问题
I want to create a audio player that support Flac format of audio files.For this i tried to implement the algorithm of flac to wav conversion which is as follow
Please help me.
All time it gives me error
ERROR: initializing decoder: FLAC__STREAM_DECODER_INIT_STATUS_ERROR_OPENING_FILE
static FLAC__bool write_little_endian_uint16(FILE *f, FLAC__uint16 x)
{
return
fputc(x, f) != EOF &&
fputc(x >> 8, f) != EOF
;
}
static FLAC__bool write_little_endian_int16(FILE *f, FLAC__int16 x)
{
return write_little_endian_uint16(f, (FLAC__uint16)x);
}
static FLAC__bool write_little_endian_uint32(FILE *f, FLAC__uint32 x)
{
return
fputc(x, f) != EOF &&
fputc(x >> 8, f) != EOF &&
fputc(x >> 16, f) != EOF &&
fputc(x >> 24, f) != EOF
;
}
int main(int argc, char *argv[])
{
const char *input = "demo_audio_shaer.flac";
printf("aa[0]====%s",argv[0]);
FLAC__bool ok = true;
FLAC__StreamDecoder *decoder = 0;
FLAC__StreamDecoderInitStatus init_status;
FILE *fout;
if((fout = fopen(argv[2], "wb")) == NULL) {
fprintf(stderr, "ERROR: opening %s for output\n", argv[2]);
return 1;
}
if((decoder = FLAC__stream_decoder_new()) == NULL) {
fprintf(stderr, "ERROR: allocating decoder\n");
fclose(fout);
return 1;
}
(void)FLAC__stream_decoder_set_md5_checking(decoder, true);
init_status = FLAC__stream_decoder_init_file(decoder, input, write_callback, metadata_callback, error_callback, /*client_data=*/fout);
if(init_status != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
fprintf(stderr, "ERROR: initializing decoder: %s\n", FLAC__StreamDecoderInitStatusString[init_status]);
ok = false;
}
if(ok) {
ok = FLAC__stream_decoder_process_until_end_of_stream(decoder);
fprintf(stderr, "decoding: %s\n", ok? "succeeded" : "FAILED");
fprintf(stderr, " state: %s\n", FLAC__StreamDecoderStateString[FLAC__stream_decoder_get_state(decoder)]);
}
FLAC__stream_decoder_delete(decoder);
fclose(fout);
return 0;
}
FLAC__StreamDecoderWriteStatus write_callback(const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data)
{
FILE *f = (FILE*)client_data;
const FLAC__uint32 total_size = (FLAC__uint32)(total_samples * channels * (bps/8));
size_t i;
(void)decoder;
if(total_samples == 0) {
fprintf(stderr, "ERROR: this example only works for FLAC files that have a total_samples count in STREAMINFO\n");
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
if(channels != 2 || bps != 16) {
fprintf(stderr, "ERROR: this example only supports 16bit stereo streams\n");
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
/* write WAVE header before we write the first frame */
if(frame->header.number.sample_number == 0) {
if(
fwrite("RIFF", 1, 4, f) < 4 ||
!write_little_endian_uint32(f, total_size + 36) ||
fwrite("WAVEfmt ", 1, 8, f) < 8 ||
!write_little_endian_uint32(f, 16) ||
!write_little_endian_uint16(f, 1) ||
!write_little_endian_uint16(f, (FLAC__uint16)channels) ||
!write_little_endian_uint32(f, sample_rate) ||
!write_little_endian_uint32(f, sample_rate * channels * (bps/8)) ||
!write_little_endian_uint16(f, (FLAC__uint16)(channels * (bps/8))) || /* block align */
!write_little_endian_uint16(f, (FLAC__uint16)bps) ||
fwrite("data", 1, 4, f) < 4 ||
!write_little_endian_uint32(f, total_size)
)
{
fprintf(stderr, "ERROR: write error\n");
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
}
/* write decoded PCM samples */
for(i = 0; i < frame->header.blocksize; i++) {
if(
!write_little_endian_int16(f, (FLAC__int16)buffer[0][i]) || /* left channel */
!write_little_endian_int16(f, (FLAC__int16)buffer[1][i]) /* right channel */
) {
fprintf(stderr, "ERROR: write error\n");
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
void metadata_callback(const FLAC__StreamDecoder *decoder, const FLAC__StreamMetadata *metadata, void *client_data)
{
(void)decoder, (void)client_data;
/* print some stats */
if(metadata->type == FLAC__METADATA_TYPE_STREAMINFO) {
/* save for later */
total_samples = metadata->data.stream_info.total_samples;
sample_rate = metadata->data.stream_info.sample_rate;
channels = metadata->data.stream_info.channels;
bps = metadata->data.stream_info.bits_per_sample;
fprintf(stderr, "sample rate : %u Hz\n", sample_rate);
fprintf(stderr, "channels : %u\n", channels);
fprintf(stderr, "bits per sample: %u\n", bps);
#ifdef _MSC_VER
fprintf(stderr, "total samples : %I64u\n", total_samples);
#else
fprintf(stderr, "total samples : %llu\n", total_samples);
#endif
}
}
void error_callback(const FLAC__StreamDecoder *decoder, FLAC__StreamDecoderErrorStatus status, void *client_data)
{
(void)decoder, (void)client_data;
fprintf(stderr, "Got error callback: %s\n", FLAC__StreamDecoderErrorStatusString[status]);
}
回答1:
I can suggest my solution. https://github.com/Krivoblotsky/KSAudioPlayer
It supports *.flac and many other file formats.
回答2:
Its not possible to play FLAC
file directly using AVFoundation
framework. Supported are:
*AAC
*HE-AAC
*AMR (Adaptive Multi-Rate, a format for speech)
*ALAC (Apple Lossless)
*iLBC (internet Low Bitrate Codec, another format for speech)
*IMA4 (IMA/ADPCM)
*linear PCM (uncompressed)
*µ-law and a-law
*MP3 (MPEG-1 audio layer 3
But after converting
to feasible format
as given above then it can be played
using AVFoundation
framework.
EDIT : Library for converting to wav
. Good example here
EDIT : Use FFMPEG library to play FLAC
来源:https://stackoverflow.com/questions/16080888/how-to-convert-flac-file-to-wav-file-in-ios