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fix: Fixed typos
1 parent eccd176 commit aab7b1a

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libjamesdsp/JdspImpResToolbox.c

Lines changed: 5 additions & 5 deletions
Original file line numberDiff line numberDiff line change
@@ -222,13 +222,13 @@ float* loadAudioFile(const char *filename, double targetFs, unsigned int *channe
222222
const char *ext = get_filename_ext(filename);
223223
float *pSampleData = 0;
224224
if (!strncmp(ext, "wav", 5) || !strncmp(ext, "irs", 5))
225-
pSampleData = drwav_open_file_and_read_pcm_frames_f32(filename, channels, &fs, totalPCMFrameCount, SRC_SINC_BEST_QUALITY);
225+
pSampleData = drwav_open_file_and_read_pcm_frames_f32(filename, channels, &fs, totalPCMFrameCount, 0);
226226
if (!strncmp(ext, "flac", 5))
227-
pSampleData = drflac_open_file_and_read_pcm_frames_f32(filename, channels, &fs, totalPCMFrameCount, SRC_SINC_BEST_QUALITY);
227+
pSampleData = drflac_open_file_and_read_pcm_frames_f32(filename, channels, &fs, totalPCMFrameCount, 0);
228228
/*if (!strncmp(ext, "mp3", 5))
229229
{
230230
drmp3_config mp3Conf;
231-
pSampleData = drmp3_open_file_and_read_pcm_frames_f32(filename, &mp3Conf, totalPCMFrameCount, SRC_SINC_BEST_QUALITY);
231+
pSampleData = drmp3_open_file_and_read_pcm_frames_f32(filename, &mp3Conf, totalPCMFrameCount, 0);
232232
*channels = mp3Conf.channels;
233233
fs = mp3Conf.sampleRate;
234234
}*/
@@ -270,7 +270,7 @@ float* ReadImpulseResponseToFloat
270270
if (strlen(mIRFileName) <= 0) return 0;
271271
unsigned int channels;
272272
drwav_uint64 frameCount;
273-
float *pFrameBuffer = loadAudioFile(mIRFileName, targetSampleRate, &channels, &frameCount, 1);
273+
float *pFrameBuffer = loadAudioFile(mIRFileName, targetSampleRate, &channels, &frameCount, SRC_SINC_BEST_QUALITY);
274274
if (channels == 0 || channels == 3 || channels > 4)
275275
{
276276
free(pFrameBuffer);
@@ -395,7 +395,7 @@ float* ReadImpulseResponseToFloat
395395
snprintf(filenameIR, needed, "%s%s", jnipath, mIRFileName);
396396
unsigned int channels;
397397
drwav_uint64 frameCount;
398-
float *pFrameBuffer = loadAudioFile(filenameIR, targetSampleRate, &channels, &frameCount, 0);
398+
float *pFrameBuffer = loadAudioFile(filenameIR, targetSampleRate, &channels, &frameCount, SRC_SINC_BEST_QUALITY);
399399
free(filenameIR);
400400
if (!pFrameBuffer)
401401
{

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