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NOP: remove commented-out code segments that were re-implemented

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Jörn Nettingsmeier
2019-02-15 17:58:50 +00:00
parent 23d4ec2bee
commit 444c114231
1 changed files with 0 additions and 127 deletions
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audio_jack.c
-127
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@@ -106,49 +106,9 @@ int play(void *buf, int samples) {
if (bytes_transferred < bytes_to_transfer) {
debug(1, "JACK ringbuffer overrun. Only wrote %d of %d bytes.", bytes_transferred, bytes_to_transfer);
}
/*
size_t space_to_end_of_buffer = audio_umb - audio_eoq;
if (space_to_end_of_buffer >= bytes_to_transfer) {
memcpy(audio_eoq, buf, bytes_to_transfer);
pthread_mutex_lock(&buffer_mutex);
audio_occupancy += samples;
audio_eoq += bytes_to_transfer;
pthread_mutex_unlock(&buffer_mutex);
} else {
memcpy(audio_eoq, buf, space_to_end_of_buffer);
buf += space_to_end_of_buffer;
memcpy(audio_lmb, buf, bytes_to_transfer - space_to_end_of_buffer);
pthread_mutex_lock(&buffer_mutex);
audio_occupancy += samples;
audio_eoq = audio_lmb + bytes_to_transfer - space_to_end_of_buffer;
pthread_mutex_unlock(&buffer_mutex);
}
*/
return 0;
}
void deinterleave_and_convert_stream(const char *interleaved_frames,
const jack_default_audio_sample_t *jack_frame_buffer,
jack_nframes_t number_of_frames, enum ift_type side) {
jack_nframes_t i;
short *ifp = (short *)interleaved_frames;
jack_default_audio_sample_t *fp = (jack_default_audio_sample_t *)jack_frame_buffer;
if (side == IFT_frame_right_sample)
ifp++;
for (i = 0; i < number_of_frames; i++) {
short sample = *ifp;
jack_default_audio_sample_t converted_value;
if (sample >= 0)
converted_value = (1.0 * sample) / SHRT_MAX;
else
converted_value = -(1.0 * sample) / SHRT_MIN;
*fp = converted_value;
ifp++;
ifp++;
fp++;
}
}
static inline jack_default_audio_sample_t sample_conv(short sample) {
return ((sample < 0) ? (-1.0 * sample / SHRT_MIN) : (1.0 * sample / SHRT_MAX));
}
@@ -212,50 +172,6 @@ int jack_stream_write_cb(jack_nframes_t nframes, __attribute__((unused)) void *a
frames_written++;
nframes--;
}
/*
size_t frames_we_can_transfer = nframes;
// lock
pthread_mutex_lock(&buffer_mutex);
if (audio_occupancy < frames_we_can_transfer) {
frames_we_can_transfer = audio_occupancy;
// This means we effectively have underflow from the Shairport Sync source.
// In fact, it may be that there is nothing at all coming from the source,
// but the Shairport Sync client is open and active, so it must continue to output something.
}
if (frames_we_can_transfer * 2 * 2 <= (size_t)(audio_umb - audio_toq)) {
// the bytes are all in a row in the audio buffer
deinterleave_and_convert_stream(audio_toq, &left_buffer[0], frames_we_can_transfer,
IFT_frame_left_sample);
deinterleave_and_convert_stream(audio_toq, &right_buffer[0], frames_we_can_transfer,
IFT_frame_right_sample);
audio_toq += frames_we_can_transfer * 2 * 2;
} else {
// the bytes are in two places in the audio buffer
size_t first_portion_to_write = (audio_umb - audio_toq) / (2 * 2);
if (first_portion_to_write != 0) {
deinterleave_and_convert_stream(audio_toq, &left_buffer[0], first_portion_to_write,
IFT_frame_left_sample);
deinterleave_and_convert_stream(audio_toq, &right_buffer[0], first_portion_to_write,
IFT_frame_right_sample);
}
deinterleave_and_convert_stream(audio_lmb, &left_buffer[first_portion_to_write],
frames_we_can_transfer - first_portion_to_write,
IFT_frame_left_sample);
deinterleave_and_convert_stream(audio_lmb, &right_buffer[first_portion_to_write],
frames_we_can_transfer - first_portion_to_write,
IFT_frame_right_sample);
audio_toq = audio_lmb + (frames_we_can_transfer - first_portion_to_write) * 2 * 2;
}
audio_occupancy -= frames_we_can_transfer;
jack_port_get_latency_range(left_port, JackPlaybackLatency, &latest_left_latency_range);
jack_port_get_latency_range(right_port, JackPlaybackLatency, &latest_right_latency_range);
time_of_latest_transfer = get_absolute_time_in_fp();
pthread_mutex_unlock(&buffer_mutex);
// unlock
*/
return 0;
}
@@ -265,26 +181,6 @@ void default_jack_error_callback(const char *desc) { debug(2, "jackd error: \"%s
void default_jack_info_callback(const char *desc) { inform("jackd information: \"%s\"", desc); }
/*
void default_jack_set_latency_callback(jack_latency_callback_mode_t mode,
__attribute__((unused)) void *arg) {
if (mode == JackPlaybackLatency) {
jack_latency_range_t left_latency_range, right_latency_range;
jack_port_get_latency_range(left_port, JackPlaybackLatency, &left_latency_range);
jack_port_get_latency_range(right_port, JackPlaybackLatency, &right_latency_range);
jack_nframes_t b_latency = (left_latency_range.min + left_latency_range.max) / 2;
if (b_latency == 0)
b_latency = (right_latency_range.min + right_latency_range.max) / 2;
// jack_latency = b_latency; // actually, we are not interested in the latency of the jack
// devices connected...
// FIXME: yes, we are.
jack_latency = 0;
debug(1, "playback latency callback: %" PRIu32 ".", jack_latency);
}
}
*/
int jack_is_running() {
int reply = -1; // meaning jack is not running
if (client_is_open) {
@@ -329,19 +225,12 @@ int jack_client_open_if_needed(void) {
sample_rate = jack_get_sample_rate(client);
// debug(1, "jackaudio sample rate = %" PRId32 ".", sample_rate);
if (sample_rate == 44100) {
// FIXME: shairport-sync has no need for a latency callback, it's a leaf node with only outputs.
// If in the future some jack video player wants to resync to shairplay-sync audio, then yes, but
// the current usage seems to suggest a misunderstanding.
// if (jack_set_latency_callback(client, default_jack_set_latency_callback, NULL) == 0) {
if (jack_activate(client)) {
debug(1, "jackaudio cannot activate client");
} else {
debug(2, "jackaudio client opened.");
client_is_open = 1;
}
// } else {
// debug(1, "jackaudio cannot set latency callback");
// }
} else {
inform(
"jackaudio is running at the wrong speed (%d) for Shairport Sync, which must be 44100",
@@ -453,16 +342,6 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a
jack_set_error_function(default_jack_error_callback);
jack_set_info_function(default_jack_info_callback);
/*
// allocate space for the audio buffer
audio_lmb = malloc(buffer_size);
if (audio_lmb == NULL)
die("Can't allocate %d bytes for jackaudio buffer.", buffer_size);
audio_toq = audio_eoq = audio_lmb;
audio_umb = audio_lmb + buffer_size;
audio_occupancy = 0; // frames
*/
client_is_open = 0;
// now, if selected, start a thread to automatically open a client when there is a server.
@@ -534,12 +413,6 @@ int jack_delay(long *the_delay) {
void jack_flush() {
debug(1, "Only the consumer can safely flush a lock-free ringbuffer. Asking the process callback to do it...");
flush_please = 1;
/*
// debug(1,"jack flush");
audio_toq = audio_eoq = audio_lmb;
audio_umb = audio_lmb + buffer_size;
audio_occupancy = 0; // frames
*/
}
void jack_stop(void) {