From d4bdd0473d6f43132257c9fb3848d829755167a3 Mon Sep 17 00:00:00 2001 From: wm4 Date: Mon, 5 Nov 2012 17:02:04 +0100 Subject: Rename directories, move files (step 1 of 2) (does not compile) Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code. --- libao2/ao_alsa.c | 868 ------------------------------------------------------- 1 file changed, 868 deletions(-) delete mode 100644 libao2/ao_alsa.c (limited to 'libao2/ao_alsa.c') diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c deleted file mode 100644 index 27119112cb..0000000000 --- a/libao2/ao_alsa.c +++ /dev/null @@ -1,868 +0,0 @@ -/* - * ALSA 0.9.x-1.x audio output driver - * - * Copyright (C) 2004 Alex Beregszaszi - * - * modified for real ALSA 0.9.0 support by Zsolt Barat - * additional AC-3 passthrough support by Andy Lo A Foe - * 08/22/2002 iec958-init rewritten and merged with common init, zsolt - * 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka - * 04/25/2004 printfs converted to mp_msg, Zsolt. - * - * This file is part of MPlayer. - * - * MPlayer is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * MPlayer is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with MPlayer; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include "config.h" -#include "subopt-helper.h" -#include "mixer.h" -#include "mp_msg.h" - -#define ALSA_PCM_NEW_HW_PARAMS_API -#define ALSA_PCM_NEW_SW_PARAMS_API - -#include - -#include "audio_out.h" -#include "audio_out_internal.h" -#include "libaf/format.h" - -static const ao_info_t info = -{ - "ALSA-0.9.x-1.x audio output", - "alsa", - "Alex Beregszaszi, Zsolt Barat ", - "under development" -}; - -LIBAO_EXTERN(alsa) - -static snd_pcm_t *alsa_handler; -static snd_pcm_format_t alsa_format; - -#define BUFFER_TIME 500000 // 0.5 s -#define FRAGCOUNT 16 - -static size_t bytes_per_sample; - -static int alsa_can_pause; -static snd_pcm_sframes_t prepause_frames; - -#define ALSA_DEVICE_SIZE 256 - -static void alsa_error_handler(const char *file, int line, const char *function, - 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); -} - -/* 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: - { - 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_AC3(ao_data.format) || 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; - } - break; - } - } - snd_mixer_close(handle); - return CONTROL_OK; - mixer_error: - snd_mixer_close(handle); - return CONTROL_ERROR; - } - - } //end switch - return CONTROL_UNKNOWN; -} - -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] = ':'; -} - -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"); -} - -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}"); - } - } - 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); -} - -/* - open & setup audio device - return: 1=success 0=fail -*/ -static int init(int rate_hz, int channels, int format, int flags) -{ - int err; - int block; - strarg_t device; - snd_pcm_uframes_t chunk_size; - 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} - }; - - 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, - channels, format); - alsa_handler = NULL; - mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version()); - - prepause_frames = 0; - - snd_lib_error_set_handler(alsa_error_handler); - - ao_data.samplerate = rate_hz; - ao_data.format = format; - ao_data.channels = channels; - - 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; - case AF_FORMAT_MU_LAW: - alsa_format = SND_PCM_FORMAT_MU_LAW; - break; - case AF_FORMAT_A_LAW: - alsa_format = SND_PCM_FORMAT_A_LAW; - break; - - default: - alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1 - break; - } - - //subdevice parsing - // set defaults - block = 1; - /* switch for spdif - * sets opening sequence for SPDIF - * sets also the playback and other switches 'on the fly' - * while opening the abstract alias for the spdif subdevice - * 'iec958' - */ - if (AF_FORMAT_IS_AC3(format) || AF_FORMAT_IS_IEC61937(format)) { - device.str = "iec958"; - mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", channels); - } - else - /* in any case for multichannel playback we should select - * appropriate device - */ - switch (channels) { - 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",channels); - } - device.len = strlen(device.str); - if (subopt_parse(ao_subdevice, subopts) != 0) { - print_help(); - return 0; - } - parse_device(alsa_device, device.str, device.len); - - 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_AC3(format) || 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; - } - - if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams, - &ao_data.channels)) < 0) - { - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n", - snd_strerror(err)); - return 0; - } - - /* 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; - 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, (int)bytes_per_sample, ao_data.buffersize, - snd_pcm_format_description(alsa_format)); - - } // end switch alsa_handler (spdif) - return 1; -} // end init - - -/* close audio device */ -static void uninit(int immed) -{ - - 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"); - } -} - -static void audio_pause(void) -{ - int err; - - if (alsa_can_pause) { - 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"); - } else { - if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0 - || prepause_frames < 0) - prepause_frames = 0; - - 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; - } - } -} - -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); - } - 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; - } - 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; - } - if (prepause_frames) { - void *silence = calloc(prepause_frames, bytes_per_sample); - play(silence, prepause_frames * bytes_per_sample, 0); - free(silence); - } - } -} - -/* stop playing and empty buffers (for seeking/pause) */ -static void reset(void) -{ - int err; - - prepause_frames = 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; -} - -/* - plays 'len' bytes of 'data' - 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) -{ - 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)); - return 0; - break; - } - } - } while (res == 0); - - return res < 0 ? res : res * bytes_per_sample; -} - -/* how many byes are free in the buffer */ -static int get_space(void) -{ - snd_pcm_status_t *status; - int ret; - - 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; - } - - unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample; - if (space > ao_data.buffersize) // Buffer underrun? - space = ao_data.buffersize; - return space; -} - -/* delay in seconds between first and last sample in buffer */ -static float get_delay(void) -{ - if (alsa_handler) { - snd_pcm_sframes_t delay; - - 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; - } -} -- cgit v1.2.3