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.

3 files changed, 0 insertions, 408 deletions

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 <sys/asoundlib.h>
-#elif defined(HAVE_ALSA_ASOUNDLIB_H)
 #include <alsa/asoundlib.h>
-#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 <errno.h>
-#include <sys/asoundlib.h>
-
-#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(&params, 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, &params)) < 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