#include "stream_encoder_helpers.h" static unsigned round_to_bytes(unsigned bits) { return (bits + (bits % 8)) / 8; } FLAC__bool FLAC__stream_encoder_process_helper ( FLAC__StreamEncoder *encoder , FLAC__uint64 data_offset , FLAC__uint64 data_size , const char *ifile_name ) { unsigned channels = FLAC__stream_encoder_get_channels(encoder); unsigned bits_per_sample = FLAC__stream_encoder_get_bits_per_sample(encoder); unsigned msw = round_to_bytes(bits_per_sample); /* mono sample width */ unsigned block_align = channels * msw; FLAC__uint64 samples_to_process = data_size / block_align; FLAC__uint64 read_size = 4096; void *buffer_raw = malloc(read_size * block_align + 1); FLAC__int32 *buffer = malloc(read_size * sizeof(FLAC__int32) * channels); FILE *ifile = fopen(ifile_name, "r"); FLAC__uint64 samples; unsigned i; FLAC__int32 x; if (data_size % block_align) { free(buffer_raw); free(buffer); fclose(ifile); return false; } /* move position indicator to beginning of audio data */ fseek(ifile, data_offset, SEEK_SET); while (samples_to_process) { /* The reading happens by blocks. Every block has read_size samples in * it (multi-channel samples, that is). Since we will be using the * value returned by fread as indicator of whether something went * wrong or not, we need to know beforehand how many samples we need * to read. */ samples = read_size <= samples_to_process ? read_size : samples_to_process; /* Read the samples into the “raw” buffer, fail by returning false if * number of read samples differs from the expected. */ if (fread(buffer_raw, block_align, samples, ifile) != samples) { free(buffer_raw); free(buffer); fclose(ifile); return false; } /* libFLAC wants samples as FLAC__int32 values, so we need to copy * the data into an array of FLAC__int32 values, because what we have * read so far probably has different sample width. */ switch (msw) /* mono sample width in bytes */ { case 1: /* If sample width is equal or less than 8 bit, we deal with one * byte per sample and samples are unsigned as per WAVE spec. */ for (i = 0; i < samples * channels; i++) { /* Need to center the range at 0 and use signed integer as * per FLAC docs. */ *(buffer + i) = *((FLAC__uint8 *)buffer_raw + i) - 0x80; } break; case 2: /* Here we have signed samples, each having width equal to 16 * bits. */ for (i = 0; i < samples * channels; i++) { /* FIXME Only works on little-endian architectures. */ *(buffer + i) = *((FLAC__int16 *)buffer_raw + i); } break; case 3: /* Singed 24 bit samples. Going with 3 bytes step is not so * handy. */ for (i = 0; i < samples * channels; i++) { /* FIXME Only works on little-endian architectures. */ x = *(FLAC__int32 *)((FLAC__byte *)buffer_raw + i * 3); if (x & 0x800000) /* do sign extension */ x |= 0xff000000; /* negative */ else x &= 0x00ffffff; /* positive */ *(buffer + i) = x; } break; default: /* Have something else? You are screwed. */ free(buffer_raw); free(buffer); fclose(ifile); return false; } /* Finally the easy part: call FLAC encoder function and process the * block of data. */ if (!FLAC__stream_encoder_process_interleaved(encoder, buffer, samples)) { free(buffer_raw); free(buffer); fclose(ifile); return false; } samples_to_process -= samples; } free(buffer_raw); free(buffer); fclose(ifile); return true; }