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authorwm4 <wm4@nowhere>2013-04-07 21:34:09 +0200
committerwm4 <wm4@nowhere>2013-05-12 21:24:55 +0200
commitd2e5b500413c379ff11bf1fb5acab6608e355862 (patch)
treea4c31261dc76816d4e11a099aa83a832457e89ea /audio
parent7f0f33fc8f105144eaac9653564e91599692e1e7 (diff)
downloadmpv-d2e5b500413c379ff11bf1fb5acab6608e355862.tar.bz2
mpv-d2e5b500413c379ff11bf1fb5acab6608e355862.tar.xz
ao_alsa: cosmetics, macro-fy error reporting
Add a CHECK_ALSA_ERROR macro to report ALSA errors. This is similar to what vo_vdpau does. This removes lots of boiler plate, it almost gives me the feeling the ao_alsa initialization code is now readable. This change is squashed with the reformatting, because both changes are just as noisy and useless.
Diffstat (limited to 'audio')
-rw-r--r--audio/out/ao_alsa.c502
1 files changed, 181 insertions, 321 deletions
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index 7c925687c9..366af2e854 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -60,7 +60,7 @@ static const ao_info_t info =
LIBAO_EXTERN(alsa)
-static snd_pcm_t * alsa_handler;
+static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
#define BUFFER_TIME 500000 // 0.5 s
@@ -74,6 +74,15 @@ static float delay_before_pause;
#define ALSA_DEVICE_SIZE 256
+#define CHECK_ALSA_ERROR(message) \
+ do { \
+ if (err < 0) { \
+ mp_msg(MSGT_VO, MSGL_ERR, "[AO_ALSA] %s: %s\n", \
+ (message), snd_strerror(err)); \
+ goto alsa_error; \
+ } \
+ } while (0)
+
static void alsa_error_handler(const char *file, int line, const char *function,
int err, const char *format, ...)
{
@@ -84,17 +93,19 @@ static void alsa_error_handler(const char *file, int line, const char *function,
vsnprintf(tmp, sizeof tmp, format, va);
va_end(va);
- if (err)
+ if (err) {
mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
file, line, function, tmp, snd_strerror(err));
- else
+ } else {
mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
file, line, function, tmp);
+ }
}
/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
+ snd_mixer_t *handle = NULL;
switch (cmd) {
case AOCONTROL_GET_MUTE:
case AOCONTROL_SET_MUTE:
@@ -102,7 +113,6 @@ static int control(int cmd, void *arg)
case AOCONTROL_SET_VOLUME:
{
int err;
- snd_mixer_t *handle;
snd_mixer_elem_t *elem;
snd_mixer_selem_id_t *sid;
@@ -148,44 +158,25 @@ static int control(int cmd, void *arg)
mix_name = NULL;
}
- if ((err = snd_mixer_open(&handle, 0)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Mixer open error: %s\n",
- snd_strerror(
- err));
- return CONTROL_ERROR;
- }
+ err = snd_mixer_open(&handle, 0);
+ CHECK_ALSA_ERROR("Mixer open error");
- if ((err = snd_mixer_attach(handle, card)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Mixer attach %s error: %s\n",
- card, snd_strerror(err));
- snd_mixer_close(handle);
- return CONTROL_ERROR;
- }
+ err = snd_mixer_attach(handle, card);
+ CHECK_ALSA_ERROR("Mixer attach error");
+
+ err = snd_mixer_selem_register(handle, NULL, NULL);
+ CHECK_ALSA_ERROR("Mixer register error");
- if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Mixer register error: %s\n",
- snd_strerror(
- err));
- snd_mixer_close(handle);
- return CONTROL_ERROR;
- }
err = snd_mixer_load(handle);
- if (err < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Mixer load error: %s\n",
- snd_strerror(
- err));
- snd_mixer_close(handle);
- return CONTROL_ERROR;
- }
+ CHECK_ALSA_ERROR("Mixer load error");
elem = snd_mixer_find_selem(handle, sid);
if (!elem) {
mp_tmsg(MSGT_AO, MSGL_ERR,
"[AO_ALSA] Unable to find simple control '%s',%i.\n",
- snd_mixer_selem_id_get_name(
- sid), snd_mixer_selem_id_get_index(sid));
- snd_mixer_close(handle);
- return CONTROL_ERROR;
+ snd_mixer_selem_id_get_name(sid),
+ snd_mixer_selem_id_get_index(sid));
+ goto alsa_error;
}
snd_mixer_selem_get_playback_volume_range(elem, &pmin, &pmax);
@@ -197,30 +188,16 @@ static int control(int cmd, void *arg)
set_vol = vol->left / f_multi + pmin + 0.5;
//setting channels
- if ((err =
- snd_mixer_selem_set_playback_volume(elem,
- SND_MIXER_SCHN_FRONT_LEFT,
- set_vol)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Error setting left channel, %s\n",
- snd_strerror(
- err));
- goto mixer_error;
- }
+ err = snd_mixer_selem_set_playback_volume
+ (elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol);
+ CHECK_ALSA_ERROR("Error setting left channel");
mp_msg(MSGT_AO, MSGL_DBG2, "left=%li, ", set_vol);
set_vol = vol->right / f_multi + pmin + 0.5;
- if ((err =
- snd_mixer_selem_set_playback_volume(elem,
- SND_MIXER_SCHN_FRONT_RIGHT,
- set_vol)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Error setting right channel, %s\n",
- snd_strerror(
- err));
- goto mixer_error;
- }
+ err = snd_mixer_selem_set_playback_volume
+ (elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol);
+ CHECK_ALSA_ERROR("Error setting right channel");
mp_msg(MSGT_AO, MSGL_DBG2,
"right=%li, pmin=%li, pmax=%li, mult=%f\n",
set_vol, pmin, pmax,
@@ -229,12 +206,11 @@ static int control(int cmd, void *arg)
}
case AOCONTROL_GET_VOLUME: {
ao_control_vol_t *vol = arg;
- snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT,
- &get_vol);
+ snd_mixer_selem_get_playback_volume
+ (elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
vol->left = (get_vol - pmin) * f_multi;
- snd_mixer_selem_get_playback_volume(elem,
- SND_MIXER_SCHN_FRONT_RIGHT,
- &get_vol);
+ snd_mixer_selem_get_playback_volume
+ (elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
vol->right = (get_vol - pmin) * f_multi;
mp_msg(MSGT_AO, MSGL_DBG2, "left=%f, right=%f\n", vol->left,
vol->right);
@@ -243,26 +219,26 @@ static int control(int cmd, void *arg)
case AOCONTROL_SET_MUTE: {
bool *mute = arg;
if (!snd_mixer_selem_has_playback_switch(elem))
- goto mixer_error;
+ goto alsa_error;
if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
- snd_mixer_selem_set_playback_switch(
- elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
+ snd_mixer_selem_set_playback_switch
+ (elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
}
- snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
- !*mute);
+ snd_mixer_selem_set_playback_switch
+ (elem, SND_MIXER_SCHN_FRONT_LEFT, !*mute);
break;
}
case AOCONTROL_GET_MUTE: {
bool *mute = arg;
if (!snd_mixer_selem_has_playback_switch(elem))
- goto mixer_error;
+ goto alsa_error;
int tmp = 1;
- snd_mixer_selem_get_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
- &tmp);
+ snd_mixer_selem_get_playback_switch
+ (elem, SND_MIXER_SCHN_FRONT_LEFT, &tmp);
*mute = !tmp;
if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
- snd_mixer_selem_get_playback_switch(
- elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
+ snd_mixer_selem_get_playback_switch
+ (elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
*mute &= !tmp;
}
break;
@@ -270,13 +246,15 @@ static int control(int cmd, void *arg)
}
snd_mixer_close(handle);
return CONTROL_OK;
-mixer_error:
- snd_mixer_close(handle);
- return CONTROL_ERROR;
}
} //end switch
return CONTROL_UNKNOWN;
+
+alsa_error:
+ if (handle)
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
}
static void parse_device(char *dest, const char *src, int len)
@@ -326,9 +304,9 @@ static int try_open_device(const char *device, int open_mode, int try_ac3)
/* no existing parameters: add it behind device name */
strcat(ac3_device, ":AES0=6");
} else {
- do
+ do {
++args;
- while (isspace(*args));
+ } while (isspace(*args));
if (*args == '\0') {
/* ":" but no parameters */
strcat(ac3_device, "AES0=6");
@@ -337,21 +315,21 @@ static int try_open_device(const char *device, int open_mode, int try_ac3)
strcat(ac3_device, ",AES0=6");
} else {
/* parameters in config syntax: add it inside the { } block */
- do
+ do {
--len;
- while (len > 0 && isspace(ac3_device[len]));
+ } while (len > 0 && isspace(ac3_device[len]));
if (ac3_device[len] == '}')
strcpy(ac3_device + len, " AES0=6}");
}
}
- err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
- open_mode);
+ err = snd_pcm_open
+ (&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK, open_mode);
free(ac3_device);
if (!err)
return 0;
}
- return snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
- open_mode);
+ return snd_pcm_open
+ (&alsa_handler, device, SND_PCM_STREAM_PLAYBACK, open_mode);
}
/*
@@ -462,7 +440,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
mp_msg(MSGT_AO, MSGL_V,
"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n",
ao_data.channels.num);
- } else
+ } else {
/* in any case for multichannel playback we should select
* appropriate device
*/
@@ -500,6 +478,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
"[AO_ALSA] %d channels are not supported.\n",
ao_data.channels.num);
}
+ }
device.len = strlen(device.str);
if (subopt_parse(ao_subdevice, subopts) != 0) {
print_help();
@@ -515,31 +494,24 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
int open_mode = block ? 0 : SND_PCM_NONBLOCK;
int isac3 = AF_FORMAT_IS_IEC61937(format);
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
- if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0) {
+ err = try_open_device(alsa_device, open_mode, isac3);
+ if (err < 0) {
if (err != -EBUSY && !block) {
- mp_tmsg(
- MSGT_AO, MSGL_INFO,
- "[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
- if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Playback open error: %s\n", snd_strerror(
- err));
- return 0;
- }
- } else {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Playback open error: %s\n", snd_strerror(
- err));
- return 0;
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Open in nonblock-mode "
+ "failed, trying to open in block-mode.\n");
+ err = try_open_device(alsa_device, 0, isac3);
}
+ CHECK_ALSA_ERROR("Playback open error");
}
- if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0)
+ err = snd_pcm_nonblock(alsa_handler, 0);
+ if (err < 0) {
mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(
- err));
- else
+ "[AL_ALSA] Error setting block-mode %s.\n",
+ snd_strerror(err));
+ } else {
mp_msg(MSGT_AO, MSGL_V, "alsa-init: pcm opened in blocking mode\n");
+ }
snd_pcm_hw_params_t *alsa_hwparams;
snd_pcm_sw_params_t *alsa_swparams;
@@ -548,32 +520,20 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
snd_pcm_sw_params_alloca(&alsa_swparams);
// setting hw-parameters
- if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to get initial parameters: %s\n",
- snd_strerror(
- err));
- return 0;
- }
+ err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams);
+ CHECK_ALSA_ERROR("Unable to get initial parameters");
- err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
- SND_PCM_ACCESS_RW_INTERLEAVED);
- if (err < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to set access type: %s\n",
- snd_strerror(
- err));
- return 0;
- }
+ err = snd_pcm_hw_params_set_access
+ (alsa_handler, alsa_hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
+ CHECK_ALSA_ERROR("Unable to set access type");
/* workaround for nonsupported formats
sets default format to S16_LE if the given formats aren't supported */
- if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
- alsa_format)) < 0) {
- mp_tmsg(
- MSGT_AO, MSGL_INFO,
- "[AO_ALSA] Format %s is not supported by hardware, trying default.\n",
- af_fmt2str_short(format));
+ err = snd_pcm_hw_params_test_format
+ (alsa_handler, alsa_hwparams, alsa_format);
+ if (err < 0) {
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Format %s is not supported "
+ "by hardware, trying default.\n", af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao_data.format))
ao_data.format = AF_FORMAT_AC3_LE;
@@ -583,168 +543,89 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
ao_data.format = AF_FORMAT_S16_LE;
}
- if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
- alsa_format)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to set format: %s\n",
- snd_strerror(err));
- return 0;
- }
+ err = snd_pcm_hw_params_set_format
+ (alsa_handler, alsa_hwparams, alsa_format);
+ CHECK_ALSA_ERROR("Unable to set format");
int num_channels = ao_data.channels.num;
- if ((err =
- snd_pcm_hw_params_set_channels_near(alsa_handler,
- alsa_hwparams,
- &num_channels)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to set channels: %s\n",
- snd_strerror(err));
- return 0;
- }
+ err = snd_pcm_hw_params_set_channels_near
+ (alsa_handler, alsa_hwparams, &num_channels);
+ CHECK_ALSA_ERROR("Unable to set channels");
+
mp_chmap_from_channels(&ao_data.channels, num_channels);
if (!AF_FORMAT_IS_IEC61937(format))
mp_chmap_reorder_to_alsa(&ao_data.channels);
+
/* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
prefer our own resampler, since that allows users to choose the resampler,
even per file if desired */
- if ((err =
- snd_pcm_hw_params_set_rate_resample(alsa_handler,
- alsa_hwparams,
- 0)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to disable resampling: %s\n",
- snd_strerror(
- err));
- return 0;
- }
+ err = snd_pcm_hw_params_set_rate_resample
+ (alsa_handler, alsa_hwparams, 0);
+ CHECK_ALSA_ERROR("Unable to disable resampling");
- if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
- &ao_data.samplerate,
- NULL)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to set samplerate-2: %s\n",
- snd_strerror(
- err));
- return 0;
- }
+ err = snd_pcm_hw_params_set_rate_near
+ (alsa_handler, alsa_hwparams, &ao_data.samplerate, NULL);
+ CHECK_ALSA_ERROR("Unable to set samplerate-2");
bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
bytes_per_sample *= ao_data.channels.num;
ao_data.bps = ao_data.samplerate * bytes_per_sample;
- if ((err =
- snd_pcm_hw_params_set_buffer_time_near(alsa_handler,
- alsa_hwparams,
- &(unsigned int){
- BUFFER_TIME},
- NULL)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to set buffer time near: %s\n",
- snd_strerror(
- err));
- return 0;
- }
+ err = snd_pcm_hw_params_set_buffer_time_near
+ (alsa_handler, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL);
+ CHECK_ALSA_ERROR("Unable to set buffer time near");
- if ((err =
- snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
- &(unsigned int){FRAGCOUNT},
- NULL)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to set periods: %s\n",
- snd_strerror(err));
- return 0;
- }
+ err = snd_pcm_hw_params_set_periods_near
+ (alsa_handler, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
+ CHECK_ALSA_ERROR("Unable to set periods");
/* finally install hardware parameters */
- if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to set hw-parameters: %s\n",
- snd_strerror(
- err));
- return 0;
- }
- // end setting hw-params
+ err = snd_pcm_hw_params(alsa_handler, alsa_hwparams);
+ CHECK_ALSA_ERROR("Unable to set hw-parameters");
+ // end setting hw-params
// gets buffersize for control
- if ((err =
- snd_pcm_hw_params_get_buffer_size(alsa_hwparams,
- &bufsize)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(
- err));
- return 0;
- } else {
- ao_data.buffersize = bufsize * bytes_per_sample;
- mp_msg(MSGT_AO, MSGL_V, "alsa-init: got buffersize=%i\n",
- ao_data.buffersize);
- }
+ err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
+ CHECK_ALSA_ERROR("Unable to get buffersize");
- if ((err =
- snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size,
- NULL)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO ALSA] Unable to get period size: %s\n", snd_strerror(
- err));
- return 0;
- } else
- mp_msg(MSGT_AO, MSGL_V, "alsa-init: got period size %li\n",
- chunk_size);
+ ao_data.buffersize = bufsize * bytes_per_sample;
+ mp_msg(MSGT_AO, MSGL_V, "alsa-init: got buffersize=%i\n",
+ ao_data.buffersize);
+
+ err = snd_pcm_hw_params_get_period_size
+ (alsa_hwparams, &chunk_size, NULL);
+ CHECK_ALSA_ERROR("Unable to get period size");
+
+ mp_msg(MSGT_AO, MSGL_V, "alsa-init: got period size %li\n", chunk_size);
ao_data.outburst = chunk_size * bytes_per_sample;
/* setting software parameters */
- if ((err =
- snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to get sw-parameters: %s\n",
- snd_strerror(
- err));
- return 0;
- }
- if ((err =
- snd_pcm_sw_params_get_boundary(alsa_swparams,
- &boundary)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to get boundary: %s\n",
- snd_strerror(err));
- return 0;
- }
+ err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams);
+ CHECK_ALSA_ERROR("Unable to get sw-parameters");
+
+ err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
+ CHECK_ALSA_ERROR("Unable to get boundary");
+
/* start playing when one period has been written */
- if ((err =
- snd_pcm_sw_params_set_start_threshold(alsa_handler,
- alsa_swparams,
- chunk_size)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to set start threshold: %s\n",
- snd_strerror(
- err));
- return 0;
- }
+ err = snd_pcm_sw_params_set_start_threshold
+ (alsa_handler, alsa_swparams, chunk_size);
+ CHECK_ALSA_ERROR("Unable to set start threshold");
+
/* disable underrun reporting */
- if ((err =
- snd_pcm_sw_params_set_stop_threshold(alsa_handler,
- alsa_swparams,
- boundary)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to set stop threshold: %s\n",
- snd_strerror(
- err));
- return 0;
- }
+ err = snd_pcm_sw_params_set_stop_threshold
+ (alsa_handler, alsa_swparams, boundary);
+ CHECK_ALSA_ERROR("Unable to set stop threshold");
+
/* play silence when there is an underrun */
- if ((err =
- snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams,
- boundary)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to set silence size: %s\n",
- snd_strerror(
- err));
- return 0;
- }
- if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR,
- "[AO_ALSA] Unable to get sw-parameters: %s\n",
- snd_strerror(
- err));
- return 0;
- }
+ err = snd_pcm_sw_params_set_silence_size
+ (alsa_handler, alsa_swparams, boundary);
+ CHECK_ALSA_ERROR("Unable to set silence size");
+
+ err = snd_pcm_sw_params(alsa_handler, alsa_swparams);
+ CHECK_ALSA_ERROR("Unable to get sw-parameters");
+
/* end setting sw-params */
alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
@@ -752,12 +633,13 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
mp_msg(MSGT_AO, MSGL_V,
"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
ao_data.samplerate, ao_data.channels.num, (int)bytes_per_sample,
- ao_data.buffersize,
- snd_pcm_format_description(
- alsa_format));
+ ao_data.buffersize, snd_pcm_format_description(alsa_format));
} // end switch alsa_handler (spdif)
return 1;
+
+alsa_error:
+ return 0;
} // end init
@@ -771,17 +653,16 @@ static void uninit(int immed)
if (!immed)
snd_pcm_drain(alsa_handler);
- if ((err = snd_pcm_close(alsa_handler)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm close error: %s\n",
- snd_strerror(
- err));
- return;
- } else {
- alsa_handler = NULL;
- mp_msg(MSGT_AO, MSGL_V, "alsa-uninit: pcm closed\n");
- }
- } else
+ err = snd_pcm_close(alsa_handler);
+ CHECK_ALSA_ERROR("pcm close error");
+
+ alsa_handler = NULL;
+ mp_msg(MSGT_AO, MSGL_V, "alsa-uninit: pcm closed\n");
+ } else {
mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] No handler defined!\n");
+ }
+
+alsa_error: ;
}
static void audio_pause(void)
@@ -790,12 +671,8 @@ static void audio_pause(void)
if (alsa_can_pause) {
delay_before_pause = get_delay();
- if ((err = snd_pcm_pause(alsa_handler, 1)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm pause error: %s\n",
- snd_strerror(
- err));
- return;
- }
+ err = snd_pcm_pause(alsa_handler, 1);
+ CHECK_ALSA_ERROR("pcm pause error");
mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n");
} else {
if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
@@ -803,13 +680,11 @@ static void audio_pause(void)
prepause_frames = 0;
delay_before_pause = prepause_frames / (float)ao_data.samplerate;
- if ((err = snd_pcm_drop(alsa_handler)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm drop error: %s\n",
- snd_strerror(
- err));
- return;
- }
+ err = snd_pcm_drop(alsa_handler);
+ CHECK_ALSA_ERROR("pcm drop error");
}
+
+alsa_error: ;
}
static void audio_resume(void)
@@ -823,26 +698,20 @@ static void audio_resume(void)
sleep(1);
}
if (alsa_can_pause) {
- if ((err = snd_pcm_pause(alsa_handler, 0)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm resume error: %s\n",
- snd_strerror(
- err));
- return;
- }
+ err = snd_pcm_pause(alsa_handler, 0);
+ CHECK_ALSA_ERROR("pcm resume error");
mp_msg(MSGT_AO, MSGL_V, "alsa-resume: resume supported by hardware\n");
} else {
- if ((err = snd_pcm_prepare(alsa_handler)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm prepare error: %s\n",
- snd_strerror(
- err));
- return;
- }
+ err = snd_pcm_prepare(alsa_handler);
+ CHECK_ALSA_ERROR("pcm prepare error");
if (prepause_frames) {
void *silence = calloc(prepause_frames, bytes_per_sample);
play(silence, prepause_frames * bytes_per_sample, 0);
free(silence);
}
}
+
+alsa_error: ;
}
/* stop playing and empty buffers (for seeking/pause) */
@@ -852,19 +721,12 @@ static void reset(void)
prepause_frames = 0;
delay_before_pause = 0;
- if ((err = snd_pcm_drop(alsa_handler)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm prepare error: %s\n",
- snd_strerror(
- err));
- return;
- }
- if ((err = snd_pcm_prepare(alsa_handler)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm prepare error: %s\n",
- snd_strerror(
- err));
- return;
- }
- return;
+ err = snd_pcm_drop(alsa_handler);
+ CHECK_ALSA_ERROR("pcm prepare error");
+ err = snd_pcm_prepare(alsa_handler);
+ CHECK_ALSA_ERROR("pcm prepare error");
+
+alsa_error: ;
}
/*
@@ -906,42 +768,40 @@ static int play(void *data, int len, int flags)
}
if (res < 0) {
mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Write error: %s\n",
- snd_strerror(
- res));
+ snd_strerror(res));
mp_tmsg(MSGT_AO, MSGL_INFO,
"[AO_ALSA] Trying to reset soundcard.\n");
- if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm prepare error: %s\n",
- snd_strerror(
- res));
- break;
- }
+ res = snd_pcm_prepare(alsa_handler);
+ int err = res;
+ CHECK_ALSA_ERROR("pcm prepare error");
res = 0;
}
} while (res == 0);
return res < 0 ? 0 : res * bytes_per_sample;
+
+alsa_error:
+ return 0;
}
/* how many byes are free in the buffer */
static int get_space(void)
{
snd_pcm_status_t *status;
- int ret;
+ int err;
snd_pcm_status_alloca(&status);
- if ((ret = snd_pcm_status(alsa_handler, status)) < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Cannot get pcm status: %s\n",
- snd_strerror(
- ret));
- return 0;
- }
+ err = snd_pcm_status(alsa_handler, status);
+ CHECK_ALSA_ERROR("cannot get pcm status");
unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
if (space > ao_data.buffersize) // Buffer underrun?
space = ao_data.buffersize;
return space;
+
+alsa_error:
+ return 0;
}
/* delay in seconds between first and last sample in buffer */