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Modify upstream webrtc NetEq to pass bounds information to speex

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Just in case

Closes #6334
// FREEBIE
This commit is contained in:
Moxie Marlinspike 2017-03-10 09:28:54 -08:00
parent 057c348d08
commit 8e25689c24
14 changed files with 22 additions and 114 deletions
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8
jni/redphone/AudioCodec.cpp
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@ -102,12 +102,14 @@ int AudioCodec::encode(short *rawData, char* encodedData, int maxEncodedDataLen)
return speex_bits_write(&enc_bits, encodedData, maxEncodedDataLen);
}
int AudioCodec::decode(char* encodedData, int encodedDataLen, short *rawData) {
int rawDataOffset = 0;
int AudioCodec::decode(char* encodedData, int encodedDataLen, short *rawData, size_t decodeMaxSize) {
uint32_t rawDataOffset = 0;
speex_bits_read_from(&dec_bits, encodedData, encodedDataLen);
while (speex_decode_int(dec, &dec_bits, rawData + rawDataOffset) == 0) { // TODO bounds?
while ((rawDataOffset + dec_frame_size <= decodeMaxSize) &&
(speex_decode_int(dec, &dec_bits, rawData + rawDataOffset) == 0))
{
WebRtcAecm_BufferFarend(aecm, rawData + rawDataOffset, dec_frame_size);
rawDataOffset += dec_frame_size;
}

2
jni/redphone/AudioCodec.h
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@ -36,7 +36,7 @@ public:
int init();
int encode(short *rawData, char* encodedData, int encodedDataLen);
int decode(char* encodedData, int encodedDataLen, short* rawData);
int decode(char* encodedData, int encodedDataLen, short* rawData, size_t decodeMaxSize);
int conceal(int frames, short *rawData);
};

5
jni/redphone/WebRtcCodec.h
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@ -17,10 +17,11 @@ public:
{}
int Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type)
int16_t* decoded, size_t decodedMaxSize,
SpeechType* speech_type)
{
*speech_type = kSpeech;
return codec.decode((char*)encoded, encoded_len, decoded);
return codec.decode((char*)encoded, encoded_len, decoded, decodedMaxSize);
}
bool HasDecodePlc() const {

104
jni/webrtc/modules/audio_coding/neteq/audio_decoder.cc
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@ -19,8 +19,9 @@ namespace webrtc {
int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int16_t* decoded,
size_t maxDecodedSize,
SpeechType* speech_type) {
return Decode(encoded, encoded_len, decoded, speech_type);
return Decode(encoded, encoded_len, decoded, maxDecodedSize, speech_type);
}
bool AudioDecoder::HasDecodePlc() const { return false; }
@ -56,48 +57,6 @@ NetEqDecoder AudioDecoder::codec_type() const { return codec_type_; }
bool AudioDecoder::CodecSupported(NetEqDecoder codec_type) {
switch (codec_type) {
case kDecoderPCMu:
case kDecoderPCMa:
case kDecoderPCMu_2ch:
case kDecoderPCMa_2ch:
#ifdef WEBRTC_CODEC_ILBC
case kDecoderILBC:
#endif
#if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
case kDecoderISAC:
#endif
#ifdef WEBRTC_CODEC_ISAC
case kDecoderISACswb:
case kDecoderISACfb:
#endif
#ifdef WEBRTC_CODEC_PCM16
case kDecoderPCM16B:
case kDecoderPCM16Bwb:
case kDecoderPCM16Bswb32kHz:
case kDecoderPCM16Bswb48kHz:
case kDecoderPCM16B_2ch:
case kDecoderPCM16Bwb_2ch:
case kDecoderPCM16Bswb32kHz_2ch:
case kDecoderPCM16Bswb48kHz_2ch:
case kDecoderPCM16B_5ch:
#endif
#ifdef WEBRTC_CODEC_G722
case kDecoderG722:
case kDecoderG722_2ch:
#endif
#ifdef WEBRTC_CODEC_CELT
case kDecoderCELT_32:
case kDecoderCELT_32_2ch:
#endif
#ifdef WEBRTC_CODEC_OPUS
case kDecoderOpus:
case kDecoderOpus_2ch:
#endif
case kDecoderRED:
case kDecoderAVT:
case kDecoderCNGnb:
case kDecoderCNGwb:
case kDecoderCNGswb32kHz:
case kDecoderCNGswb48kHz:
case kDecoderArbitrary: {
return true;
}
@ -182,65 +141,6 @@ AudioDecoder* AudioDecoder::CreateAudioDecoder(NetEqDecoder codec_type) {
switch (codec_type) {
case kDecoderPCMu:
return new AudioDecoderPcmU;
case kDecoderPCMa:
return new AudioDecoderPcmA;
case kDecoderPCMu_2ch:
return new AudioDecoderPcmUMultiCh(2);
case kDecoderPCMa_2ch:
return new AudioDecoderPcmAMultiCh(2);
#ifdef WEBRTC_CODEC_ILBC
case kDecoderILBC:
return new AudioDecoderIlbc;
#endif
#if defined(WEBRTC_CODEC_ISACFX)
case kDecoderISAC:
return new AudioDecoderIsacFix;
#elif defined(WEBRTC_CODEC_ISAC)
case kDecoderISAC:
return new AudioDecoderIsac;
#endif
#ifdef WEBRTC_CODEC_ISAC
case kDecoderISACswb:
return new AudioDecoderIsacSwb;
case kDecoderISACfb:
return new AudioDecoderIsacFb;
#endif
#ifdef WEBRTC_CODEC_PCM16
case kDecoderPCM16B:
case kDecoderPCM16Bwb:
case kDecoderPCM16Bswb32kHz:
case kDecoderPCM16Bswb48kHz:
return new AudioDecoderPcm16B(codec_type);
case kDecoderPCM16B_2ch:
case kDecoderPCM16Bwb_2ch:
case kDecoderPCM16Bswb32kHz_2ch:
case kDecoderPCM16Bswb48kHz_2ch:
case kDecoderPCM16B_5ch:
return new AudioDecoderPcm16BMultiCh(codec_type);
#endif
#ifdef WEBRTC_CODEC_G722
case kDecoderG722:
return new AudioDecoderG722;
case kDecoderG722_2ch:
return new AudioDecoderG722Stereo;
#endif
#ifdef WEBRTC_CODEC_CELT
case kDecoderCELT_32:
case kDecoderCELT_32_2ch:
return new AudioDecoderCelt(codec_type);
#endif
#ifdef WEBRTC_CODEC_OPUS
case kDecoderOpus:
case kDecoderOpus_2ch:
return new AudioDecoderOpus(codec_type);
#endif
case kDecoderCNGnb:
case kDecoderCNGwb:
case kDecoderCNGswb32kHz:
case kDecoderCNGswb48kHz:
return new AudioDecoderCng(codec_type);
case kDecoderRED:
case kDecoderAVT:
case kDecoderArbitrary:
default: {
return NULL;

3
jni/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
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@ -41,7 +41,8 @@ namespace webrtc {
// PCMu
int AudioDecoderPcmU::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t* decoded, size_t decodedSize,
SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG711_DecodeU(
state_, reinterpret_cast<int16_t*>(const_cast<uint8_t*>(encoded)),

3
jni/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
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@ -28,7 +28,8 @@ class AudioDecoderPcmU : public AudioDecoder {
public:
AudioDecoderPcmU() : AudioDecoder(kDecoderPCMu) {}
virtual int Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type);
int16_t* decoded, size_t decodedSize,
SpeechType* speech_type);
virtual int Init() { return 0; }
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len);

6
jni/webrtc/modules/audio_coding/neteq/interface/audio_decoder.h
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@ -76,12 +76,14 @@ class AudioDecoder {
// the return value. If the decoder produced comfort noise, |speech_type|
// is set to kComfortNoise, otherwise it is kSpeech.
virtual int Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) = 0;
int16_t* decoded, size_t decoded_size,
SpeechType* speech_type) = 0;
// Same as Decode(), but interfaces to the decoders redundant decode function.
// The default implementation simply calls the regular Decode() method.
virtual int DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type);
int16_t* decoded, size_t decoded_size,
SpeechType* speech_type);
// Indicates if the decoder implements the DecodePlc method.
virtual bool HasDecodePlc() const;

5
jni/webrtc/modules/audio_coding/neteq/neteq_impl.cc
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@ -1253,7 +1253,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
", len=" << packet->payload_length;
decode_length = decoder->DecodeRedundant(
packet->payload, packet->payload_length,
&decoded_buffer_[*decoded_length], speech_type);
&decoded_buffer_[*decoded_length], decoded_buffer_length_ - *decoded_length, speech_type);
} else {
LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
@ -1263,6 +1263,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
decode_length = decoder->Decode(packet->payload,
packet->payload_length,
&decoded_buffer_[*decoded_length],
decoded_buffer_length_ - *decoded_length,
speech_type);
}
@ -1591,7 +1592,7 @@ void NetEqImpl::DoCodecInternalCng() {
if (decoder) {
const uint8_t* dummy_payload = NULL;
AudioDecoder::SpeechType speech_type;
length = decoder->Decode(dummy_payload, 0, decoded_buffer, &speech_type);
length = decoder->Decode(dummy_payload, 0, decoded_buffer, kMaxFrameSize, &speech_type);
}
assert(mute_factor_array_.get());
normal_->Process(decoded_buffer, length, last_mode_, mute_factor_array_.get(),

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libs/armeabi-v7a/libnative-utils.so
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libs/armeabi-v7a/libredphone-audio.so
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libs/armeabi/libnative-utils.so
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libs/armeabi/libredphone-audio.so
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libs/x86/libnative-utils.so
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libs/x86/libredphone-audio.so
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