From 7f0f33fc8f105144eaac9653564e91599692e1e7 Mon Sep 17 00:00:00 2001 From: wm4 Date: Sun, 7 Apr 2013 21:34:09 +0200 Subject: ao_alsa: uncrustify --- audio/out/ao_alsa.c | 1340 +++++++++++++++++++++++++++------------------------ 1 file changed, 717 insertions(+), 623 deletions(-) (limited to 'audio') diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c index a194ce0478..7c925687c9 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 @@ -75,263 +75,289 @@ static float delay_before_pause; #define ALSA_DEVICE_SIZE 256 static void alsa_error_handler(const char *file, int line, const char *function, - int err, const char *format, ...) + int err, const char *format, ...) { - char tmp[0xc00]; - va_list va; - - va_start(va, format); - vsnprintf(tmp, sizeof tmp, format, va); - va_end(va); - - 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 - mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n", - file, line, function, tmp); + char tmp[0xc00]; + va_list va; + + va_start(va, format); + vsnprintf(tmp, sizeof tmp, format, va); + va_end(va); + + 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 + 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) { - switch(cmd) { - case AOCONTROL_GET_MUTE: - case AOCONTROL_SET_MUTE: - case AOCONTROL_GET_VOLUME: - case AOCONTROL_SET_VOLUME: + switch (cmd) { + case AOCONTROL_GET_MUTE: + case AOCONTROL_SET_MUTE: + case AOCONTROL_GET_VOLUME: + case AOCONTROL_SET_VOLUME: { - int err; - snd_mixer_t *handle; - snd_mixer_elem_t *elem; - snd_mixer_selem_id_t *sid; - - char *mix_name = "Master"; - char *card = "default"; - int mix_index = 0; - - long pmin, pmax; - long get_vol, set_vol; - float f_multi; - - if(AF_FORMAT_IS_IEC61937(ao_data.format)) - return CONTROL_TRUE; - - if(mixer_channel) { - char *test_mix_index; - - mix_name = strdup(mixer_channel); - if ((test_mix_index = strchr(mix_name, ','))){ - *test_mix_index = 0; - test_mix_index++; - mix_index = strtol(test_mix_index, &test_mix_index, 0); - - if (*test_mix_index){ - mp_tmsg(MSGT_AO,MSGL_ERR, - "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n"); - mix_index = 0 ; - } - } - } - if(mixer_device) card = mixer_device; - - //allocate simple id - snd_mixer_selem_id_alloca(&sid); - - //sets simple-mixer index and name - snd_mixer_selem_id_set_index(sid, mix_index); - snd_mixer_selem_id_set_name(sid, mix_name); - - if (mixer_channel) { - free(mix_name); - 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; - } - - 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; - } - - 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; - } - - 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_get_playback_volume_range(elem,&pmin,&pmax); - f_multi = (100 / (float)(pmax - pmin)); - - switch (cmd) { - case AOCONTROL_SET_VOLUME: { - ao_control_vol_t *vol = 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; - } - 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; - } - mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n", - set_vol, pmin, pmax, f_multi); - break; - } - case AOCONTROL_GET_VOLUME: { - ao_control_vol_t *vol = arg; - 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); - vol->right = (get_vol - pmin) * f_multi; - mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right); - break; - } - case AOCONTROL_SET_MUTE: { - bool *mute = arg; - if (!snd_mixer_selem_has_playback_switch(elem)) - goto mixer_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_LEFT, - !*mute); - break; - } - case AOCONTROL_GET_MUTE: { - bool *mute = arg; - if (!snd_mixer_selem_has_playback_switch(elem)) - goto mixer_error; - int tmp = 1; - 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); - *mute &= !tmp; + int err; + snd_mixer_t *handle; + snd_mixer_elem_t *elem; + snd_mixer_selem_id_t *sid; + + char *mix_name = "Master"; + char *card = "default"; + int mix_index = 0; + + long pmin, pmax; + long get_vol, set_vol; + float f_multi; + + if (AF_FORMAT_IS_IEC61937(ao_data.format)) + return CONTROL_TRUE; + + if (mixer_channel) { + char *test_mix_index; + + mix_name = strdup(mixer_channel); + if ((test_mix_index = strchr(mix_name, ','))) { + *test_mix_index = 0; + test_mix_index++; + mix_index = strtol(test_mix_index, &test_mix_index, 0); + + if (*test_mix_index) { + mp_tmsg(MSGT_AO, MSGL_ERR, + "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n"); + mix_index = 0; + } + } } - break; - } - } - snd_mixer_close(handle); - return CONTROL_OK; - mixer_error: - snd_mixer_close(handle); - return CONTROL_ERROR; + if (mixer_device) + card = mixer_device; + + //allocate simple id + snd_mixer_selem_id_alloca(&sid); + + //sets simple-mixer index and name + snd_mixer_selem_id_set_index(sid, mix_index); + snd_mixer_selem_id_set_name(sid, mix_name); + + if (mixer_channel) { + free(mix_name); + 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; + } + + 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; + } + + 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; + } + + 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_get_playback_volume_range(elem, &pmin, &pmax); + f_multi = (100 / (float)(pmax - pmin)); + + switch (cmd) { + case AOCONTROL_SET_VOLUME: { + ao_control_vol_t *vol = 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; + } + 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; + } + mp_msg(MSGT_AO, MSGL_DBG2, + "right=%li, pmin=%li, pmax=%li, mult=%f\n", + set_vol, pmin, pmax, + f_multi); + break; + } + case AOCONTROL_GET_VOLUME: { + ao_control_vol_t *vol = arg; + 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); + vol->right = (get_vol - pmin) * f_multi; + mp_msg(MSGT_AO, MSGL_DBG2, "left=%f, right=%f\n", vol->left, + vol->right); + break; + } + case AOCONTROL_SET_MUTE: { + bool *mute = arg; + if (!snd_mixer_selem_has_playback_switch(elem)) + goto mixer_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_LEFT, + !*mute); + break; + } + case AOCONTROL_GET_MUTE: { + bool *mute = arg; + if (!snd_mixer_selem_has_playback_switch(elem)) + goto mixer_error; + int tmp = 1; + 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); + *mute &= !tmp; + } + break; + } + } + snd_mixer_close(handle); + return CONTROL_OK; +mixer_error: + snd_mixer_close(handle); + return CONTROL_ERROR; } - } //end switch - return CONTROL_UNKNOWN; + } //end switch + return CONTROL_UNKNOWN; } -static void parse_device (char *dest, const char *src, int len) +static void parse_device(char *dest, const char *src, int len) { - char *tmp; - memmove(dest, src, len); - dest[len] = 0; - while ((tmp = strrchr(dest, '.'))) - tmp[0] = ','; - while ((tmp = strrchr(dest, '='))) - tmp[0] = ':'; + char *tmp; + memmove(dest, src, len); + dest[len] = 0; + while ((tmp = strrchr(dest, '.'))) + tmp[0] = ','; + while ((tmp = strrchr(dest, '='))) + tmp[0] = ':'; } -static void print_help (void) +static void print_help(void) { - mp_tmsg (MSGT_AO, MSGL_FATAL, - "\n[AO_ALSA] -ao alsa commandline help:\n"\ - "[AO_ALSA] Example: mpv -ao alsa:device=hw=0.3\n"\ - "[AO_ALSA] Sets first card fourth hardware device.\n\n"\ - "[AO_ALSA] Options:\n"\ - "[AO_ALSA] noblock\n"\ - "[AO_ALSA] Opens device in non-blocking mode.\n"\ - "[AO_ALSA] device=\n"\ - "[AO_ALSA] Sets device (change , to . and : to =)\n"); + mp_tmsg(MSGT_AO, MSGL_FATAL, + "\n[AO_ALSA] -ao alsa commandline help:\n" \ + "[AO_ALSA] Example: mpv -ao alsa:device=hw=0.3\n" \ + "[AO_ALSA] Sets first card fourth hardware device.\n\n" \ + "[AO_ALSA] Options:\n" \ + "[AO_ALSA] noblock\n" \ + "[AO_ALSA] Opens device in non-blocking mode.\n" \ + "[AO_ALSA] device=\n" \ + "[AO_ALSA] Sets device (change , to . and : to =)\n"); } -static int str_maxlen(void *strp) { - strarg_t *str = strp; - return str->len <= ALSA_DEVICE_SIZE; +static int str_maxlen(void *strp) +{ + strarg_t *str = strp; + return str->len <= ALSA_DEVICE_SIZE; } static int try_open_device(const char *device, int open_mode, int try_ac3) { - int err, len; - char *ac3_device, *args; - - if (try_ac3) { - /* to set the non-audio bit, use AES0=6 */ - len = strlen(device); - ac3_device = malloc(len + 7 + 1); - if (!ac3_device) - return -ENOMEM; - strcpy(ac3_device, device); - args = strchr(ac3_device, ':'); - if (!args) { - /* no existing parameters: add it behind device name */ - strcat(ac3_device, ":AES0=6"); - } else { - do - ++args; - while (isspace(*args)); - if (*args == '\0') { - /* ":" but no parameters */ - strcat(ac3_device, "AES0=6"); - } else if (*args != '{') { - /* a simple list of parameters: add it at the end of the list */ - strcat(ac3_device, ",AES0=6"); - } else { - /* parameters in config syntax: add it inside the { } block */ - do - --len; - while (len > 0 && isspace(ac3_device[len])); - if (ac3_device[len] == '}') - strcpy(ac3_device + len, " AES0=6}"); - } + int err, len; + char *ac3_device, *args; + + if (try_ac3) { + /* to set the non-audio bit, use AES0=6 */ + len = strlen(device); + ac3_device = malloc(len + 7 + 1); + if (!ac3_device) + return -ENOMEM; + strcpy(ac3_device, device); + args = strchr(ac3_device, ':'); + if (!args) { + /* no existing parameters: add it behind device name */ + strcat(ac3_device, ":AES0=6"); + } else { + do + ++args; + while (isspace(*args)); + if (*args == '\0') { + /* ":" but no parameters */ + strcat(ac3_device, "AES0=6"); + } else if (*args != '{') { + /* a simple list of parameters: add it at the end of the list */ + strcat(ac3_device, ",AES0=6"); + } else { + /* parameters in config syntax: add it inside the { } block */ + do + --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); + free(ac3_device); + if (!err) + return 0; } - 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); } /* open & setup audio device return: 1=success 0=fail -*/ + */ static int init(int rate_hz, const struct mp_chmap *channels, int format, int flags) { @@ -342,84 +368,85 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format, snd_pcm_uframes_t bufsize; snd_pcm_uframes_t boundary; const opt_t subopts[] = { - {"block", OPT_ARG_BOOL, &block, NULL}, - {"device", OPT_ARG_STR, &device, str_maxlen}, - {NULL} + {"block", OPT_ARG_BOOL, &block, NULL}, + {"device", OPT_ARG_STR, &device, str_maxlen}, + {NULL} }; char alsa_device[ALSA_DEVICE_SIZE + 1]; // make sure alsa_device is null-terminated even when using strncpy etc. memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1); - mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz, - ao_data.channels.num, format); + mp_msg(MSGT_AO, MSGL_V, + "alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz, + ao_data.channels.num, + format); alsa_handler = NULL; - mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version()); + mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version()); prepause_frames = 0; delay_before_pause = 0; snd_lib_error_set_handler(alsa_error_handler); - switch (format) - { - case AF_FORMAT_S8: - alsa_format = SND_PCM_FORMAT_S8; - break; - case AF_FORMAT_U8: - alsa_format = SND_PCM_FORMAT_U8; - break; - case AF_FORMAT_U16_LE: - alsa_format = SND_PCM_FORMAT_U16_LE; - break; - case AF_FORMAT_U16_BE: - alsa_format = SND_PCM_FORMAT_U16_BE; - break; - case AF_FORMAT_AC3_LE: - case AF_FORMAT_S16_LE: - case AF_FORMAT_IEC61937_LE: - alsa_format = SND_PCM_FORMAT_S16_LE; - break; - case AF_FORMAT_AC3_BE: - case AF_FORMAT_S16_BE: - case AF_FORMAT_IEC61937_BE: - alsa_format = SND_PCM_FORMAT_S16_BE; - break; - case AF_FORMAT_U32_LE: - alsa_format = SND_PCM_FORMAT_U32_LE; - break; - case AF_FORMAT_U32_BE: - alsa_format = SND_PCM_FORMAT_U32_BE; - break; - case AF_FORMAT_S32_LE: - alsa_format = SND_PCM_FORMAT_S32_LE; - break; - case AF_FORMAT_S32_BE: - alsa_format = SND_PCM_FORMAT_S32_BE; - break; - case AF_FORMAT_U24_LE: - alsa_format = SND_PCM_FORMAT_U24_3LE; - break; - case AF_FORMAT_U24_BE: - alsa_format = SND_PCM_FORMAT_U24_3BE; - break; - case AF_FORMAT_S24_LE: - alsa_format = SND_PCM_FORMAT_S24_3LE; - break; - case AF_FORMAT_S24_BE: - alsa_format = SND_PCM_FORMAT_S24_3BE; - break; - case AF_FORMAT_FLOAT_LE: - alsa_format = SND_PCM_FORMAT_FLOAT_LE; - break; - case AF_FORMAT_FLOAT_BE: - alsa_format = SND_PCM_FORMAT_FLOAT_BE; - break; - - default: - alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1 - break; - } + switch (format) { + case AF_FORMAT_S8: + alsa_format = SND_PCM_FORMAT_S8; + break; + case AF_FORMAT_U8: + alsa_format = SND_PCM_FORMAT_U8; + break; + case AF_FORMAT_U16_LE: + alsa_format = SND_PCM_FORMAT_U16_LE; + break; + case AF_FORMAT_U16_BE: + alsa_format = SND_PCM_FORMAT_U16_BE; + break; + case AF_FORMAT_AC3_LE: + case AF_FORMAT_S16_LE: + case AF_FORMAT_IEC61937_LE: + alsa_format = SND_PCM_FORMAT_S16_LE; + break; + case AF_FORMAT_AC3_BE: + case AF_FORMAT_S16_BE: + case AF_FORMAT_IEC61937_BE: + alsa_format = SND_PCM_FORMAT_S16_BE; + break; + case AF_FORMAT_U32_LE: + alsa_format = SND_PCM_FORMAT_U32_LE; + break; + case AF_FORMAT_U32_BE: + alsa_format = SND_PCM_FORMAT_U32_BE; + break; + case AF_FORMAT_S32_LE: + alsa_format = SND_PCM_FORMAT_S32_LE; + break; + case AF_FORMAT_S32_BE: + alsa_format = SND_PCM_FORMAT_S32_BE; + break; + case AF_FORMAT_U24_LE: + alsa_format = SND_PCM_FORMAT_U24_3LE; + break; + case AF_FORMAT_U24_BE: + alsa_format = SND_PCM_FORMAT_U24_3BE; + break; + case AF_FORMAT_S24_LE: + alsa_format = SND_PCM_FORMAT_S24_3LE; + break; + case AF_FORMAT_S24_BE: + alsa_format = SND_PCM_FORMAT_S24_3BE; + break; + case AF_FORMAT_FLOAT_LE: + alsa_format = SND_PCM_FORMAT_FLOAT_LE; + break; + case AF_FORMAT_FLOAT_BE: + alsa_format = SND_PCM_FORMAT_FLOAT_BE; + break; + + default: + alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1 + break; + } //subdevice parsing // set defaults @@ -431,45 +458,47 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format, * 'iec958' */ if (AF_FORMAT_IS_IEC61937(format)) { - device.str = "iec958"; - mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", ao_data.channels.num); - } - else + device.str = "iec958"; + mp_msg(MSGT_AO, MSGL_V, + "alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", + ao_data.channels.num); + } else /* in any case for multichannel playback we should select * appropriate device */ switch (ao_data.channels.num) { - case 1: - case 2: - device.str = "default"; - mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n"); - break; - case 4: - if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) - // hack - use the converter plugin - device.str = "plug:surround40"; - else - device.str = "surround40"; - mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n"); - break; - case 6: - if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) - device.str = "plug:surround51"; - else - device.str = "surround51"; - mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n"); - break; - case 8: - if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) - device.str = "plug:surround71"; - else - device.str = "surround71"; - mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n"); - break; - default: - device.str = "default"; - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n", - ao_data.channels.num); + case 1: + case 2: + device.str = "default"; + mp_msg(MSGT_AO, MSGL_V, "alsa-init: setup for 1/2 channel(s)\n"); + break; + case 4: + if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) + // hack - use the converter plugin + device.str = "plug:surround40"; + else + device.str = "surround40"; + mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround40\n"); + break; + case 6: + if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) + device.str = "plug:surround51"; + else + device.str = "surround51"; + mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround51\n"); + break; + case 8: + if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) + device.str = "plug:surround71"; + else + device.str = "surround71"; + mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround71\n"); + break; + default: + device.str = "default"; + mp_tmsg(MSGT_AO, MSGL_ERR, + "[AO_ALSA] %d channels are not supported.\n", + ao_data.channels.num); } device.len = strlen(device.str); if (subopt_parse(ao_subdevice, subopts) != 0) { @@ -478,200 +507,254 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format, } parse_device(alsa_device, device.str, device.len); - mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device); + mp_msg(MSGT_AO, MSGL_V, "alsa-init: using device %s\n", alsa_device); alsa_can_pause = 1; if (!alsa_handler) { - 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) - { - 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; - } - } - - if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) { - mp_tmsg(MSGT_AO,MSGL_ERR,"[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; - - snd_pcm_hw_params_alloca(&alsa_hwparams); - 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_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; - } - - /* 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)); - alsa_format = SND_PCM_FORMAT_S16_LE; - if (AF_FORMAT_IS_AC3(ao_data.format)) - ao_data.format = AF_FORMAT_AC3_LE; - else if (AF_FORMAT_IS_IEC61937(ao_data.format)) - ao_data.format = AF_FORMAT_IEC61937_LE; - else - 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; - } - - 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; - } - 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; - } - - 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; - } - - 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; - } - - 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; - } - - /* 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 - - - // 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); - } - - 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.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; - } - /* 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; - } - /* 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; - } - /* 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; - } - /* end setting sw-params */ - - alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams); - - 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)); + 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) { + 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; + } + } + + if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) + mp_tmsg(MSGT_AO, MSGL_ERR, + "[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; + + snd_pcm_hw_params_alloca(&alsa_hwparams); + 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_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; + } + + /* 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)); + alsa_format = SND_PCM_FORMAT_S16_LE; + if (AF_FORMAT_IS_AC3(ao_data.format)) + ao_data.format = AF_FORMAT_AC3_LE; + else if (AF_FORMAT_IS_IEC61937(ao_data.format)) + ao_data.format = AF_FORMAT_IEC61937_LE; + else + 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; + } + + 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; + } + 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; + } + + 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; + } + + 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; + } + + 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; + } + + /* 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 + + + // 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); + } + + 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.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; + } + /* 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; + } + /* 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; + } + /* 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; + } + /* end setting sw-params */ + + alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams); + + 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)); } // end switch alsa_handler (spdif) return 1; @@ -682,25 +765,23 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format, static void uninit(int immed) { - if (alsa_handler) { - int err; + if (alsa_handler) { + int err; - 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 { - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n"); - } + 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 + mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] No handler defined!\n"); } static void audio_pause(void) @@ -709,21 +790,23 @@ 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)); + 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; } - mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n"); + mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n"); } else { if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0 || prepause_frames < 0) 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)); + 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; } } @@ -734,20 +817,24 @@ static void audio_resume(void) int err; if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) { - mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n"); - while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1); + mp_tmsg(MSGT_AO, MSGL_INFO, + "[AO_ALSA] Pcm in suspend mode, trying to resume.\n"); + while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) + 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)); + 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; } - mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n"); + 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)); + 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; } if (prepause_frames) { @@ -765,15 +852,17 @@ 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_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; + 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; } @@ -783,50 +872,55 @@ static void reset(void) returns: number of bytes played modified last at 29.06.02 by jp thanxs for marius for giving us the light ;) -*/ + */ -static int play(void* data, int len, int flags) +static int play(void *data, int len, int flags) { - int num_frames; - snd_pcm_sframes_t res = 0; - if (!(flags & AOPLAY_FINAL_CHUNK)) - len = len / ao_data.outburst * ao_data.outburst; - num_frames = len / bytes_per_sample; - - //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len); - - if (!alsa_handler) { - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Device configuration error."); - return 0; - } - - if (num_frames == 0) - return 0; - - do { - res = snd_pcm_writei(alsa_handler, data, num_frames); - - if (res == -EINTR) { - /* nothing to do */ - res = 0; - } - else if (res == -ESTRPIPE) { /* suspend */ - mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n"); - while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN) - sleep(1); - } - if (res < 0) { - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", 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 = 0; - } - } while (res == 0); - - return res < 0 ? 0 : res * bytes_per_sample; + int num_frames; + snd_pcm_sframes_t res = 0; + if (!(flags & AOPLAY_FINAL_CHUNK)) + len = len / ao_data.outburst * ao_data.outburst; + num_frames = len / bytes_per_sample; + + //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len); + + if (!alsa_handler) { + mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Device configuration error."); + return 0; + } + + if (num_frames == 0) + return 0; + + do { + res = snd_pcm_writei(alsa_handler, data, num_frames); + + if (res == -EINTR) { + /* nothing to do */ + res = 0; + } else if (res == -ESTRPIPE) { /* suspend */ + mp_tmsg(MSGT_AO, MSGL_INFO, + "[AO_ALSA] Pcm in suspend mode, trying to resume.\n"); + while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN) + sleep(1); + } + if (res < 0) { + mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Write error: %s\n", + 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 = 0; + } + } while (res == 0); + + return res < 0 ? 0 : res * bytes_per_sample; } /* how many byes are free in the buffer */ @@ -837,10 +931,11 @@ static int get_space(void) 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; + 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; } unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample; @@ -852,22 +947,21 @@ static int get_space(void) /* delay in seconds between first and last sample in buffer */ static float get_delay(void) { - if (alsa_handler) { - snd_pcm_sframes_t delay; + if (alsa_handler) { + snd_pcm_sframes_t delay; - if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_PAUSED) - return delay_before_pause; + if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_PAUSED) + return delay_before_pause; - if (snd_pcm_delay(alsa_handler, &delay) < 0) - return 0; + if (snd_pcm_delay(alsa_handler, &delay) < 0) + return 0; - if (delay < 0) { - /* underrun - move the application pointer forward to catch up */ - snd_pcm_forward(alsa_handler, -delay); - delay = 0; - } - return (float)delay / (float)ao_data.samplerate; - } else { - return 0; - } + if (delay < 0) { + /* underrun - move the application pointer forward to catch up */ + snd_pcm_forward(alsa_handler, -delay); + delay = 0; + } + return (float)delay / (float)ao_data.samplerate; + } else + return 0; } -- cgit v1.2.3