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. --- audio/out/ao_alsa.c | 868 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 868 insertions(+) create mode 100644 audio/out/ao_alsa.c (limited to 'audio/out/ao_alsa.c') diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c new file mode 100644 index 0000000000..27119112cb --- /dev/null +++ b/audio/out/ao_alsa.c @@ -0,0 +1,868 @@ +/* + * 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