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More spelling fixes

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This commit is contained in:
Maarten Bosmans 2011-08-24 18:24:46 +02:00 • committed by Colin Guthrie
parent 231645d4fc
commit c5dca7cf2b
81 changed files with 165 additions and 167 deletions
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4
src/modules/alsa/alsa-mixer.c
View file

@ -3826,7 +3826,7 @@ static int profile_verify(pa_alsa_profile *p) {
continue;
if (!(m = pa_hashmap_get(p->profile_set->mappings, *name)) || m->direction == PA_ALSA_DIRECTION_INPUT) {
pa_log("Profile '%s' refers to unexistant mapping '%s'.", p->name, *name);
pa_log("Profile '%s' refers to nonexistent mapping '%s'.", p->name, *name);
return -1;
}
@ -3862,7 +3862,7 @@ static int profile_verify(pa_alsa_profile *p) {
continue;
if (!(m = pa_hashmap_get(p->profile_set->mappings, *name)) || m->direction == PA_ALSA_DIRECTION_OUTPUT) {
pa_log("Profile '%s' refers to unexistant mapping '%s'.", p->name, *name);
pa_log("Profile '%s' refers to nonexistent mapping '%s'.", p->name, *name);
return -1;
}

2
src/modules/alsa/alsa-mixer.h
View file

@ -277,7 +277,7 @@ struct pa_alsa_decibel_fix {
long max_step;
/* An array that maps alsa volume element steps to decibels. The steps can
* be used as indices to this array, after substracting min_step from the
* be used as indices to this array, after subtracting min_step from the
* real value.
*
* The values are actually stored as integers representing millibels,

8
src/modules/alsa/alsa-sink.c
View file

@ -69,10 +69,10 @@
#define TSCHED_WATERMARK_INC_STEP_USEC (10*PA_USEC_PER_MSEC) /* 10ms -- On underrun, increase watermark by this */
#define TSCHED_WATERMARK_DEC_STEP_USEC (5*PA_USEC_PER_MSEC) /* 5ms -- When everything's great, decrease watermark by this */
#define TSCHED_WATERMARK_VERIFY_AFTER_USEC (20*PA_USEC_PER_SEC) /* 20s -- How long after a drop out recheck if things are good now */
#define TSCHED_WATERMARK_INC_THRESHOLD_USEC (0*PA_USEC_PER_MSEC) /* 0ms -- If the buffer level ever below this theshold, increase the watermark */
#define TSCHED_WATERMARK_DEC_THRESHOLD_USEC (100*PA_USEC_PER_MSEC) /* 100ms -- If the buffer level didn't drop below this theshold in the verification time, decrease the watermark */
#define TSCHED_WATERMARK_INC_THRESHOLD_USEC (0*PA_USEC_PER_MSEC) /* 0ms -- If the buffer level ever below this threshold, increase the watermark */
#define TSCHED_WATERMARK_DEC_THRESHOLD_USEC (100*PA_USEC_PER_MSEC) /* 100ms -- If the buffer level didn't drop below this threshold in the verification time, decrease the watermark */
/* Note that TSCHED_WATERMARK_INC_THRESHOLD_USEC == 0 means tht we
/* Note that TSCHED_WATERMARK_INC_THRESHOLD_USEC == 0 means that we
* will increase the watermark only if we hit a real underrun. */
#define TSCHED_MIN_SLEEP_USEC (10*PA_USEC_PER_MSEC) /* 10ms -- Sleep at least 10ms on each iteration */
@ -929,7 +929,7 @@ static int update_sw_params(struct userdata *u) {
pa_assert(u);
/* Use the full buffer if noone asked us for anything specific */
/* Use the full buffer if no one asked us for anything specific */
u->hwbuf_unused = 0;
if (u->use_tsched) {

2
src/modules/alsa/alsa-source.c
View file

@ -850,7 +850,7 @@ static int update_sw_params(struct userdata *u) {
pa_assert(u);
/* Use the full buffer if noone asked us for anything specific */
/* Use the full buffer if no one asked us for anything specific */
u->hwbuf_unused = 0;
if (u->use_tsched) {

14
src/modules/echo-cancel/module-echo-cancel.c
View file

@ -55,7 +55,7 @@
#include "module-echo-cancel-symdef.h"
PA_MODULE_AUTHOR("Wim Taymans");
PA_MODULE_DESCRIPTION("Echo Cancelation");
PA_MODULE_DESCRIPTION("Echo Cancellation");
PA_MODULE_VERSION(PACKAGE_VERSION);
PA_MODULE_LOAD_ONCE(FALSE);
PA_MODULE_USAGE(
@ -132,10 +132,10 @@ static const pa_echo_canceller ec_table[] = {
*
* Alignment is performed in two steps:
*
* 1) when something happens that requires quick adjustement of the alignment of
* 1) when something happens that requires quick adjustment of the alignment of
* capture and playback samples, we perform a resync. This adjusts the
* position in the playback memblock to the requested sample. Quick
* adjustements include moving the playback samples before the capture
* adjustments include moving the playback samples before the capture
* samples (because else the echo canceler does not work) or when the
* playback pointer drifts too far away.
*
@ -250,7 +250,7 @@ static int64_t calc_diff(struct userdata *u, struct snapshot *snapshot) {
buffer += snapshot->source_delay + snapshot->sink_delay;
/* add the amount of samples not yet transfered to the source context */
/* add the amount of samples not yet transferred to the source context */
if (snapshot->recv_counter <= snapshot->send_counter)
buffer += (int64_t) (snapshot->send_counter - snapshot->recv_counter);
else
@ -322,7 +322,7 @@ static void time_callback(pa_mainloop_api *a, pa_time_event *e, const struct tim
new_rate = base_rate;
}
/* make sure we don't make too big adjustements because that sounds horrible */
/* make sure we don't make too big adjustments because that sounds horrible */
if (new_rate > base_rate * 1.1 || new_rate < base_rate * 0.9)
new_rate = base_rate;
@ -724,7 +724,7 @@ static void source_output_push_cb(pa_source_output *o, const pa_memchunk *chunk)
fwrite(pdata, 1, u->blocksize, u->played_file);
}
/* perform echo cancelation */
/* perform echo cancellation */
u->ec->run(u->ec, rdata, pdata, cdata);
/* preprecessor is run after AEC. This is not a mistake! */
@ -1441,7 +1441,7 @@ int pa__init(pa_module*m) {
u->ec->echo_suppress_attenuation_active = DEFAULT_ECHO_SUPPRESS_ATTENUATION;
if (pa_modargs_get_value_s32(ma, "echo_suppress_attenuation_active", &u->ec->echo_suppress_attenuation_active) < 0) {
pa_log("Failed to parse echo_supress_attenuation_active value");
pa_log("Failed to parse echo_suppress_attenuation_active value");
goto fail;
}
if (u->ec->echo_suppress_attenuation_active > 0) {

2
src/modules/jack/module-jack-sink.c
View file

@ -47,7 +47,7 @@
/* General overview:
*
* Because JACK has a very unflexible event loop management which
* Because JACK has a very inflexible event loop management which
* doesn't allow us to add our own event sources to the event thread
* we cannot use the JACK real-time thread for dispatching our PA
* work. Instead, we run an additional RT thread which does most of

4
src/modules/module-combine-sink.c
View file

@ -102,7 +102,7 @@ struct output {
/* For communication of the stream latencies to the main thread */
pa_usec_t total_latency;
/* For coomunication of the stream parameters to the sink thread */
/* For communication of the stream parameters to the sink thread */
pa_atomic_t max_request;
pa_atomic_t requested_latency;
@ -991,7 +991,7 @@ static void output_disable(struct output *o) {
* pass any further data to this output */
pa_asyncmsgq_send(o->userdata->sink->asyncmsgq, PA_MSGOBJECT(o->userdata->sink), SINK_MESSAGE_REMOVE_OUTPUT, o, 0, NULL);
/* Now dellocate the stream */
/* Now deallocate the stream */
pa_sink_input_unref(o->sink_input);
o->sink_input = NULL;

4
src/modules/module-default-device-restore.c
View file

@ -74,7 +74,7 @@ static void load(struct userdata *u) {
pa_namereg_set_default_sink(u->core, s);
pa_log_info("Restored default sink '%s'.", ln);
} else
pa_log_info("Saved default sink '%s' not existant, not restoring default sink setting.", ln);
pa_log_info("Saved default sink '%s' not existent, not restoring default sink setting.", ln);
} else if (errno != ENOENT)
pa_log("Failed to load default sink: %s", pa_cstrerror(errno));
@ -95,7 +95,7 @@ static void load(struct userdata *u) {
pa_namereg_set_default_source(u->core, s);
pa_log_info("Restored default source '%s'.", ln);
} else
pa_log_info("Saved default source '%s' not existant, not restoring default source setting.", ln);
pa_log_info("Saved default source '%s' not existent, not restoring default source setting.", ln);
} else if (errno != ENOENT)
pa_log("Failed to load default sink: %s", pa_cstrerror(errno));

8
src/modules/module-device-manager.c
View file

@ -1334,7 +1334,7 @@ static int extension_cb(pa_native_protocol *p, pa_module *m, pa_native_connectio
goto fail;
}
/* Add the device to our hashmap. If it's alredy in it, free it now and carry on */
/* Add the device to our hashmap. If it's already in it, free it now and carry on */
device = pa_xnew(struct device_t, 1);
device->device = pa_xstrdup(s);
if (pa_hashmap_put(h, device->device, device) == 0) {
@ -1352,7 +1352,7 @@ static int extension_cb(pa_native_protocol *p, pa_module *m, pa_native_connectio
}*/
/* Now cycle through our list and add all the devices.
This has the effect of addign in any in our DB,
This has the effect of adding in any in our DB,
not specified in the device list (and thus will be
tacked on at the end) */
offset = idx;
@ -1368,10 +1368,10 @@ static int extension_cb(pa_native_protocol *p, pa_module *m, pa_native_connectio
if ((sink_mode && 0 == strncmp("sink:", device->device, 5))
|| (!sink_mode && 0 == strncmp("source:", device->device, 7))) {
/* Add the device to our hashmap. If it's alredy in it, free it now and carry on */
/* Add the device to our hashmap. If it's already in it, free it now and carry on */
if (pa_hashmap_put(h, device->device, device) == 0
&& (e = entry_read(u, device->device))) {
/* We add offset on to the existing priorirty so that when we order, the
/* We add offset on to the existing priority so that when we order, the
existing entries are always lower priority than the new ones. */
device->prio = (offset + e->priority[role_index]);
pa_xfree(e);

2
src/modules/module-device-restore.c
View file

@ -99,7 +99,7 @@ struct userdata {
pa_bool_t restore_formats:1;
};
/* Protocol extention commands */
/* Protocol extension commands */
enum {
SUBCOMMAND_TEST,
SUBCOMMAND_SUBSCRIBE,

22
src/modules/module-equalizer-sink.c
View file

@ -114,7 +114,7 @@ struct userdata {
//size_t samplings;
float **Xs;
float ***Hs;//thread updatable copies of the freq response filters (magintude based)
float ***Hs;//thread updatable copies of the freq response filters (magnitude based)
pa_aupdate **a_H;
pa_memblockq *input_q;
char *output_buffer;
@ -356,10 +356,10 @@ static void dsp_logic(
*automatically cycled in routine
*/
float * restrict overlap,
const float X,//multipliar
const float X,//multiplier
const float * restrict H,//The freq. magnitude scalers filter
const float * restrict W,//The windowing function
fftwf_complex * restrict output_window,//The transformed window'd src
fftwf_complex * restrict output_window,//The transformed windowed src
struct userdata *u){
//use a linear-phase sliding STFT and overlap-add method (for each channel)
@ -367,7 +367,7 @@ static void dsp_logic(
for(size_t j = 0; j < u->window_size; ++j){
dst[j] = X * W[j] * src[j];
}
//zero padd the the remaining fft window
//zero pad the the remaining fft window
memset(dst + u->window_size, 0, (u->fft_size - u->window_size) * sizeof(float));
//Processing is done here!
//do fft
@ -379,7 +379,7 @@ static void dsp_logic(
}
//inverse fft
fftwf_execute_dft_c2r(u->inverse_plan, output_window, dst);
////debug: tests overlaping add
////debug: tests overlapping add
////and negates ALL PREVIOUS processing
////yields a perfect reconstruction if COLA is held
//for(size_t j = 0; j < u->window_size; ++j){
@ -397,7 +397,7 @@ static void dsp_logic(
// u->work_buffer[j] = u->input[c][j];
//}
//preseve the needed input for the next window's overlap
//preserve the needed input for the next window's overlap
memmove(src, src + u->R,
(u->samples_gathered - u->R) * sizeof(float)
);
@ -418,10 +418,10 @@ static void dsp_logic(
*automatically cycled in routine
*/
float * restrict overlap,//The size of the overlap
const float X,//multipliar
const float X,//multiplier
const float * restrict H,//The freq. magnitude scalers filter
const float * restrict W,//The windowing function
fftwf_complex * restrict output_window,//The transformed window'd src
fftwf_complex * restrict output_window,//The transformed windowed src
struct userdata *u){//Collection of constants
const size_t overlap_size = PA_ROUND_UP(u->overlap_size, v_size);
float_vector_t x;
@ -439,7 +439,7 @@ static void dsp_logic(
// d->v = x->v * w->v * s->v;
//#endif
}
//zero padd the the remaining fft window
//zero pad the the remaining fft window
memset(dst + u->window_size, 0, (u->fft_size - u->window_size) * sizeof(float));
//Processing is done here!
@ -463,7 +463,7 @@ static void dsp_logic(
//inverse fft
fftwf_execute_dft_c2r(u->inverse_plan, output_window, dst);
////debug: tests overlaping add
////debug: tests overlapping add
////and negates ALL PREVIOUS processing
////yields a perfect reconstruction if COLA is held
//for(size_t j = 0; j < u->window_size; ++j){
@ -494,7 +494,7 @@ static void dsp_logic(
// dst[j] = src[j];
//}
//preseve the needed input for the next window's overlap
//preserve the needed input for the next window's overlap
memmove(src, src + u->R,
(u->samples_gathered - u->R) * sizeof(float)
);

2
src/modules/module-hal-detect.c
View file

@ -206,7 +206,7 @@ static int hal_device_load_alsa(struct userdata *u, const char *udi, struct devi
/* For each ALSA card that appears the control device will be the
* last one to be created, this is considered part of the ALSA
* usperspace API. We rely on this and load our modules only when
* userspace API. We rely on this and load our modules only when
* the control device is available assuming that *all* device
* nodes have been properly created and assigned the right ACLs at
* that time. Also see:

2
src/modules/module-ladspa-sink.c
View file

@ -58,7 +58,7 @@ PA_MODULE_USAGE(
"channel_map=<input channel map> "
"plugin=<ladspa plugin name> "
"label=<ladspa plugin label> "
"control=<comma seperated list of input control values> "
"control=<comma separated list of input control values> "
"input_ladspaport_map=<comma separated list of input LADSPA port names> "
"output_ladspaport_map=<comma separated list of output LADSPA port names> "));

2
src/modules/module-position-event-sounds.c
View file

@ -87,7 +87,7 @@ static pa_hook_result_t sink_input_fixate_hook_callback(pa_core *core, pa_sink_i
if ((id = pa_proplist_gets(data->proplist, PA_PROP_EVENT_ID))) {
/* The test sounds should never be positioned in space, since
* they might be trigered themselves to configure the speakers
* they might be triggered themselves to configure the speakers
* in space, which we don't want to mess up. */
if (pa_startswith(id, "audio-channel-"))

4
src/modules/module-protocol-stub.c
View file

@ -133,7 +133,7 @@
PA_MODULE_DESCRIPTION("Native protocol "SOCKET_DESCRIPTION);
PA_MODULE_USAGE("auth-anonymous=<don't check for cookies?> "
"auth-cookie=<path to cookie file> "
"auth-cookie-enabled=<enable cookie authentification?> "
"auth-cookie-enabled=<enable cookie authentication?> "
AUTH_USAGE
SOCKET_USAGE);
#elif defined(USE_PROTOCOL_ESOUND)
@ -162,7 +162,7 @@
"source=<source to connect to> "
"auth-anonymous=<don't verify cookies?> "
"auth-cookie=<path to cookie file> "
"auth-cookie-enabled=<enable cookie authentification?> "
"auth-cookie-enabled=<enable cookie authentication?> "
AUTH_USAGE
SOCKET_USAGE);
#else

2
src/modules/module-tunnel.c
View file

@ -485,7 +485,7 @@ static int sink_process_msg(pa_msgobject *o, int code, void *data, int64_t offse
case PA_SINK_MESSAGE_SET_STATE: {
int r;
/* First, change the state, because otherwide pa_sink_render() would fail */
/* First, change the state, because otherwise pa_sink_render() would fail */
if ((r = pa_sink_process_msg(o, code, data, offset, chunk)) >= 0) {
stream_cork_within_thread(u, u->sink->state == PA_SINK_SUSPENDED);

6
src/modules/module-udev-detect.c
View file

@ -46,7 +46,7 @@ PA_MODULE_LOAD_ONCE(TRUE);
PA_MODULE_USAGE(
"tsched=<enable system timer based scheduling mode?> "
"ignore_dB=<ignore dB information from the device?> "
"sync_volume=<syncronize sw and hw voluchanges in IO-thread?>");
"sync_volume=<syncronize sw and hw volume changes in IO-thread?>");
struct device {
char *path;
@ -318,7 +318,7 @@ static void verify_access(struct userdata *u, struct device *d) {
* A clean fix would be if we would be able to ignore
* our own inotify close events. However, inotify
* lacks such functionality. Also, during probing of
* the device we cannot really distuingish between
* the device we cannot really distinguish between
* other processes causing EBUSY or ourselves, which
* means we have no way to figure out if the probing
* during opening was canceled by a "try again"
@ -721,7 +721,7 @@ int pa__init(pa_module *m) {
pa_log("Failed to enable monitor: %s", pa_cstrerror(errno));
if (errno == EPERM)
pa_log_info("Most likely your kernel is simply too old and "
"allows only priviliged processes to listen to device events. "
"allows only privileged processes to listen to device events. "
"Please upgrade your kernel to at least 2.6.30.");
goto fail;
}

2
src/modules/oss/module-oss.c
View file

@ -27,7 +27,7 @@
* We make no difference between IDLE and RUNNING in our handling.
*
* As long as we are in RUNNING/IDLE state we will *always* write data to
* the device. If none is avilable from the inputs, we write silence
* the device. If none is available from the inputs, we write silence
* instead.
*
* If power should be saved on IDLE module-suspend-on-idle should be used.

4
src/modules/raop/module-raop-sink.c
View file

@ -248,7 +248,7 @@ static int sink_process_msg(pa_msgobject *o, int code, void *data, int64_t offse
pa_rtpoll_item_free(u->rtpoll_item);
u->rtpoll_item = NULL;
} else {
/* Quesiton: is this valid here: or should we do some sort of:
/* Question: is this valid here: or should we do some sort of:
return pa_sink_process_msg(PA_MSGOBJECT(u->core), PA_CORE_MESSAGE_UNLOAD_MODULE, u->module, 0, NULL);
?? */
pa_module_unload_request(u->module, TRUE);
@ -287,7 +287,7 @@ static void sink_set_volume_cb(pa_sink *s) {
pa_log_debug("Got hardware volume: %s", pa_cvolume_snprint(t, sizeof(t), &hw));
pa_log_debug("Calculated software volume: %s", pa_cvolume_snprint(t, sizeof(t), &s->soft_volume));
/* Any necessary software volume manipulateion is done so set
/* Any necessary software volume manipulation is done so set
our hw volume (or v as a single value) on the device */
pa_raop_client_set_volume(u->raop, v);
}

6
src/modules/raop/raop_client.c
View file

@ -110,7 +110,7 @@ static inline void bit_writer(uint8_t **buffer, uint8_t *bit_pos, int *size, uin
if (!data_bit_len)
return;
/* If bit pos is zero, we will definatly use at least one bit from the current byte so size increments. */
/* If bit pos is zero, we will definately use at least one bit from the current byte so size increments. */
if (!*bit_pos)
*size += 1;
@ -128,7 +128,7 @@ static inline void bit_writer(uint8_t **buffer, uint8_t *bit_pos, int *size, uin
**buffer = bit_data;
/* If our data fits exactly into the current byte, we need to increment our pointer */
if (0 == bit_overflow) {
/* Do not increment size as it will be incremeneted on next call as bit_pos is zero */
/* Do not increment size as it will be incremented on next call as bit_pos is zero */
*buffer += 1;
*bit_pos = 0;
} else {
@ -523,7 +523,7 @@ int pa_raop_client_encode_sample(pa_raop_client* c, pa_memchunk* raw, pa_memchun
pa_memblock_release(raw->memblock);
encoded->length = header_size + size;
/* store the lenght (endian swapped: make this better) */
/* store the length (endian swapped: make this better) */
len = size + header_size - 4;
*(b + 2) = len >> 8;
*(b + 3) = len & 0xff;

2
src/modules/reserve-monitor.h
View file

@ -45,7 +45,7 @@ typedef void (*rm_change_cb_t)(rm_monitor *m);
* DBus error might be set as well if the error was caused D-Bus. */
int rm_watch(
rm_monitor **m, /* On success a pointer to the newly allocated rm_device object will be filled in here */
DBusConnection *connection, /* Session bus (when D-Bus learns about user busses we should switchg to user busses) */
DBusConnection *connection, /* Session bus (when D-Bus learns about user busses we should switch to user busses) */
const char *device_name, /* The device to monitor, e.g. "Audio0" */
rm_change_cb_t change_cb, /* Will be called whenever the lock status changes. May be NULL */
DBusError *error); /* If we fail due to a D-Bus related issue the error will be filled in here. May be NULL. */

2
src/modules/reserve.h
View file

@ -51,7 +51,7 @@ typedef int (*rd_request_cb_t)(
* the error was caused D-Bus. */
int rd_acquire(
rd_device **d, /* On success a pointer to the newly allocated rd_device object will be filled in here */
DBusConnection *connection, /* Session bus (when D-Bus learns about user busses we should switchg to user busses) */
DBusConnection *connection, /* Session bus (when D-Bus learns about user busses we should switch to user busses) */
const char *device_name, /* The device to lock, e.g. "Audio0" */
const char *application_name, /* A human readable name of the application, e.g. "PulseAudio Sound Server" */
int32_t priority, /* The priority for this application. If unsure use 0 */

2
src/modules/udev-util.c
View file

@ -244,7 +244,7 @@ int pa_udev_get_info(int card_idx, pa_proplist *p) {
if ((v = udev_device_get_property_value(card, "SOUND_FORM_FACTOR")) && *v)
pa_proplist_sets(p, PA_PROP_DEVICE_FORM_FACTOR, v);
/* This is normaly not set by the udev rules but may be useful to
/* This is normally not set by the udev rules but may be useful to
* allow administrators to overwrite the device description.*/
if (!pa_proplist_contains(p, PA_PROP_DEVICE_DESCRIPTION))
if ((v = udev_device_get_property_value(card, "SOUND_DESCRIPTION")) && *v)