From 9ab501443c37888ee0d286897ebb985b2056ba49 Mon Sep 17 00:00:00 2001 From: Uoti Urpala Date: Mon, 27 Feb 2012 16:29:21 +0200 Subject: configure, ao_alsa: drop support for obsolete ALSA versions Drop compatibility code for ALSA versions prior to 1.0.9. Change the configure check to use pkg-config only. --- Makefile | 7 +- configure | 107 ++------------- libao2/ao_alsa.c | 21 --- libao2/ao_alsa5.c | 383 ----------------------------------------------------- libao2/audio_out.c | 4 - stream/ai_alsa.c | 184 ------------------------- 6 files changed, 11 insertions(+), 695 deletions(-) delete mode 100644 libao2/ao_alsa5.c delete mode 100644 stream/ai_alsa.c diff --git a/Makefile b/Makefile index 55d69f9fa1..1b2d144091 100644 --- a/Makefile +++ b/Makefile @@ -23,8 +23,7 @@ include config.mak ###### variable declarations ####### -SRCS_AUDIO_INPUT-$(ALSA1X) += stream/ai_alsa1x.c -SRCS_AUDIO_INPUT-$(ALSA9) += stream/ai_alsa.c +SRCS_AUDIO_INPUT-$(ALSA) += stream/ai_alsa1x.c SRCS_AUDIO_INPUT-$(OSS) += stream/ai_oss.c SRCS_COMMON-$(AUDIO_INPUT) += $(SRCS_AUDIO_INPUT-yes) SRCS_COMMON-$(BITMAP_FONT) += sub/font_load.c @@ -431,9 +430,7 @@ SRCS_COMMON = asxparser.c \ SRCS_MPLAYER-$(3DFX) += libvo/vo_3dfx.c SRCS_MPLAYER-$(AA) += libvo/vo_aa.c -SRCS_MPLAYER-$(ALSA1X) += libao2/ao_alsa.c -SRCS_MPLAYER-$(ALSA5) += libao2/ao_alsa5.c -SRCS_MPLAYER-$(ALSA9) += libao2/ao_alsa.c +SRCS_MPLAYER-$(ALSA) += libao2/ao_alsa.c SRCS_MPLAYER-$(APPLE_IR) += input/appleir.c SRCS_MPLAYER-$(APPLE_REMOTE) += input/ar.c SRCS_MPLAYER-$(ARTS) += libao2/ao_arts.c diff --git a/configure b/configure index df7058aa73..ad4e1f9b46 100755 --- a/configure +++ b/configure @@ -4950,100 +4950,18 @@ if test "$_alloca" != yes ; then _alsa=no res_comment="alloca missing" fi -if test "$_alsa" != no ; then - _alsa=no - cat > $TMPC << EOF -#include -#if !((SND_LIB_MAJOR == 0) && (SND_LIB_MINOR == 5)) -#error "alsa version != 0.5.x" -#endif -int main(void) { return 0; } -EOF - cc_check -lasound $_ld_dl $_ld_pthread && _alsaver='0.5.x' - - cat > $TMPC << EOF -#include -#if !((SND_LIB_MAJOR == 0) && (SND_LIB_MINOR == 9)) -#error "alsa version != 0.9.x" -#endif -int main(void) { return 0; } -EOF - cc_check -lasound $_ld_dl $_ld_pthread && _alsaver='0.9.x-sys' - cat > $TMPC << EOF -#include -#if !((SND_LIB_MAJOR == 0) && (SND_LIB_MINOR == 9)) -#error "alsa version != 0.9.x" -#endif -int main(void) { return 0; } -EOF - cc_check -lasound $_ld_dl $_ld_pthread && _alsaver='0.9.x-alsa' - - cat > $TMPC << EOF -#include -#if !((SND_LIB_MAJOR == 1) && (SND_LIB_MINOR == 0)) -#error "alsa version != 1.0.x" -#endif -int main(void) { return 0; } -EOF - cc_check -lasound $_ld_dl $_ld_pthread && _alsaver='1.0.x-sys' - cat > $TMPC << EOF -#include -#if !((SND_LIB_MAJOR == 1) && (SND_LIB_MINOR == 0)) -#error "alsa version != 1.0.x" -#endif -int main(void) { return 0; } -EOF - cc_check -lasound $_ld_dl $_ld_pthread && _alsaver='1.0.x-alsa' +if test "$_alsa" = auto ; then + _alsa=no + if pkg_config_add "alsa >= 1.0.9" ; then + _alsa=yes + fi fi def_alsa='#undef CONFIG_ALSA' -def_alsa5='#undef CONFIG_ALSA5' -def_alsa9='#undef CONFIG_ALSA9' -def_alsa1x='#undef CONFIG_ALSA1X' -def_sys_asoundlib_h='#undef HAVE_SYS_ASOUNDLIB_H' -def_alsa_asoundlib_h='#undef HAVE_ALSA_ASOUNDLIB_H' -if test "$_alsaver" ; then - _alsa=yes - if test "$_alsaver" = '0.5.x' ; then - _alsa5=yes - aomodules="alsa5 $aomodules" - def_alsa5='#define CONFIG_ALSA5 1' - def_sys_asoundlib_h='#define HAVE_SYS_ASOUNDLIB_H 1' - res_comment="using alsa 0.5.x and sys/asoundlib.h" - elif test "$_alsaver" = '0.9.x-sys' ; then - _alsa9=yes +if test "$_alsa" = yes ; then aomodules="alsa $aomodules" def_alsa='#define CONFIG_ALSA 1' - def_alsa9='#define CONFIG_ALSA9 1' - def_sys_asoundlib_h='#define HAVE_SYS_ASOUNDLIB_H 1' - res_comment="using alsa 0.9.x and sys/asoundlib.h" - elif test "$_alsaver" = '0.9.x-alsa' ; then - _alsa9=yes - aomodules="alsa $aomodules" - def_alsa='#define CONFIG_ALSA 1' - def_alsa9='#define CONFIG_ALSA9 1' - def_alsa_asoundlib_h='#define HAVE_ALSA_ASOUNDLIB_H 1' - res_comment="using alsa 0.9.x and alsa/asoundlib.h" - elif test "$_alsaver" = '1.0.x-sys' ; then - _alsa1x=yes - aomodules="alsa $aomodules" - def_alsa='#define CONFIG_ALSA 1' - def_alsa1x="#define CONFIG_ALSA1X 1" - def_alsa_asoundlib_h='#define HAVE_SYS_ASOUNDLIB_H 1' - res_comment="using alsa 1.0.x and sys/asoundlib.h" - elif test "$_alsaver" = '1.0.x-alsa' ; then - _alsa1x=yes - aomodules="alsa $aomodules" - def_alsa='#define CONFIG_ALSA 1' - def_alsa1x="#define CONFIG_ALSA1X 1" - def_alsa_asoundlib_h='#define HAVE_ALSA_ASOUNDLIB_H 1' - res_comment="using alsa 1.0.x and alsa/asoundlib.h" - else - _alsa=no - res_comment="unknown version" - fi - extra_ldflags="$extra_ldflags -lasound $_ld_dl $_ld_pthread" else - noaomodules="alsa $noaomodules" + noaomodules="alsa $noaomodules" fi echores "$_alsa" @@ -6115,7 +6033,7 @@ echocheck "Radio interface" if test "$_radio" = yes ; then def_radio='#define CONFIG_RADIO 1' inputmodules="radio $inputmodules" - if test "$_alsa9" != yes -a "$_alsa1x" != yes -a "$_ossaudio" != yes ; then + if test "$_alsa" != yes -a "$_ossaudio" != yes ; then _radio_capture=no fi if test "$_radio_capture" = yes ; then @@ -6508,9 +6426,7 @@ NEED_VSSCANF = $need_vsscanf # features 3DFX = $_3dfx AA = $_aa -ALSA1X = $_alsa1x -ALSA9 = $_alsa9 -ALSA5 = $_alsa5 +ALSA = $_alsa APPLE_IR = $_apple_ir APPLE_REMOTE = $_apple_remote ARTS = $_arts @@ -6720,13 +6636,11 @@ $def_dvdcss /* system headers */ $def_alloca_h -$def_alsa_asoundlib_h $def_altivec_h $def_malloc_h $def_mman_h $def_mman_has_map_failed $def_soundcard_h -$def_sys_asoundlib_h $def_sys_soundcard_h $def_sys_sysinfo_h $def_sys_videoio_h @@ -6853,9 +6767,6 @@ $def_xmms /* Audio output drivers */ $def_alsa -$def_alsa1x -$def_alsa5 -$def_alsa9 $def_arts $def_coreaudio $def_dart diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c index 1806a36d56..8ce80cdfbb 100644 --- a/libao2/ao_alsa.c +++ b/libao2/ao_alsa.c @@ -43,14 +43,7 @@ #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API -#ifdef HAVE_SYS_ASOUNDLIB_H -#include -#elif defined(HAVE_ALSA_ASOUNDLIB_H) #include -#else -#error "asoundlib.h is not in sys/ or alsa/ - please bugreport" -#endif - #include "audio_out.h" #include "audio_out_internal.h" @@ -342,11 +335,7 @@ static int init(int rate_hz, int channels, int format, int flags) mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz, channels, format); alsa_handler = NULL; -#if SND_LIB_VERSION >= 0x010005 mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version()); -#else - mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR); -#endif prepause_frames = 0; @@ -554,7 +543,6 @@ static int init(int rate_hz, int channels, int format, int flags) /* 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 SND_LIB_VERSION >= 0x010009 if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams, 0)) < 0) { @@ -562,7 +550,6 @@ static int init(int rate_hz, int channels, int format, int flags) snd_strerror(err)); return 0; } -#endif if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, &ao_data.samplerate, NULL)) < 0) @@ -626,15 +613,11 @@ static int init(int rate_hz, int channels, int format, int flags) snd_strerror(err)); return 0; } -#if SND_LIB_VERSION >= 0x000901 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; } -#else - boundary = 0x7fffffff; -#endif /* 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", @@ -647,14 +630,12 @@ static int init(int rate_hz, int channels, int format, int flags) snd_strerror(err)); return 0; } -#if SND_LIB_VERSION >= 0x000901 /* 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; } -#endif 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)); @@ -851,9 +832,7 @@ static float get_delay(void) if (delay < 0) { /* underrun - move the application pointer forward to catch up */ -#if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */ snd_pcm_forward(alsa_handler, -delay); -#endif delay = 0; } return (float)delay / (float)ao_data.samplerate; diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c deleted file mode 100644 index d6ab7eeb22..0000000000 --- a/libao2/ao_alsa5.c +++ /dev/null @@ -1,383 +0,0 @@ -/* - * ALSA 0.5.x audio output driver - * - * Copyright (C) 2001 Alex Beregszaszi - * - * Thanks to Arpi for helping me ;) - * - * 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 "config.h" - -#include "audio_out.h" -#include "audio_out_internal.h" -#include "libaf/af_format.h" - -#include "mp_msg.h" - -static const ao_info_t info = -{ - "ALSA-0.5.x audio output", - "alsa5", - "Alex Beregszaszi", - "" -}; - -LIBAO_EXTERN(alsa5) - -static snd_pcm_t *alsa_handler; -static snd_pcm_format_t alsa_format; -static int alsa_rate = SND_PCM_RATE_CONTINUOUS; - -/* to set/get/query special features/parameters */ -static int control(int cmd, void *arg) -{ - return CONTROL_UNKNOWN; -} - -/* - open & setup audio device - return: 1=success 0=fail -*/ -static int init(int rate_hz, int channels, int format, int flags) -{ - int err; - int cards = -1; - snd_pcm_channel_params_t params; - snd_pcm_channel_setup_t setup; - snd_pcm_info_t info; - snd_pcm_channel_info_t chninfo; - - mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ALSA5] alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz, - channels, af_fmt2str_short(format)); - - alsa_handler = NULL; - - mp_msg(MSGT_AO, MSGL_V, "alsa-init: compiled for ALSA-%s (%d)\n", SND_LIB_VERSION_STR, - SND_LIB_VERSION); - - if ((cards = snd_cards()) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: no soundcards found.\n"); - return 0; - } - - ao_data.format = format; - ao_data.channels = channels; - ao_data.samplerate = rate_hz; - ao_data.bps = ao_data.samplerate*ao_data.channels; - ao_data.outburst = OUTBURST; - ao_data.buffersize = 16384; - - memset(&alsa_format, 0, sizeof(alsa_format)); - switch (format) - { - case AF_FORMAT_S8: - alsa_format.format = SND_PCM_SFMT_S8; - break; - case AF_FORMAT_U8: - alsa_format.format = SND_PCM_SFMT_U8; - break; - case AF_FORMAT_U16_LE: - alsa_format.format = SND_PCM_SFMT_U16_LE; - break; - case AF_FORMAT_U16_BE: - alsa_format.format = SND_PCM_SFMT_U16_BE; - break; - case AF_FORMAT_AC3_LE: - case AF_FORMAT_S16_LE: - alsa_format.format = SND_PCM_SFMT_S16_LE; - break; - case AF_FORMAT_AC3_BE: - case AF_FORMAT_S16_BE: - alsa_format.format = SND_PCM_SFMT_S16_BE; - break; - default: - alsa_format.format = SND_PCM_SFMT_MPEG; - break; - } - - switch(alsa_format.format) - { - case SND_PCM_SFMT_S16_LE: - case SND_PCM_SFMT_U16_LE: - ao_data.bps *= 2; - break; - case -1: - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: invalid format (%s) requested - output disabled.\n",af_fmt2str_short(format)); - return 0; - default: - break; - } - - switch(rate_hz) - { - case 8000: - alsa_rate = SND_PCM_RATE_8000; - break; - case 11025: - alsa_rate = SND_PCM_RATE_11025; - break; - case 16000: - alsa_rate = SND_PCM_RATE_16000; - break; - case 22050: - alsa_rate = SND_PCM_RATE_22050; - break; - case 32000: - alsa_rate = SND_PCM_RATE_32000; - break; - case 44100: - alsa_rate = SND_PCM_RATE_44100; - break; - case 48000: - alsa_rate = SND_PCM_RATE_48000; - break; - case 88200: - alsa_rate = SND_PCM_RATE_88200; - break; - case 96000: - alsa_rate = SND_PCM_RATE_96000; - break; - case 176400: - alsa_rate = SND_PCM_RATE_176400; - break; - case 192000: - alsa_rate = SND_PCM_RATE_192000; - break; - default: - alsa_rate = SND_PCM_RATE_CONTINUOUS; - break; - } - - alsa_format.rate = ao_data.samplerate; - alsa_format.voices = ao_data.channels; - alsa_format.interleave = 1; - - if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: playback open error: %s\n", snd_strerror(err)); - return 0; - } - - if ((err = snd_pcm_info(alsa_handler, &info)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: PCM info error: %s\n", snd_strerror(err)); - return 0; - } - - mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ALSA5] alsa-init: %d soundcard(s) found, using: %s\n", - cards, info.name); - - if (info.flags & SND_PCM_INFO_PLAYBACK) - { - memset(&chninfo, 0, sizeof(chninfo)); - chninfo.channel = SND_PCM_CHANNEL_PLAYBACK; - if ((err = snd_pcm_channel_info(alsa_handler, &chninfo)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: PCM channel info error: %s\n", snd_strerror(err)); - return 0; - } - -#ifndef __QNX__ - if (chninfo.buffer_size) - ao_data.buffersize = chninfo.buffer_size; -#endif - - mp_msg(MSGT_AO, MSGL_V, "alsa-init: setting preferred buffer size from driver: %d bytes\n", - ao_data.buffersize); - } - - memset(¶ms, 0, sizeof(params)); - params.channel = SND_PCM_CHANNEL_PLAYBACK; - params.mode = SND_PCM_MODE_STREAM; - params.format = alsa_format; - params.start_mode = SND_PCM_START_DATA; - params.stop_mode = SND_PCM_STOP_ROLLOVER; - params.buf.stream.queue_size = ao_data.buffersize; - params.buf.stream.fill = SND_PCM_FILL_NONE; - - if ((err = snd_pcm_channel_params(alsa_handler, ¶ms)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: error setting parameters: %s\n", snd_strerror(err)); - return 0; - } - - memset(&setup, 0, sizeof(setup)); - setup.channel = SND_PCM_CHANNEL_PLAYBACK; - setup.mode = SND_PCM_MODE_STREAM; - setup.format = alsa_format; - setup.buf.stream.queue_size = ao_data.buffersize; - setup.msbits_per_sample = ao_data.bps; - - if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: error setting up channel: %s\n", snd_strerror(err)); - return 0; - } - - if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: channel prepare error: %s\n", snd_strerror(err)); - return 0; - } - - mp_msg(MSGT_AO, MSGL_INFO, "AUDIO: %d Hz/%d channels/%d bps/%d bytes buffer/%s\n", - ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize, - snd_pcm_get_format_name(alsa_format.format)); - return 1; -} - -/* close audio device */ -static void uninit(int immed) -{ - int err; - - if ((err = snd_pcm_playback_drain(alsa_handler)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-uninit: playback drain error: %s\n", snd_strerror(err)); - return; - } - - if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-uninit: playback flush error: %s\n", snd_strerror(err)); - return; - } - - if ((err = snd_pcm_close(alsa_handler)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-uninit: PCM close error: %s\n", snd_strerror(err)); - return; - } -} - -/* stop playing and empty buffers (for seeking/pause) */ -static void reset(void) -{ - int err; - - if ((err = snd_pcm_playback_drain(alsa_handler)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-reset: playback drain error: %s\n", snd_strerror(err)); - return; - } - - if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-reset: playback flush error: %s\n", snd_strerror(err)); - return; - } - - if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-reset: channel prepare error: %s\n", snd_strerror(err)); - return; - } -} - -/* stop playing, keep buffers (for pause) */ -static void audio_pause(void) -{ - int err; - - if ((err = snd_pcm_playback_drain(alsa_handler)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-pause: playback drain error: %s\n", snd_strerror(err)); - return; - } - - if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-pause: playback flush error: %s\n", snd_strerror(err)); - return; - } -} - -/* resume playing, after audio_pause() */ -static void audio_resume(void) -{ - int err; - if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-resume: channel prepare error: %s\n", snd_strerror(err)); - return; - } -} - -/* - plays 'len' bytes of 'data' - returns: number of bytes played -*/ -static int play(void* data, int len, int flags) -{ - int got_len; - - if (!len) - return 0; - - if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0) - { - if (got_len == -EPIPE) /* underrun? */ - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-play: alsa underrun, resetting stream.\n"); - if ((got_len = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-play: playback prepare error: %s\n", snd_strerror(got_len)); - return 0; - } - if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0) - { - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-play: write error after reset: %s - giving up.\n", - snd_strerror(got_len)); - return 0; - } - return got_len; /* 2nd write was ok */ - } - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-play: output error: %s\n", snd_strerror(got_len)); - return 0; - } - return got_len; -} - -/* how many byes are free in the buffer */ -static int get_space(void) -{ - snd_pcm_channel_status_t ch_stat; - - ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK; - - if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0) - return 0; /* error occurred */ - else - return ch_stat.free; -} - -/* delay in seconds between first and last sample in buffer */ -static float get_delay(void) -{ - snd_pcm_channel_status_t ch_stat; - - ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK; - - if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0) - return (float)ao_data.buffersize/(float)ao_data.bps; /* error occurred */ - else - return (float)ch_stat.count/(float)ao_data.bps; -} diff --git a/libao2/audio_out.c b/libao2/audio_out.c index b1b5c0e901..a91a0d6d72 100644 --- a/libao2/audio_out.c +++ b/libao2/audio_out.c @@ -41,7 +41,6 @@ extern const struct ao_driver audio_out_pulse; extern const struct ao_driver audio_out_jack; extern const struct ao_driver audio_out_openal; extern const struct ao_driver audio_out_null; -extern const struct ao_driver audio_out_alsa5; extern const struct ao_driver audio_out_alsa; extern const struct ao_driver audio_out_nas; extern const struct ao_driver audio_out_sdl; @@ -83,9 +82,6 @@ static const struct ao_driver * const audio_out_drivers[] = { #ifdef CONFIG_OSS_AUDIO &audio_out_oss, #endif -#ifdef CONFIG_ALSA5 - &audio_out_alsa5, -#endif #ifdef CONFIG_SGI_AUDIO &audio_out_sgi, #endif diff --git a/stream/ai_alsa.c b/stream/ai_alsa.c deleted file mode 100644 index c5e2f3f623..0000000000 --- a/stream/ai_alsa.c +++ /dev/null @@ -1,184 +0,0 @@ -/* - * 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 "config.h" - -#include -#include "audio_in.h" -#include "mp_msg.h" - -int ai_alsa_setup(audio_in_t *ai) -{ - snd_pcm_hw_params_t *params; - snd_pcm_sw_params_t *swparams; - int buffer_size; - int err; - unsigned int rate; - - snd_pcm_hw_params_alloca(¶ms); - snd_pcm_sw_params_alloca(&swparams); - - err = snd_pcm_hw_params_any(ai->alsa.handle, params); - if (err < 0) { - mp_tmsg(MSGT_TV, MSGL_ERR, "Broken configuration for this PCM: no configurations available.\n"); - return -1; - } - err = snd_pcm_hw_params_set_access(ai->alsa.handle, params, - SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) { - mp_tmsg(MSGT_TV, MSGL_ERR, "Access type not available.\n"); - return -1; - } - err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE); - if (err < 0) { - mp_tmsg(MSGT_TV, MSGL_ERR, "Sample format not available.\n"); - return -1; - } - err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels); - if (err < 0) { - ai->channels = snd_pcm_hw_params_get_channels(params); - mp_tmsg(MSGT_TV, MSGL_ERR, "Channel count not available - reverting to default: %d\n", - ai->channels); - } else { - ai->channels = ai->req_channels; - } - - err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, ai->req_samplerate, 0); - assert(err >= 0); - rate = err; - ai->samplerate = rate; - - ai->alsa.buffer_time = 1000000; - ai->alsa.buffer_time = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params, - ai->alsa.buffer_time, 0); - assert(ai->alsa.buffer_time >= 0); - ai->alsa.period_time = ai->alsa.buffer_time / 4; - ai->alsa.period_time = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params, - ai->alsa.period_time, 0); - assert(ai->alsa.period_time >= 0); - err = snd_pcm_hw_params(ai->alsa.handle, params); - if (err < 0) { - mp_tmsg(MSGT_TV, MSGL_ERR, "Unable to install hardware parameters: %s"); - snd_pcm_hw_params_dump(params, ai->alsa.log); - return -1; - } - ai->alsa.chunk_size = snd_pcm_hw_params_get_period_size(params, 0); - buffer_size = snd_pcm_hw_params_get_buffer_size(params); - if (ai->alsa.chunk_size == buffer_size) { - mp_tmsg(MSGT_TV, MSGL_ERR, "Can't use period equal to buffer size (%u == %lu)\n", ai->alsa.chunk_size, (long)buffer_size); - return -1; - } - snd_pcm_sw_params_current(ai->alsa.handle, swparams); - err = snd_pcm_sw_params_set_sleep_min(ai->alsa.handle, swparams,0); - assert(err >= 0); - err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size); - assert(err >= 0); - - err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0); - assert(err >= 0); - err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size); - assert(err >= 0); - - assert(err >= 0); - if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) { - mp_tmsg(MSGT_TV, MSGL_ERR, "Unable to install software parameters:\n"); - snd_pcm_sw_params_dump(swparams, ai->alsa.log); - return -1; - } - - if (mp_msg_test(MSGT_TV, MSGL_V)) { - snd_pcm_dump(ai->alsa.handle, ai->alsa.log); - } - - ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE); - ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels; - ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8; - ai->samplesize = ai->alsa.bits_per_sample; - ai->bytes_per_sample = ai->alsa.bits_per_sample/8; - - return 0; -} - -int ai_alsa_init(audio_in_t *ai) -{ - int err; - - err = snd_pcm_open(&ai->alsa.handle, ai->alsa.device, SND_PCM_STREAM_CAPTURE, 0); - if (err < 0) { - mp_tmsg(MSGT_TV, MSGL_ERR, "Error opening audio: %s\n", snd_strerror(err)); - return -1; - } - - err = snd_output_stdio_attach(&ai->alsa.log, stderr, 0); - - if (err < 0) { - return -1; - } - - err = ai_alsa_setup(ai); - - return err; -} - -#ifndef timersub -#define timersub(a, b, result) \ -do { \ - (result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \ - (result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \ - if ((result)->tv_usec < 0) { \ - --(result)->tv_sec; \ - (result)->tv_usec += 1000000; \ - } \ -} while (0) -#endif - -int ai_alsa_xrun(audio_in_t *ai) -{ - snd_pcm_status_t *status; - int res; - - snd_pcm_status_alloca(&status); - if ((res = snd_pcm_status(ai->alsa.handle, status))<0) { - mp_tmsg(MSGT_TV, MSGL_ERR, "ALSA status error: %s", snd_strerror(res)); - return -1; - } - if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) { - struct timeval now, diff, tstamp; - gettimeofday(&now, 0); - snd_pcm_status_get_trigger_tstamp(status, &tstamp); - timersub(&now, &tstamp, &diff); - mp_tmsg(MSGT_TV, MSGL_ERR, "ALSA xrun!!! (at least %.3f ms long)\n", - diff.tv_sec * 1000 + diff.tv_usec / 1000.0); - if (mp_msg_test(MSGT_TV, MSGL_V)) { - mp_tmsg(MSGT_TV, MSGL_ERR, "ALSA Status:\n"); - snd_pcm_status_dump(status, ai->alsa.log); - } - if ((res = snd_pcm_prepare(ai->alsa.handle))<0) { - mp_tmsg(MSGT_TV, MSGL_ERR, "ALSA xrun: prepare error: %s", snd_strerror(res)); - return -1; - } - return 0; /* ok, data should be accepted again */ - } - mp_tmsg(MSGT_TV, MSGL_ERR, "ALSA read/write error"); - return -1; -} -- cgit v1.2.3