Chromium Code Reviews| Index: webrtc/modules/audio_coding/codecs/isac/main/source/isac.c |
| diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c |
| index ac211e9a2c2429667a494d9fadf1deb4ca260546..190277eb66ee98c5fc1cdd17495bdf989c6f4c2a 100644 |
| --- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c |
| +++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c |
| @@ -507,7 +507,7 @@ int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, |
| int streamLenLB = 0; |
| int streamLenUB = 0; |
| int streamLen = 0; |
| - int16_t k = 0; |
| + size_t k = 0; |
| uint8_t garbageLen = 0; |
| int32_t bottleneck = 0; |
| int16_t bottleneckIdx = 0; |
| @@ -528,12 +528,12 @@ int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, |
| if (instISAC->in_sample_rate_hz == 48000) { |
| /* Samples in 10 ms @ 48 kHz. */ |
| - const int kNumInputSamples = FRAMESAMPLES_10ms * 3; |
| + const size_t kNumInputSamples = FRAMESAMPLES_10ms * 3; |
| /* Samples 10 ms @ 32 kHz. */ |
| - const int kNumOutputSamples = FRAMESAMPLES_10ms * 2; |
| + const size_t kNumOutputSamples = FRAMESAMPLES_10ms * 2; |
| /* Resampler divide the input into blocks of 3 samples, i.e. |
| * kNumInputSamples / 3. */ |
| - const int kNumResamplerBlocks = FRAMESAMPLES_10ms; |
| + const size_t kNumResamplerBlocks = FRAMESAMPLES_10ms; |
| int32_t buffer32[FRAMESAMPLES_10ms * 3 + SIZE_RESAMPLER_STATE]; |
| /* Restore last samples from the past to the beginning of the buffer |
| @@ -1006,7 +1006,7 @@ int16_t WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) { |
| */ |
| int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst, |
| const uint8_t* encoded, |
| - int32_t packet_size, |
| + size_t packet_size, |
| uint16_t rtp_seq_number, |
| uint32_t send_ts, |
| uint32_t arr_ts) { |
| @@ -1056,7 +1056,7 @@ int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst, |
| static int Decode(ISACStruct* ISAC_main_inst, |
| const uint8_t* encoded, |
| - int16_t lenEncodedBytes, |
| + size_t lenEncodedBytes, |
| int16_t* decoded, |
| int16_t* speechType, |
| int16_t isRCUPayload) { |
| @@ -1069,13 +1069,14 @@ static int Decode(ISACStruct* ISAC_main_inst, |
| float outFrame[MAX_FRAMESAMPLES]; |
| int16_t outFrameLB[MAX_FRAMESAMPLES]; |
| int16_t outFrameUB[MAX_FRAMESAMPLES]; |
| - int numDecodedBytesLB; |
| + int numDecodedBytesLBint; |
| + size_t numDecodedBytesLB; |
| int numDecodedBytesUB; |
| - int16_t lenEncodedLBBytes; |
| + size_t lenEncodedLBBytes; |
| int16_t validChecksum = 1; |
| int16_t k; |
| uint16_t numLayer; |
| - int16_t totSizeBytes; |
| + size_t totSizeBytes; |
| int16_t err; |
| ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; |
| @@ -1089,7 +1090,7 @@ static int Decode(ISACStruct* ISAC_main_inst, |
| return -1; |
| } |
| - if (lenEncodedBytes <= 0) { |
| + if (lenEncodedBytes == 0) { |
| /* return error code if the packet length is null. */ |
|
minyue-webrtc
2015/07/31 14:10:14
I am not sure if lenEncodedBytes < 0 has been a va
Peter Kasting
2015/08/04 02:02:53
If I changed the argument type, then I also change
|
| instISAC->errorCode = ISAC_EMPTY_PACKET; |
| return -1; |
| @@ -1115,11 +1116,12 @@ static int Decode(ISACStruct* ISAC_main_inst, |
| /* Regardless of that the current codec is setup to work in |
| * wideband or super-wideband, the decoding of the lower-band |
| * has to be performed. */ |
| - numDecodedBytesLB = WebRtcIsac_DecodeLb(&instISAC->transform_tables, |
| - outFrame, decInstLB, |
| - &numSamplesLB, isRCUPayload); |
| - |
| - if ((numDecodedBytesLB < 0) || (numDecodedBytesLB > lenEncodedLBBytes) || |
| + numDecodedBytesLBint = WebRtcIsac_DecodeLb(&instISAC->transform_tables, |
| + outFrame, decInstLB, |
| + &numSamplesLB, isRCUPayload); |
| + numDecodedBytesLB = (size_t)numDecodedBytesLBint; |
| + if ((numDecodedBytesLBint < 0) || |
| + (numDecodedBytesLB > lenEncodedLBBytes) || |
| (numSamplesLB > MAX_FRAMESAMPLES)) { |
| instISAC->errorCode = ISAC_LENGTH_MISMATCH; |
| return -1; |
| @@ -1362,7 +1364,7 @@ static int Decode(ISACStruct* ISAC_main_inst, |
| int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst, |
| const uint8_t* encoded, |
| - int16_t lenEncodedBytes, |
| + size_t lenEncodedBytes, |
| int16_t* decoded, |
| int16_t* speechType) { |
| int16_t isRCUPayload = 0; |
| @@ -1394,7 +1396,7 @@ int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst, |
| int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst, |
| const uint8_t* encoded, |
| - int16_t lenEncodedBytes, |
| + size_t lenEncodedBytes, |
| int16_t* decoded, |
| int16_t* speechType) { |
| int16_t isRCUPayload = 1; |
| @@ -1417,13 +1419,12 @@ int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst, |
| * Output: |
| * - decoded : The decoded vector |
| * |
| - * Return value : >0 - number of samples in decoded PLC vector |
| - * -1 - Error |
| + * Return value : Number of samples in decoded PLC vector |
| */ |
| -int16_t WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst, |
| - int16_t* decoded, |
| - int16_t noOfLostFrames) { |
| - int16_t numSamples = 0; |
| +size_t WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst, |
| + int16_t* decoded, |
| + size_t noOfLostFrames) { |
| + size_t numSamples = 0; |
| ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; |
| /* Limit number of frames to two = 60 millisecond. |