcm16B::AudioDecoderPcm16B() {}
int AudioDecoderPcm16B::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
@ -117,11 +117,11 @@ AudioDecoderIlbc::~AudioDecoderIlbc() {
WebRtcIlbcfix_DecoderFree(dec_state_);
}
int AudioDecoderIlbc::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded,
@ -150,11 +150,11 @@ AudioDecoderG722::~AudioDecoderG722() {
WebRtcG722_FreeDecoder(dec_state_);
}
int AudioDecoderG722::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 16000);
int16_t temp_type = 1; // Default is speech.
int16_t ret =
@ -185,11 +185,11 @@ AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
WebRtcG722_FreeDecoder(dec_state_right_);
}
int AudioDecoderG722Stereo::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 16000);
int16_t temp_type = 1; // Default is speech.
// De-interleave the bit-stream into two separate payloads.
@ -270,11 +270,11 @@ AudioDecoderOpus::~AudioDecoderOpus() {
WebRtcOpus_DecoderFree(dec_state_);
}
int AudioDecoderOpus::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcOpus_Decode(dec_state_, encoded,
@ -286,16 +286,18 @@ int AudioDecoderOpus::Decode(const uint8_t* encoded,
return ret;
}
int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
if (!PacketHasFec(encoded, encoded_len)) {
// This packet is a RED packet.
return Decode(encoded, encoded_len, sample_rate_hz, decoded, speech_type);
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
DCHECK_EQ(sample_rate_hz, 48000);
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,