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audio testcard: WAV reading to separate func

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refactorization only

+ revert a condition (use positive branch first)
This commit is contained in:
Martin Pulec
2022-10-10 09:20:14 +02:00
parent ab152abb04
commit bc0ce5e439
1 changed files with 73 additions and 61 deletions
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134
src/audio/capture/testcard.cpp
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@@ -173,6 +173,68 @@ static char *get_ebu_signal(int sample_rate, int bps, int channels, int frequenc
return ret;
}
static bool audio_testcard_read_wav(const char *wav_filename, struct audio_frame *audio_frame,
char **audio_samples, unsigned int *total_samples,
unsigned long long int *chunk_size) {
FILE *wav = fopen(wav_filename, "r");
if(!wav) {
LOG(LOG_LEVEL_ERROR) << MOD_NAME << "Unable to open WAV file: " << wav_filename << ".\n";
return false;
}
struct wav_metadata metadata;
int ret = read_wav_header(wav, &metadata);
if(ret != WAV_HDR_PARSE_OK) {
log_msg(LOG_LEVEL_ERROR, "%s\n", get_wav_error(ret));
fclose(wav);
return false;
}
audio_frame->bps = audio_capture_bps ? audio_capture_bps : metadata.bits_per_sample / 8;
audio_frame->ch_count = metadata.ch_count;
audio_frame->sample_rate = metadata.sample_rate;
if (*chunk_size == 0) {
*chunk_size = audio_frame->sample_rate / CHUNKS_PER_SEC;
}
*total_samples = metadata.data_size * 8ULL / metadata.ch_count / metadata.bits_per_sample;
LOG(LOG_LEVEL_VERBOSE) << MOD_NAME << *total_samples << " samples read from file " << wav_filename << "\n";
const int headroom = (*chunk_size - 1) * audio_frame->ch_count * audio_frame->bps;
const int samples_data_size = *total_samples * audio_frame->ch_count * audio_frame->bps;
const int audio_samples_size = samples_data_size + headroom;
*audio_samples = (char *) calloc(1, audio_samples_size);
char *read_to;
char *tmp = NULL;
if ((unsigned) audio_frame->bps == metadata.bits_per_sample / 8) {
read_to = *audio_samples;
} else {
tmp = (char *) calloc(1, metadata.data_size);
read_to = tmp;
}
unsigned int samples = wav_read(read_to, *total_samples, wav, &metadata);
int bytes = samples * (metadata.bits_per_sample / 8) * metadata.ch_count;
if (samples != *total_samples) {
LOG(samples > 0 ? LOG_LEVEL_WARNING : LOG_LEVEL_ERROR) << MOD_NAME << "Warning: premature end of WAV file (" << bytes << " read, " << metadata.data_size << " expected)!\n";
if (samples == 0) {
fclose(wav);
return false;
}
*total_samples = samples;
}
fclose(wav);
if ((unsigned) audio_frame->bps != metadata.bits_per_sample / 8){
change_bps(*audio_samples, audio_frame->bps, tmp, metadata.bits_per_sample / 8, bytes);
free(tmp);
}
memcpy(*audio_samples + samples_data_size, *audio_samples, headroom);
return true;
}
static void * audio_cap_testcard_init(struct module *parent, const char *cfg)
{
UNUSED(parent);
@@ -181,7 +243,6 @@ static void * audio_cap_testcard_init(struct module *parent, const char *cfg)
double volume = DEFAULT_VOLUME;
int frequency = DEFAULT_FREQUENCY;
int chunk_size = 0;
enum {
SINE,
EBU,
@@ -224,7 +285,7 @@ static void * audio_cap_testcard_init(struct module *parent, const char *cfg)
wav_file = item + strlen("file=");
pattern = WAV;
} else if(strncasecmp(item, "frames=", strlen("frames=")) == 0) {
chunk_size = atoi(item + strlen("frames="));
s->chunk_size = atoi(item + strlen("frames="));
} else if (strncasecmp(item, "frequency=", strlen("frequency=")) == 0) {
frequency = atoi(item + strlen("frequency="));
} else if(strstr(item, "crescendo") == item) {
@@ -255,12 +316,20 @@ static void * audio_cap_testcard_init(struct module *parent, const char *cfg)
return nullptr;
}
if (pattern != WAV) {
if (pattern == WAV) {
if (!audio_testcard_read_wav(wav_file.c_str(), &s->audio,
&s->audio_samples, &s->total_samples, &s->chunk_size)) {
delete s;
return NULL;
}
} else {
s->audio.ch_count = audio_capture_channels > 0 ? audio_capture_channels : DEFAULT_AUDIO_CAPTURE_CHANNELS;
s->audio.sample_rate = audio_capture_sample_rate ? audio_capture_sample_rate :
DEFAULT_AUDIO_SAMPLE_RATE;
s->audio.bps = audio_capture_bps ? audio_capture_bps : DEFAULT_AUDIO_BPS;
s->chunk_size = chunk_size ? chunk_size : s->audio.sample_rate / CHUNKS_PER_SEC;
if (s->chunk_size == 0) {
s->chunk_size = s->audio.sample_rate / CHUNKS_PER_SEC;
}
assert(s->chunk_size > 0);
switch (pattern) {
case CRESCENDO:
@@ -291,63 +360,6 @@ static void * audio_cap_testcard_init(struct module *parent, const char *cfg)
s->audio.ch_count * s->audio.bps) + s->chunk_size - 1);
memcpy(s->audio_samples + s->total_samples * s->audio.bps * s->audio.ch_count,
s->audio_samples, s->chunk_size - 1);
} else {
FILE *wav = fopen(wav_file.c_str(), "r");
if(!wav) {
LOG(LOG_LEVEL_ERROR) << MOD_NAME << "Unable to open WAV file: " << wav_file << ".\n";
delete s;
return NULL;
}
struct wav_metadata metadata;
int ret = read_wav_header(wav, &metadata);
if(ret != WAV_HDR_PARSE_OK) {
log_msg(LOG_LEVEL_ERROR, "%s\n", get_wav_error(ret));
fclose(wav);
delete s;
return NULL;
}
s->audio.bps = audio_capture_bps ? audio_capture_bps : metadata.bits_per_sample / 8;
s->audio.ch_count = metadata.ch_count;
s->audio.sample_rate = metadata.sample_rate;
s->chunk_size = chunk_size ? chunk_size : s->audio.sample_rate / CHUNKS_PER_SEC;
s->total_samples = metadata.data_size * 8ULL / metadata.ch_count / metadata.bits_per_sample;
LOG(LOG_LEVEL_VERBOSE) << MOD_NAME << s->total_samples << " samples read from file " << wav_file << "\n";
const int headroom = (s->chunk_size - 1) * s->audio.ch_count * s->audio.bps;
const int samples_data_size = s->total_samples * s->audio.ch_count * s->audio.bps;
const int audio_samples_size = samples_data_size + headroom;
s->audio_samples = (char *) calloc(1, audio_samples_size);
char *read_to;
char *tmp = NULL;
if ((unsigned) s->audio.bps == metadata.bits_per_sample / 8) {
read_to = s->audio_samples;
} else {
tmp = (char *) calloc(1, metadata.data_size);
read_to = tmp;
}
unsigned int samples = wav_read(read_to, s->total_samples, wav, &metadata);
int bytes = samples * (metadata.bits_per_sample / 8) * metadata.ch_count;
if (samples != s->total_samples) {
LOG(samples > 0 ? LOG_LEVEL_WARNING : LOG_LEVEL_ERROR) << MOD_NAME << "Warning: premature end of WAV file (" << bytes << " read, " << metadata.data_size << " expected)!\n";
if (samples == 0) {
fclose(wav);
delete s;
return NULL;
}
s->total_samples = samples;
}
fclose(wav);
if ((unsigned) s->audio.bps != metadata.bits_per_sample / 8){
change_bps(s->audio_samples, s->audio.bps, tmp, metadata.bits_per_sample / 8, bytes);
free(tmp);
}
memcpy(s->audio_samples + samples_data_size, s->audio_samples, headroom);
}
s->audio.data_len = s->chunk_size * s->audio.bps * s->audio.ch_count;