Jack output add soxr resampling
This commit is contained in:
committed by
Pieter De Gendt
parent
b1a5056494
commit
7285b1ee03
+137
-25
@@ -29,10 +29,20 @@
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#include <jack/jack.h>
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#include <jack/ringbuffer.h>
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// Two-channel, 16bit audio:
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static const int bytes_per_frame = 4;
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#ifdef CONFIG_SOXR
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#include <soxr.h>
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#endif
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#define NPORTS 2
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typedef jack_default_audio_sample_t sample_t;
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#define jack_sample_size sizeof(sample_t)
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// Two-channel, 32bit audio:
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static const int bytes_per_frame = NPORTS * jack_sample_size;
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// Four seconds buffer -- should be plenty
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#define buffer_size (44100 * 4 * bytes_per_frame)
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#define buffer_size (48000u * 4u * bytes_per_frame)
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static pthread_mutex_t buffer_mutex = PTHREAD_MUTEX_INITIALIZER;
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static pthread_mutex_t client_mutex = PTHREAD_MUTEX_INITIALIZER;
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@@ -62,7 +72,6 @@ audio_output audio_jack = {.name = "jack",
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// This also affects deinterlacing.
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// So make it exactly the number of incoming audio channels!
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#define NPORTS 2
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static jack_port_t *port[NPORTS];
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static const char *port_name[NPORTS] = {"out_L", "out_R"};
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@@ -76,7 +85,36 @@ static int flush_please = 0;
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static jack_latency_range_t latest_latency_range[NPORTS];
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static int64_t time_of_latest_transfer;
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static inline jack_default_audio_sample_t sample_conv(short sample) {
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#ifdef CONFIG_SOXR
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typedef struct soxr_recipe {
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int recipe;
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const char *name;
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} soxr_recipe_t;
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static soxr_recipe_t soxr_quality_table[] = {
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{ SOXR_VHQ, "very high" },
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{ SOXR_HQ, "high" },
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{ SOXR_MQ, "medium" },
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{ SOXR_LQ, "low" },
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{ SOXR_QQ, "quick" },
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{ -1, NULL }
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};
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static int parse_soxr_recipe_name(const char *name) {
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for (soxr_recipe_t *s = soxr_quality_table; s->name != NULL; ++s) {
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if (!strcmp(s->name, name)) {
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return s->recipe;
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}
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}
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return -1;
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}
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static soxr_t soxr = NULL;
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static soxr_quality_spec_t quality_spec;
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static soxr_io_spec_t io_spec;
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#endif
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static inline sample_t sample_conv(short sample) {
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// It sounds correct, but I don't understand it.
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// Zero int needs to be zero float. Check.
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// Plus 32767 int is 1.0. Check.
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@@ -86,17 +124,17 @@ static inline jack_default_audio_sample_t sample_conv(short sample) {
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return ((sample < 0) ? (-1.0 * sample / SHRT_MIN) : (1.0 * sample / SHRT_MAX));
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}
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static void deinterleave_and_convert(const char *interleaved_input_buffer,
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jack_default_audio_sample_t *jack_output_buffer[],
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jack_nframes_t offset, jack_nframes_t nframes) {
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static void deinterleave(const char *interleaved_input_buffer,
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sample_t *jack_output_buffer[],
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jack_nframes_t offset, jack_nframes_t nframes) {
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jack_nframes_t f;
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// We're dealing with 16bit audio here:
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short *ifp = (short *)interleaved_input_buffer;
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sample_t *ifp = (sample_t *)interleaved_input_buffer;
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// Zero-copy, we're working directly on the target and destination buffers,
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// so deal with an offset for the second part of the input ringbuffer
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for (f = offset; f < (nframes + offset); f++) {
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for (int i = 0; i < NPORTS; i++) {
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jack_output_buffer[i][f] = sample_conv(*ifp++);
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jack_output_buffer[i][f] = *ifp++;
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}
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}
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}
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@@ -107,7 +145,7 @@ static void deinterleave_and_convert(const char *interleaved_input_buffer,
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// output, no file access, no mutexes...
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// The JACK ringbuffer we use to get the data in here is explicitly lock-free.
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static int process(jack_nframes_t nframes, __attribute__((unused)) void *arg) {
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jack_default_audio_sample_t *buffer[NPORTS];
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sample_t *buffer[NPORTS];
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// Expect an array of two elements because of possible ringbuffer wrap-around:
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jack_ringbuffer_data_t v[2] = {0};
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jack_nframes_t i, thisbuf;
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@@ -115,7 +153,7 @@ static int process(jack_nframes_t nframes, __attribute__((unused)) void *arg) {
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int frames_required = 0;
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for (i = 0; i < NPORTS; i++) {
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buffer[i] = (jack_default_audio_sample_t *)jack_port_get_buffer(port[i], nframes);
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buffer[i] = (sample_t *)jack_port_get_buffer(port[i], nframes);
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}
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if (flush_please) {
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// We just move the read pointer ahead without doing anything with the data.
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@@ -131,7 +169,7 @@ static int process(jack_nframes_t nframes, __attribute__((unused)) void *arg) {
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} else {
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frames_required = thisbuf;
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}
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deinterleave_and_convert(v[i].buf, buffer, frames_written, frames_required);
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deinterleave(v[i].buf, buffer, frames_written, frames_required);
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frames_written += frames_required;
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nframes -= frames_required;
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}
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@@ -183,6 +221,9 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a
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// Do the "general" audio options. Note, these options are in the "general" stanza!
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parse_general_audio_options();
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#ifdef CONFIG_SOXR
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config.jack_soxr_resample_recipe = -1; // don't resample by default
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#endif
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// Now the options specific to the backend, from the "jack" stanza:
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if (config.cfg != NULL) {
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@@ -193,6 +234,12 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a
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if (config_lookup_string(config.cfg, "jack.autoconnect_pattern", &str)) {
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config.jack_autoconnect_pattern = (char *)str;
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}
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#ifdef CONFIG_SOXR
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if (config_lookup_string(config.cfg, "jack.soxr_resample_recipe", &str)) {
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debug(1, "SOXR quality %s", str);
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config.jack_soxr_resample_recipe = parse_soxr_recipe_name(str);
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}
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#endif
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}
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if (config.jack_client_name == NULL)
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config.jack_client_name = strdup("shairport-sync");
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@@ -213,6 +260,12 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a
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die("Could not start JACK server. JackStatus is %x", status);
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}
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sample_rate = jack_get_sample_rate(client);
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#ifdef CONFIG_SOXR
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if (config.jack_soxr_resample_recipe >= SOXR_QQ) {
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quality_spec = soxr_quality_spec(config.jack_soxr_resample_recipe, 0);
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io_spec = soxr_io_spec(SOXR_INT16_I, SOXR_FLOAT32_I);
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} else
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#endif
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if (sample_rate != 44100) {
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die("The JACK server is running at the wrong sample rate (%d) for Shairport Sync."
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" Must be 44100 Hz.",
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@@ -287,14 +340,38 @@ void jack_deinit() {
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warn("Error closing jack client");
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pthread_mutex_unlock(&client_mutex);
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jack_ringbuffer_free(jackbuf);
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#ifdef CONFIG_SOXR
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if (soxr) {
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soxr_delete(soxr);
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soxr = NULL;
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}
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#endif
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}
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void jack_start(__attribute__((unused)) int i_sample_rate,
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void jack_start(int i_sample_rate,
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__attribute__((unused)) int i_sample_format) {
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// Nothing to do, JACK client has already been set up at jack_init().
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// Also, we have no say over the sample rate or sample format of JACK,
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// We convert the 16bit samples to float, and die if the sample rate is != 44k1.
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// FIXME: later, resampling would be nice. Fold into soxr if possible.
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// We convert the 16bit samples to float, and die if the sample rate is != 44k1 without soxr.
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#ifdef CONFIG_SOXR
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if (config.jack_soxr_resample_recipe >= SOXR_QQ) {
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// we might improve a bit with soxr_clear if the sample_rate doesn't change
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if (soxr) {
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soxr_delete(soxr);
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}
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soxr_error_t e = NULL;
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soxr = soxr_create(i_sample_rate,
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sample_rate,
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NPORTS,
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&e,
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&io_spec,
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&quality_spec,
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NULL);
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if (!soxr) {
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die("Unable to create soxr resampler for JACK: %s", e);
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}
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}
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#endif
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}
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void jack_flush() {
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@@ -319,7 +396,7 @@ int jack_delay(long *the_delay) {
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debug(2, "audio_occupancy_now is %d.", audio_occupancy_now);
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pthread_mutex_unlock(&buffer_mutex);
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int64_t frames_processed_since_latest_latency_check = (delta * 44100) >> 32;
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int64_t frames_processed_since_latest_latency_check = (delta * sample_rate) >> 32;
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// debug(1,"delta: %" PRId64 " frames.",frames_processed_since_latest_latency_check);
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// jack_latency is set by the graph() callback, it's the average of the maximum
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// latencies of all our output ports. Adjust this constant baseline delay according
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@@ -330,18 +407,53 @@ int jack_delay(long *the_delay) {
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}
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int play(void *buf, int samples) {
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// debug(1,"jack_play of %d samples.",samples);
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// copy the samples into the queue
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size_t bytes_to_transfer, bytes_transferred;
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bytes_to_transfer = samples * bytes_per_frame;
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jack_ringbuffer_data_t v[2] = {0};
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size_t i, j, c;
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jack_nframes_t thisbuf;
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// It's ok to lock here since we're not in the realtime callback:
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pthread_mutex_lock(&buffer_mutex);
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bytes_transferred = jack_ringbuffer_write(jackbuf, buf, bytes_to_transfer);
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jack_ringbuffer_get_write_vector(jackbuf, v);
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short *in = (short *)buf;
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sample_t *out;
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for (i = 0; i < 2; ++i) {
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thisbuf = v[i].len / (jack_sample_size * NPORTS); // #samples per channel
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out = (sample_t *)v[i].buf;
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#ifdef CONFIG_SOXR
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if (soxr) {
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size_t i_done, o_done;
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soxr_error_t e;
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while (samples > 0 && thisbuf > 0) {
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e = soxr_process(soxr,
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(soxr_in_t)in,
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samples,
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&i_done,
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(soxr_out_t)out,
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thisbuf,
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&o_done);
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if (e)
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die("Error during soxr process: %s", e);
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samples -= i_done;
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thisbuf -= o_done;
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jack_ringbuffer_write_advance(jackbuf, o_done * jack_sample_size * NPORTS);
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}
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} else {
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#endif
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j = 0;
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for (j = 0; j < thisbuf && samples > 0; ++j) {
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for (c = 0; c < NPORTS; ++c)
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out[j * NPORTS + c] = sample_conv(*in++);
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--samples;
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}
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jack_ringbuffer_write_advance(jackbuf, j * jack_sample_size * NPORTS);
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#ifdef CONFIG_SOXR
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}
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#endif
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}
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time_of_latest_transfer = get_absolute_time_in_fp();
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pthread_mutex_unlock(&buffer_mutex);
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if (bytes_transferred < bytes_to_transfer) {
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warn("JACK ringbuffer overrun. Only wrote %d of %d bytes.", bytes_transferred,
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bytes_to_transfer);
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if (samples) {
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warn("JACK ringbuffer overrun. Dropped %d samples.", samples);
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}
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return 0;
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}
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