int main(int argc, char **argv)
{
demux_res_t demux_res;
unsigned int output_size, i;
set_endian();
setup_environment(argc, argv);
input_stream = stream_create_file(input_file, 1);
if (!input_stream)
{
fprintf(stderr, "failed to create input stream from file\n");
return 0;
}
/* if qtmovie_read returns successfully, the stream is up to
* the movie data, which can be used directly by the decoder */
if (!qtmovie_read(input_stream, &demux_res))
{
fprintf(stderr, "failed to load the QuickTime movie headers\n");
return 0;
}
/* initialise the sound converter */
init_sound_converter(&demux_res);
/* write wav output headers */
if (write_wav_format)
{
/* calculate output size */
output_size = 0;
for (i = 0; i < demux_res.num_sample_byte_sizes; i++)
{
unsigned int thissample_duration;
unsigned int thissample_bytesize;
get_sample_info(&demux_res, i, &thissample_duration,
&thissample_bytesize);
output_size += thissample_duration * (demux_res.sample_size / 8)
* demux_res.num_channels;
}
wavwriter_writeheaders(output_file,
output_size,
demux_res.num_channels,
demux_res.sample_rate,
demux_res.sample_size);
}
/* will convert the entire buffer */
GetBuffer(&demux_res);
stream_destroy(input_stream);
if (output_opened)
fclose(output_file);
if (input_opened)
fclose(input_file);
return 0;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
/* Note that when dealing with QuickTime/MPEG4 files, terminology is
* a bit confusing. Files with sound are split up in chunks, where
* each chunk contains one or more samples. Each sample in turn
* contains a number of "sound samples" (the kind you refer to with
* the sampling frequency).
*/
size_t n;
demux_res_t demux_res;
stream_t input_stream;
uint32_t sound_samples_done;
uint32_t elapsed_time;
int file_offset;
int framelength;
int lead_trim = 0;
unsigned int frame_samples;
unsigned int i;
unsigned char* buffer;
NeAACDecFrameInfo frame_info;
NeAACDecHandle decoder;
int err;
uint32_t seek_idx = 0;
uint32_t s = 0;
uint32_t sbr_fac = 1;
unsigned char c = 0;
void *ret;
intptr_t param;
bool empty_first_frame = false;
/* Clean and initialize decoder structures */
memset(&demux_res , 0, sizeof(demux_res));
if (codec_init()) {
LOGF("FAAD: Codec init error\n");
return CODEC_ERROR;
}
file_offset = ci->id3->offset;
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
stream_create(&input_stream,ci);
ci->seek_buffer(ci->id3->first_frame_offset);
/* if qtmovie_read returns successfully, the stream is up to
* the movie data, which can be used directly by the decoder */
if (!qtmovie_read(&input_stream, &demux_res)) {
LOGF("FAAD: File init error\n");
return CODEC_ERROR;
}
/* initialise the sound converter */
decoder = NeAACDecOpen();
if (!decoder) {
LOGF("FAAD: Decode open error\n");
return CODEC_ERROR;
}
NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
NeAACDecSetConfiguration(decoder, conf);
err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
if (err) {
LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
return CODEC_ERROR;
}
#ifdef SBR_DEC
/* Check for need of special handling for seek/resume and elapsed time. */
if (ci->id3->needs_upsampling_correction) {
sbr_fac = 2;
} else {
sbr_fac = 1;
}
#endif
i = 0;
if (file_offset > 0) {
/* Resume the desired (byte) position. Important: When resuming SBR
* upsampling files the resulting sound_samples_done must be expanded
* by a factor of 2. This is done via using sbr_fac. */
if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
&sound_samples_done, (int*) &i)) {
sound_samples_done *= sbr_fac;
} else {
sound_samples_done = 0;
}
NeAACDecPostSeekReset(decoder, i);
} else {
sound_samples_done = 0;
}
elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsed_time);
if (i == 0)
{
lead_trim = ci->id3->lead_trim;
}
/* The main decoding loop */
while (i < demux_res.num_sample_byte_sizes) {
enum codec_command_action action = ci->get_command(¶m);
if (action == CODEC_ACTION_HALT)
break;
/* Deal with any pending seek requests */
if (action == CODEC_ACTION_SEEK_TIME) {
/* Seek to the desired time position. Important: When seeking in SBR
* upsampling files the seek_time must be divided by 2 when calling
* m4a_seek and the resulting sound_samples_done must be expanded
* by a factor 2. This is done via using sbr_fac. */
if (m4a_seek(&demux_res, &input_stream,
(param/10/sbr_fac)*(ci->id3->frequency/100),
&sound_samples_done, (int*) &i)) {
sound_samples_done *= sbr_fac;
elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsed_time);
seek_idx = 0;
if (i == 0)
{
lead_trim = ci->id3->lead_trim;
}
}
NeAACDecPostSeekReset(decoder, i);
ci->seek_complete();
}
/* There can be gaps between chunks, so skip ahead if needed. It
* doesn't seem to happen much, but it probably means that a
* "proper" file can have chunks out of order. Why one would want
* that an good question (but files with gaps do exist, so who
* knows?), so we don't support that - for now, at least.
*/
file_offset = m4a_check_sample_offset(&demux_res, i, &seek_idx);
if (file_offset > ci->curpos)
{
ci->advance_buffer(file_offset - ci->curpos);
}
else if (file_offset == 0)
{
LOGF("AAC: get_sample_offset error\n");
return CODEC_ERROR;
}
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_SIZE);
/* Decode one block - returned samples will be host-endian */
ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
/* NeAACDecDecode may sometimes return NULL without setting error. */
if (ret == NULL || frame_info.error > 0) {
LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
return CODEC_ERROR;
}
/* Advance codec buffer (no need to call set_offset because of this) */
ci->advance_buffer(frame_info.bytesconsumed);
/* Output the audio */
ci->yield();
frame_samples = frame_info.samples >> 1;
if (empty_first_frame)
{
/* Remove the first frame from lead_trim, under the assumption
* that it had the same size as this frame
*/
empty_first_frame = false;
lead_trim -= frame_samples;
if (lead_trim < 0)
{
lead_trim = 0;
}
}
/* Gather number of samples for the decoded frame. */
framelength = frame_samples - lead_trim;
if (i == demux_res.num_sample_byte_sizes - 1)
{
// Size of the last frame
const uint32_t sample_duration = (demux_res.num_time_to_samples > 0) ?
demux_res.time_to_sample[demux_res.num_time_to_samples - 1].sample_duration :
frame_samples;
/* Currently limited to at most one frame of tail_trim.
* Seems to be enough.
*/
if (ci->id3->tail_trim == 0 && sample_duration < frame_samples)
{
/* Subtract lead_trim just in case we decode a file with only
* one audio frame with actual data (lead_trim is usually zero
* here).
*/
framelength = sample_duration - lead_trim;
}
else
{
framelength -= ci->id3->tail_trim;
}
}
if (framelength > 0)
{
ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
&decoder->time_out[1][lead_trim],
framelength);
sound_samples_done += framelength;
/* Update the elapsed-time indicator */
elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsed_time);
}
if (lead_trim > 0)
{
/* frame_info.samples can be 0 for frame 0. We still want to
* remove it from lead_trim, so do that during frame 1.
*/
if (0 == i && 0 == frame_info.samples)
{
empty_first_frame = true;
}
lead_trim -= frame_samples;
if (lead_trim < 0)
{
lead_trim = 0;
}
}
++i;
}
LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
return CODEC_OK;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
size_t n;
demux_res_t demux_res;
stream_t input_stream;
uint32_t samplesdone;
uint32_t elapsedtime = 0;
int samplesdecoded;
unsigned int i;
unsigned char* buffer;
alac_file alac;
intptr_t param;
/* Clean and initialize decoder structures */
memset(&demux_res , 0, sizeof(demux_res));
if (codec_init()) {
LOGF("ALAC: Error initialising codec\n");
return CODEC_ERROR;
}
ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
ci->seek_buffer(0);
stream_create(&input_stream,ci);
/* Read from ci->id3->offset before calling qtmovie_read. */
samplesdone = (uint32_t)(((uint64_t)(ci->id3->offset) * ci->id3->frequency) /
(ci->id3->bitrate*128));
/* if qtmovie_read returns successfully, the stream is up to
* the movie data, which can be used directly by the decoder */
if (!qtmovie_read(&input_stream, &demux_res)) {
LOGF("ALAC: Error initialising file\n");
return CODEC_ERROR;
}
/* initialise the sound converter */
alac_set_info(&alac, demux_res.codecdata);
/* Set i for first frame, seek to desired sample position for resuming. */
i=0;
if (samplesdone > 0) {
if (m4a_seek(&demux_res, &input_stream, samplesdone,
&samplesdone, (int*) &i)) {
elapsedtime = (samplesdone * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsedtime);
} else {
samplesdone = 0;
}
}
ci->set_elapsed(elapsedtime);
/* The main decoding loop */
while (i < demux_res.num_sample_byte_sizes) {
enum codec_command_action action = ci->get_command(¶m);
if (action == CODEC_ACTION_HALT)
break;
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
/* Deal with any pending seek requests */
if (action == CODEC_ACTION_SEEK_TIME) {
if (m4a_seek(&demux_res, &input_stream,
(param/10) * (ci->id3->frequency/100),
&samplesdone, (int *)&i)) {
elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
}
ci->set_elapsed(elapsedtime);
ci->seek_complete();
}
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
/* Decode one block - returned samples will be host-endian */
samplesdecoded=alac_decode_frame(&alac, buffer, outputbuffer, ci->yield);
ci->yield();
/* Advance codec buffer by amount of consumed bytes */
ci->advance_buffer(alac.bytes_consumed);
/* Output the audio */
ci->pcmbuf_insert(outputbuffer[0], outputbuffer[1], samplesdecoded);
/* Update the elapsed-time indicator */
samplesdone+=samplesdecoded;
elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
ci->set_elapsed(elapsedtime);
i++;
}
LOGF("ALAC: Decoded %lu samples\n",(unsigned long)samplesdone);
return CODEC_OK;
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
size_t n;
demux_res_t demux_res;
stream_t input_stream;
uint32_t samplesdone;
uint32_t elapsedtime;
uint32_t sample_duration;
uint32_t sample_byte_size;
int samplesdecoded;
unsigned int i;
unsigned char* buffer;
alac_file alac;
int retval;
/* Generic codec initialisation */
ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, ALAC_OUTPUT_DEPTH-1);
next_track:
if (codec_init()) {
LOGF("ALAC: Error initialising codec\n");
retval = CODEC_ERROR;
goto exit;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
stream_create(&input_stream,ci);
/* Read from ci->id3->offset before calling qtmovie_read. */
samplesdone = (uint32_t)(((uint64_t)(ci->id3->offset) * ci->id3->frequency) /
(ci->id3->bitrate*128));
/* if qtmovie_read returns successfully, the stream is up to
* the movie data, which can be used directly by the decoder */
if (!qtmovie_read(&input_stream, &demux_res)) {
LOGF("ALAC: Error initialising file\n");
retval = CODEC_ERROR;
goto done;
}
/* initialise the sound converter */
create_alac(demux_res.sound_sample_size, demux_res.num_channels,&alac);
alac_set_info(&alac, demux_res.codecdata);
/* Set i for first frame, seek to desired sample position for resuming. */
i=0;
if (samplesdone > 0) {
if (alac_seek(&demux_res, &input_stream, samplesdone,
&samplesdone, (int*) &i)) {
elapsedtime = (samplesdone * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsedtime);
} else {
samplesdone = 0;
}
}
/* The main decoding loop */
while (i < demux_res.num_sample_byte_sizes) {
ci->yield();
if (ci->stop_codec || ci->new_track) {
break;
}
/* Deal with any pending seek requests */
if (ci->seek_time) {
if (alac_seek(&demux_res, &input_stream,
((ci->seek_time-1)/10) * (ci->id3->frequency/100),
&samplesdone, (int *)&i)) {
elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
ci->set_elapsed(elapsedtime);
}
ci->seek_complete();
}
/* Lookup the length (in samples and bytes) of block i */
if (!get_sample_info(&demux_res, i, &sample_duration,
&sample_byte_size)) {
LOGF("ALAC: Error in get_sample_info\n");
retval = CODEC_ERROR;
goto done;
}
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n,sample_byte_size);
if (n!=sample_byte_size) {
retval = CODEC_ERROR;
goto done;
}
/* Decode one block - returned samples will be host-endian */
ci->yield();
samplesdecoded=alac_decode_frame(&alac, buffer, outputbuffer, ci->yield);
/* Advance codec buffer n bytes */
ci->advance_buffer(n);
/* Output the audio */
ci->yield();
ci->pcmbuf_insert(outputbuffer[0], outputbuffer[1], samplesdecoded);
/* Update the elapsed-time indicator */
samplesdone+=sample_duration;
elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
ci->set_elapsed(elapsedtime);
i++;
}
retval = CODEC_OK;
done:
LOGF("ALAC: Decoded %ld samples\n",samplesdone);
if (ci->request_next_track())
goto next_track;
exit:
return retval;
}
