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Diffstat (limited to 'src/readanysf~.cpp')
-rw-r--r--src/readanysf~.cpp48
1 files changed, 23 insertions, 25 deletions
diff --git a/src/readanysf~.cpp b/src/readanysf~.cpp
index e07c7b6..6358dfe 100644
--- a/src/readanysf~.cpp
+++ b/src/readanysf~.cpp
@@ -91,10 +91,10 @@ void m_play(t_readanysf *x) {
x->play = true; // this is the only place where play is true
} else {
if (x->is_opening ) {
- post("Current file is still starting.");
+ post("readanysf~: Current file is still starting.");
post("This probably means that it is a stream and it needs to buffer in from the network.");
} else {
- post("Current file is either invalid or an unsupported codec.");
+ post("readanysf~: Current file is either invalid or an unsupported codec.");
}
}
pthread_mutex_unlock(&x->mut);
@@ -160,10 +160,6 @@ void m_init_audio( t_readanysf *x) {
gavl_audio_frame_destroy(x->tmp_audio_frame);
x->tmp_audio_frame = gavl_audio_frame_create(&x->tmp_audio_format);
- /* m_out_audio_format.samples_per_frame = (m_out_audio_format.samplerate / (double)m_get_audio_format.samplerate) *
- m_get_audio_format.samples_per_frame + 10;
- */
-
if (x->i2t_audio_converter == NULL)
x->i2t_audio_converter = gavl_audio_converter_create( );
x->do_i2t_audio_convert = gavl_audio_converter_init( x->i2t_audio_converter, &x->in_audio_format, &x->tmp_audio_format);
@@ -172,7 +168,7 @@ void m_init_audio( t_readanysf *x) {
x->t2o_audio_converter = gavl_audio_converter_create( );
x->do_t2o_audio_convert = gavl_audio_converter_init_resample( x->t2o_audio_converter, &x->out_audio_format);
- // FIXME: this should be protected
+ // this should be protected
x->src_factor = x->out_audio_format.samplerate / (float) x->in_audio_format.samplerate;
/*
printf("in audio format: \n");
@@ -191,10 +187,14 @@ void m_open_callback( void * data) {
pthread_mutex_lock(&x->mut);
x->is_opening = true; // set it here again just to be safe
-
+ pthread_mutex_unlock(&x->mut);
+
if (x->rm->isReady() && x->rm->getAudioStreamCount() ) {
+ pthread_mutex_lock(&x->mut);
m_init_audio(x);
+ x->is_opening=false;
+ pthread_mutex_unlock(&x->mut);
// FIXME: is it safe to call these here?
SETFLOAT(&lst, (float)x->rm->getAudioSamplerate() );
@@ -210,6 +210,10 @@ void m_open_callback( void * data) {
outlet_anything(x->outinfo, gensym("ready"), 1, &lst);
// set time to 0 again here just to be sure
} else {
+ pthread_mutex_lock(&x->mut);
+ x->is_opening=false;
+ pthread_mutex_unlock(&x->mut);
+
SETFLOAT(&lst, 0.0 );
outlet_anything(x->outinfo, gensym("samplerate"), 1, &lst);
SETFLOAT(&lst, 0.0 );
@@ -217,10 +221,8 @@ void m_open_callback( void * data) {
SETFLOAT(&lst, 0.0 );
outlet_anything(x->outinfo, gensym("ready"), 1, &lst);
outlet_float(x->outinfo, 0.0);
- post("Invalid file or unsupported codec.");
+ post("readanysf~: Invalid file or unsupported codec.");
}
- x->is_opening=false;
- pthread_mutex_unlock(&x->mut);
}
void m_open(t_readanysf *x, t_symbol *s) {
@@ -259,7 +261,7 @@ void m_loop(t_readanysf *x, float f) {
x->rm->setLoop( false );
else
x->rm->setLoop( true );
- post("looping = %d", x->rm->getLoop());
+ post("readanysf~: looping = %d", x->rm->getLoop());
}
@@ -332,7 +334,7 @@ static void *readanysf_new(t_float f, t_float f2, t_float f3 ) {
outlet_float(x->outinfo, 0.0);
if (x->rm == NULL) {
x->rm = new ReadMedia ( ); // (int)sys_getsr(), x->num_channels, x->num_frames_in_fifo, x->num_samples_per_frame);
- post("Created new readanysf~ with %d channels and internal buffer of %d * %d = %d", x->num_channels,
+ post("Created new readanysf~ with %d channels and internal buffer of %d blocks of %d samples = %d", x->num_channels,
x->num_frames_in_fifo, x->num_samples_per_frame, x->num_frames_in_fifo * x->num_samples_per_frame);
}
x->rm->setOpenCallback( m_open_callback, (void *)x);
@@ -343,7 +345,7 @@ static void *readanysf_new(t_float f, t_float f2, t_float f3 ) {
int m_get_frame( t_readanysf *x ) {
int ret =0;
ret = x->rm->decodeAudio(x->in_audio_frame);
- if (ret != 1) // EOF
+ if (ret != 1) // EOF=0 or error=-1
return ret;
if (x->do_i2t_audio_convert) {
@@ -356,6 +358,7 @@ int m_get_frame( t_readanysf *x ) {
}
if ( x->do_t2o_audio_convert ) { // should be true all of the time
+ // protect src_factor here?
gavl_audio_converter_resample( x->t2o_audio_converter, x->tmp_audio_frame, x->out_audio_frame, x->src_factor );
// Don't know why, but on the first conversion, I get one extra sample
// THIS SHOULD NOT HAPPEN...this is a fix for now..check it out later.
@@ -382,8 +385,6 @@ int m_decode_block( t_readanysf * x ) {
while( samps_to_do > 0) {
if ( samps_to_do <= x->samplesleft) {
- //if (x->out_audio_frame->valid_samples < x->samplesleft)
- // printf("error\n");
// copy our samples out to the pd audio buffer
for (i = 0; i < x->num_channels; i++) {
for (j = 0; j < samps_to_do ; j++) {
@@ -395,8 +396,6 @@ int m_decode_block( t_readanysf * x ) {
samps_to_do = 0;
break;
} else if ( x->samplesleft > 0 ) {
- //if( x->out_audio_frame->valid_samples < x->samplesleft)
- // printf("valid_samples < samplesleft, shouldn't happen\n");
for (i = 0; i < x->num_channels; i++) {
for (j = 0; j < x->samplesleft; j++) {
x->x_outvec[i][samps_done + j] = x->out_audio_frame->channels.f[i][ x->out_audio_frame->valid_samples - x->samplesleft +j ];
@@ -407,10 +406,10 @@ int m_decode_block( t_readanysf * x ) {
x->samplesleft = 0;
} else { // samplesleft is zero
int ret = m_get_frame(x);
- if (ret == 0) {
+ if (ret == 0) { // EOF
return samps_done;
- } else if (ret == -1) {
- //printf("error getting frame...must be seeking\n");
+ } else if (ret == -1) { // error, file proly not ready
+ printf("error getting frame...must be seeking\n");
return ret;
}
}
@@ -432,6 +431,7 @@ static t_int *readanysf_perform(t_int *w) {
outlet_bang(x->outinfo);
} else if (samples_returned == -1) {
// error in getting audio, normally from seeking
+ post("readanysf~: error on m_decode_block inside perform()");
samples_returned=0;
}
}
@@ -477,14 +477,13 @@ void readanysf_dsp(t_readanysf *x, t_signal **sp) {
x->out_audio_format.channel_locations[0] = GAVL_CHID_NONE; // Reset
// leave enough room in our out format and frame for resampling
- x->out_audio_format.samples_per_frame = x->num_samples_per_frame * SRC_MAX +10;
+ x->out_audio_format.samples_per_frame = ( x->num_samples_per_frame * SRC_MAX ) +10;
gavl_set_channel_setup (&x->out_audio_format); // Set channel locations
if(x->out_audio_frame != NULL)
gavl_audio_frame_destroy( x->out_audio_frame);
x->out_audio_frame = gavl_audio_frame_create(&x->out_audio_format);
- //printf("created new out frame in readanysf_dsp\n");
- post("pd blocksize=%d, spf=%d", x->blocksize, x->num_samples_per_frame);
+ //post("readanysf~: pd blocksize=%d, spf=%d", x->blocksize, x->num_samples_per_frame);
}
for (i = 0; i < x->num_channels; i++)
@@ -496,7 +495,6 @@ void readanysf_dsp(t_readanysf *x, t_signal **sp) {
static void readanysf_free(t_readanysf *x) {
// delete the readany objs
-
if (x->in_audio_frame != NULL) gavl_audio_frame_destroy(x->in_audio_frame);
if (x->tmp_audio_frame != NULL) gavl_audio_frame_destroy(x->tmp_audio_frame);
if (x->out_audio_frame != NULL) gavl_audio_frame_destroy(x->out_audio_frame);
generated by cgit v1.2.1 (git 2.18.0) at 2024-05-03 14:31:04 +0000