Merge branch 'enh-codespell' into 'master'

codespell: config, fixup of typos + added new stage lint with codespell step for it

See merge request pulseaudio/pulseaudio!787
This commit is contained in:
Yaroslav Halchenko 2025-10-01 20:56:56 +00:00
commit 00b53eb259
35 changed files with 64 additions and 46 deletions

8
.codespellrc Normal file
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@ -0,0 +1,8 @@
[codespell]
skip = .git,*.pdf,*.svg,po,rfc*.txt
# nd,te,wath - short variables used
# numer - API used
# Mattern,Stach - names
# entrie - entrie(s) string
# HDA - domain terms used
ignore-words-list = nd,te,wath,mattern,entrie,stach,hda,numer

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@ -21,6 +21,7 @@ workflow:
stages:
- container
- lint
- build
variables:
@ -119,3 +120,12 @@ build-meson:
artifacts:
paths:
- build/
lint-codespell:
stage: lint
image: $UBUNTU_IMAGE
before_script:
- apt-get update && apt-get install -y python3-pip
- pip3 install codespell
script:
- codespell

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@ -957,7 +957,7 @@ HTML_STYLESHEET =
# user-defined cascading style sheet that is included after the standard
# style sheets created by doxygen. Using this option one can overrule
# certain style aspects. This is preferred over using HTML_STYLESHEET
# since it does not replace the standard style sheet and is therefor more
# since it does not replace the standard style sheet and is therefore more
# robust against future updates. Doxygen will copy the style sheet file to
# the output directory.
@ -1235,7 +1235,7 @@ FORMULA_TRANSPARENT = YES
USE_MATHJAX = NO
# When MathJax is enabled you can set the default output format to be used for
# thA MathJax output. Supported types are HTML-CSS, NativeMML (i.e. MathML) and
# the MathJax output. Supported types are HTML-CSS, NativeMML (i.e. MathML) and
# SVG. The default value is HTML-CSS, which is slower, but has the best
# compatibility.

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@ -137,7 +137,7 @@ static struct ucm_items item[] = {
{NULL, NULL},
};
/* UCM verb info - this should eventually be part of policy manangement */
/* UCM verb info - this should eventually be part of policy management */
static struct ucm_info verb_info[] = {
{SND_USE_CASE_VERB_INACTIVE, 0},
{SND_USE_CASE_VERB_HIFI, 8000},

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@ -271,7 +271,7 @@ int pa_alsa_set_hw_params(
(id = snd_pcm_info_get_id(pcm_info))) {
/* This horrible hack makes sure we don't disable tsched on USB
* devices, which have a low enough transfer size for timer-based
* scheduling to work. This can go away when the ALSA API supprots
* scheduling to work. This can go away when the ALSA API supports
* querying the block transfer size. */
if (pa_streq(id, "USB Audio"))
is_usb = true;

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@ -348,7 +348,7 @@ static pa_available_t calc_port_state(pa_device_port *p, struct userdata *u) {
break;
}
/* If the current availablility is unknown go the more precise no,
/* If the current availability is unknown go the more precise no,
* but otherwise don't change state */
if (pa == PA_AVAILABLE_UNKNOWN)
pa = cpa;

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@ -1447,7 +1447,7 @@ const pa_a2dp_endpoint_conf pa_a2dp_endpoint_conf_sbc = {
* SBC codec in Dual Channel mode, 8 bands, block length 16, allocation method Loudness,
* with bitpool adjusted to match target bitrates.
*
* Most commonly choosen bitrates and reasons are:
* Most commonly chosen bitrates and reasons are:
* 453000 - this yields most efficient packing of frames on Android for bluetooth EDR 2mbps
* 512000 - this looks to be old limit stated in bluetooth documents
* 552000 - this yields most efficient packing of frames on Android for bluetooth EDR 3mbps

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@ -1381,7 +1381,7 @@ pa_bluetooth_backend *pa_bluetooth_native_backend_new(pa_core *c, pa_bluetooth_d
backend->upower = pa_upower_backend_new(c, y);
/* All CIND indicators are enabled by default until overriden by AT+BIA */
/* All CIND indicators are enabled by default until overridden by AT+BIA */
for (i = 1; i < CIND_INDICATOR_MAX; i++)
backend->cind_enabled_indicators |= (1 << i);

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@ -2,7 +2,7 @@
This file is part of PulseAudio.
Copyright 2008-2013 João Paulo Rechi Vita
Copyrigth 2018-2019 Pali Rohár <pali.rohar@gmail.com>
Copyright 2018-2019 Pali Rohár <pali.rohar@gmail.com>
PulseAudio is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License as

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@ -5,7 +5,7 @@
This file is part of PulseAudio.
Copyright 2008-2013 João Paulo Rechi Vita
Copyrigth 2018-2019 Pali Rohár <pali.rohar@gmail.com>
Copyright 2018-2019 Pali Rohár <pali.rohar@gmail.com>
PulseAudio is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License as

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@ -2047,7 +2047,7 @@ static pa_hook_result_t sink_put_cb(void *hook_data, void *call_data, void *slot
pa_assert(c);
pa_assert(s);
/* We may have alredy encountered this sink, because if the new sink was
/* We may have already encountered this sink, because if the new sink was
* chosen as the default sink, the default sink change hook was fired
* first, and we saw the sink in default_sink_changed_cb(). */
if (pa_hashmap_get(c->sinks_by_index, PA_UINT32_TO_PTR(s->index)))
@ -2092,7 +2092,7 @@ static pa_hook_result_t source_put_cb(void *hook_data, void *call_data, void *sl
pa_assert(c);
pa_assert(s);
/* We may have alredy encountered this source, because if the new source
/* We may have already encountered this source, because if the new source
* was chosen as the default source, the default source change hook was
* fired first, and we saw the source in default_source_changed_cb(). */
if (pa_hashmap_get(c->sources_by_index, PA_UINT32_TO_PTR(s->index)))

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@ -121,7 +121,7 @@ struct userdata {
int64_t next_latency_with_drift;
int64_t next_latency_at_optimum_rate_with_drift;
/* Filter varables used for 2nd order filter */
/* Filter variables used for 2nd order filter */
double drift_filter;
double drift_compensation_rate;
@ -945,7 +945,7 @@ static void source_output_moving_cb(pa_source_output *o, pa_source *dest) {
u->log_counter = u->log_interval;
u->latency_error = 0;
/* Send a mesage to the output thread that the source has changed.
/* Send a message to the output thread that the source has changed.
* If the sink is invalid here during a profile switching situation
* we can safely set push_called to false directly. */
if (u->sink_input->sink)

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@ -1579,7 +1579,7 @@ static void update_preferred_device(struct userdata *u, const char *name, const
send_new_entry_signal(de);
} else {
/* We send a D-Bus signal when the device changes, but not when the
* card changes. That's becaues the D-Bus interface doesn't expose the
* card changes. That's because the D-Bus interface doesn't expose the
* card field to clients at all. */
if (!created_new_entry && !pa_safe_streq(entry->device, old->device))
send_device_updated_signal(de, entry);

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@ -673,7 +673,7 @@ static int do_init(pa_module *m) {
u->msg = pa_msgobject_new(tunnel_msg);
u->msg->parent.process_msg = tunnel_process_msg;
/* The rtpoll created here must curently only exist to avoid crashes when
/* The rtpoll created here must currently only exist to avoid crashes when
* the module is used together with module-loopback. Because module-loopback
* runs pa_asyncmsgq_process_one() from the pop callback, the rtpoll need not
* be run. We will do so anyway for potential modules similar to

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@ -1884,7 +1884,7 @@ static void setup_complete_callback(pa_pdispatch *pd, uint32_t command, uint32_t
#ifdef TUNNEL_SINK
pa_tagstruct_put_boolean(reply, false); /* volume_set */
#endif
pa_tagstruct_put_boolean(reply, true); /* early rquests */
pa_tagstruct_put_boolean(reply, true); /* early requests */
}
if (u->version >= 15) {

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@ -45,7 +45,7 @@ PA_MODULE_USAGE(
"tsched_buffer_size=<buffer size when using timer based scheduling> "
"fixed_latency_range=<disable latency range changes on underrun?> "
"ignore_dB=<ignore dB information from the device?> "
"deferred_volume=<syncronize sw and hw volume changes in IO-thread?> "
"deferred_volume=<synchronize sw and hw volume changes in IO-thread?> "
"use_ucm=<use ALSA UCM for card configuration?> "
"avoid_resampling=<use stream original sample rate if possible?>");

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@ -225,7 +225,7 @@ static void resolver_cb(
else
et = pa_xstrdup("none");
} else if (pa_streq(key, "cn")) {
/* Suported audio codecs:
/* Supported audio codecs:
* - 0 = PCM,
* - 1 = ALAC,
* - 2 = AAC,

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@ -350,7 +350,7 @@ static int sink_set_state_in_io_thread_cb(pa_sink *s, pa_sink_state_t new_state,
/* Connecting will trigger a RECORD and start streaming */
pa_raop_client_announce(u->raop);
} else if (!pa_raop_client_is_recording(u->raop)) {
/* RECORD alredy sent, simply start streaming */
/* RECORD already sent, simply start streaming */
pa_raop_client_stream(u->raop);
pa_rtpoll_set_timer_absolute(u->rtpoll, now);
u->write_count = 0;

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@ -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 switch to user busses) */
DBusConnection *connection, /* Session bus (when D-Bus learns about user buses we should switch to user buses) */
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. */

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@ -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 switch to user busses) */
DBusConnection *connection, /* Session bus (when D-Bus learns about user buses we should switch to user buses) */
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 */

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@ -87,7 +87,7 @@ static const char* const valid_modargs[] = {
"format",
"channels",
"rate",
"destination", /* Compatbility */
"destination", /* Compatibility */
"destination_ip",
"source_ip",
"port",

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@ -275,7 +275,7 @@ int pa_rtp_send(pa_rtp_context *c, pa_memblockq *q) {
return -1;
/*
* While we check here for atleast MTU worth of data being available in
* While we check here for at least MTU worth of data being available in
* memblockq, we might not have exact equivalent to MTU. Hence, we walk
* over the memchunks in memblockq and accumulate MTU bytes next.
*/

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@ -845,7 +845,7 @@ typedef enum pa_sink_flags {
* sinks). */
PA_SINK_DEFERRED_VOLUME = 0x2000000U,
/**< The HW volume changes are syncronized with SW volume. */
/**< The HW volume changes are synchronized with SW volume. */
/** \endcond */
#endif
@ -964,7 +964,7 @@ typedef enum pa_source_flags {
* sources). */
PA_SOURCE_DEFERRED_VOLUME = 0x2000000U,
/**< The HW volume changes are syncronized with SW volume. */
/**< The HW volume changes are synchronized with SW volume. */
#endif
} pa_source_flags_t;

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@ -62,7 +62,7 @@ typedef enum pa_encoding {
PA_ENCODING_DTSHD_IEC61937,
/**< DTS-HD Master Audio encapsulated in IEC 61937 header/padding. \since 13.0 */
/* Remeber to update
/* Remember to update
* https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/SupportedAudioFormats/
* when adding new encodings! */

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@ -235,7 +235,7 @@ typedef struct pa_sink_port_info {
const char *description; /**< Description of this port */
uint32_t priority; /**< The higher this value is, the more useful this port is as a default. */
int available; /**< A flags (see #pa_port_available), indicating availability status of this port. \since 2.0 */
const char *availability_group; /**< An indentifier for the group of ports that share their availability status with
const char *availability_group; /**< An identifier for the group of ports that share their availability status with
* each other. This is meant especially for handling cases where one 3.5 mm connector
* is used for headphones, headsets and microphones, and the hardware can only tell
* that something was plugged in but not what exactly. In this situation the ports for
@ -339,7 +339,7 @@ typedef struct pa_source_port_info {
const char *description; /**< Description of this port */
uint32_t priority; /**< The higher this value is, the more useful this port is as a default. */
int available; /**< A flags (see #pa_port_available), indicating availability status of this port. \since 2.0 */
const char *availability_group; /**< An indentifier for the group of ports that share their availability status with
const char *availability_group; /**< An identifier for the group of ports that share their availability status with
* each other. This is meant especially for handling cases where one 3.5 mm connector
* is used for headphones, headsets and microphones, and the hardware can only tell
* that something was plugged in but not what exactly. In this situation the ports for
@ -575,7 +575,7 @@ typedef struct pa_card_port_info {
pa_proplist *proplist; /**< Property list */
int64_t latency_offset; /**< Latency offset of the port that gets added to the sink/source latency when the port is active. \since 3.0 */
pa_card_profile_info2** profiles2; /**< Array of pointers to available profiles, or NULL. Array is terminated by an entry set to NULL. \since 5.0 */
const char *availability_group; /**< An indentifier for the group of ports that share their availability status with
const char *availability_group; /**< An identifier for the group of ports that share their availability status with
* each other. This is meant especially for handling cases where one 3.5 mm connector
* is used for headphones, headsets and microphones, and the hardware can only tell
* that something was plugged in but not what exactly. In this situation the ports for

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@ -171,7 +171,7 @@ typedef enum pa_sample_format {
PA_SAMPLE_S24_32BE,
/**< Signed 24 Bit PCM in LSB of 32 Bit words, big endian. \since 0.9.15 */
/* Remeber to update
/* Remember to update
* https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/SupportedAudioFormats/
* when adding new formats! */
@ -337,12 +337,12 @@ char* pa_bytes_snprint(char *s, size_t l, unsigned v);
/** Returns 1 when the specified format is little endian, 0 when
* big endian. Returns -1 when endianness does not apply to the
* specified format, or endianess is unknown. \since 0.9.16 */
* specified format, or endianness is unknown. \since 0.9.16 */
int pa_sample_format_is_le(pa_sample_format_t f) PA_GCC_PURE;
/** Returns 1 when the specified format is big endian, 0 when
* little endian. Returns -1 when endianness does not apply to the
* specified format, or endianess is unknown. \since 0.9.16 */
* specified format, or endianness is unknown. \since 0.9.16 */
int pa_sample_format_is_be(pa_sample_format_t f) PA_GCC_PURE;
#ifdef WORDS_BIGENDIAN
@ -351,11 +351,11 @@ int pa_sample_format_is_be(pa_sample_format_t f) PA_GCC_PURE;
#else
/** Returns 1 when the specified format is native endian, 0 when
* not. Returns -1 when endianness does not apply to the
* specified format, or endianess is unknown. \since 0.9.16 */
* specified format, or endianness is unknown. \since 0.9.16 */
#define pa_sample_format_is_ne(f) pa_sample_format_is_le(f)
/** Returns 1 when the specified format is reverse endian, 0 when
* native. Returns -1 when endianness does not apply to the
* specified format, or endianess is unknown. \since 0.9.16 */
* specified format, or endianness is unknown. \since 0.9.16 */
#define pa_sample_format_is_re(f) pa_sample_format_is_be(f)
#endif

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@ -161,7 +161,7 @@
* \li pa_stream_cork() - Start or stop the playback or recording.
* \li pa_stream_trigger() - Start playback immediately and do not wait for
* the buffer to fill up to the set trigger level.
* \li pa_stream_prebuf() - Reenable the playback trigger level.
* \li pa_stream_prebuf() - Re-enable the playback trigger level.
* \li pa_stream_drain() - Wait for the playback buffer to go empty. Will
* return a pa_operation object that will indicate when
* the buffer is completely drained.
@ -664,7 +664,7 @@ pa_operation* pa_stream_cork(pa_stream *s, int b, pa_stream_success_cb_t cb, voi
* \a seek of pa_stream_write() instead of this function. */
pa_operation* pa_stream_flush(pa_stream *s, pa_stream_success_cb_t cb, void *userdata);
/** Reenable prebuffering if specified in the pa_buffer_attr
/** Re-enable prebuffering if specified in the pa_buffer_attr
* structure. Available for playback streams only. */
pa_operation* pa_stream_prebuf(pa_stream *s, pa_stream_success_cb_t cb, void *userdata);

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@ -220,7 +220,7 @@ char *pa_sw_volume_snprint_dB(char *s, size_t l, pa_volume_t v);
/** Maximum length of the strings returned by pa_volume_snprint_verbose().
* Please note that this value can change with any release without warning and
* withou being considered API or ABI breakage. You should not use this
* without being considered API or ABI breakage. You should not use this
* definition anywhere where it might become part of an ABI. \since 5.0 */
#define PA_VOLUME_SNPRINT_VERBOSE_MAX 35

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@ -367,7 +367,7 @@ fail:
return -1;
}
/* Return a newly allocated sting containing the parent directory of the specified file */
/* Return a newly allocated string containing the parent directory of the specified file */
char *pa_parent_dir(const char *fn) {
char *slash, *dir = pa_xstrdup(fn);
@ -2085,7 +2085,7 @@ char *pa_find_config_file(const char *global, const char *local, const char *env
return NULL;
}
/* Format the specified data as a hexademical string */
/* Format the specified data as a hexadecimal string */
char *pa_hexstr(const uint8_t* d, size_t dlength, char *s, size_t slength) {
size_t i = 0, j = 0;
const char hex[] = "0123456789abcdef";
@ -2317,7 +2317,7 @@ static int prepare_number_string(const char *s, enum numtype type, char **tmp, c
/* The strtoul and strtoull functions allow a minus sign even though they
* parse an unsigned number. In case of a minus sign the original negative
* number gets negated. We don't want that kind of behviour. */
* number gets negated. We don't want that kind of behaviour. */
if (type == NUMTYPE_UINT && s[0] == '-')
return -1;

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@ -47,7 +47,7 @@ struct pa_device_port {
unsigned priority;
pa_available_t available; /* PA_AVAILABLE_UNKNOWN, PA_AVAILABLE_NO or PA_AVAILABLE_YES */
char *availability_group; /* a string indentifier which determine the group of devices handling the available state simulteneously */
char *availability_group; /* a string identifier which determine the group of devices handling the available state simulteneously */
pa_proplist *proplist;
pa_hashmap *profiles; /* Does not own the profiles */

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@ -15,7 +15,7 @@ extern "C" {
* are stored in x1 and x2, and the previous two outputs are stored in y1 and
* y2.
*
* We use double during the coefficients calculation for better accurary, but
* We use double during the coefficients calculation for better accuracy, but
* float is used during the actual filtering for faster computation.
*/
struct biquad {

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@ -4,7 +4,7 @@
* charge.
*
* SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
* THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
* THE WARRANTIES OF DESIGN, MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
*
* Sun source code is provided with no support and without any obligation on

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@ -120,7 +120,7 @@ void pa_random(void *ret_data, size_t length) {
return;
if (!has_whined) {
pa_log_warn("Failed to get proper entropy. Falling back to unsecure pseudo RNG.");
pa_log_warn("Failed to get proper entropy. Falling back to insecure pseudo RNG.");
has_whined = true;
}

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@ -562,7 +562,7 @@ size_t pa_resampler_result(pa_resampler *r, size_t in_length) {
if (!r)
return in_length;
/* Convert to intput frames */
/* Convert to input frames */
in_length = in_length / r->i_fz;
/* soxr processes samples in blocks, depending on the ratio.

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@ -42,7 +42,7 @@ static void underflow_cb(struct pa_stream *s, void *userdata) {
}
static void overflow_cb(struct pa_stream *s, void *userdata) {
pa_log_warn("Overlow");
pa_log_warn("Overflow");
}
/*