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author | Uoti Urpala <uau@mplayer2.org> | 2012-02-27 16:29:21 +0200 |
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committer | Uoti Urpala <uau@mplayer2.org> | 2012-02-27 16:46:56 +0200 |
commit | 9ab501443c37888ee0d286897ebb985b2056ba49 (patch) | |
tree | d300ec816d1154a7fe8f6da1890ff25b4a94b6ef /libao2/ao_alsa.c | |
parent | 8d20859716db53fb6a2aa4975c8172a9d6aa7a3d (diff) | |
download | mpv-9ab501443c37888ee0d286897ebb985b2056ba49.tar.bz2 mpv-9ab501443c37888ee0d286897ebb985b2056ba49.tar.xz |
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.
Diffstat (limited to 'libao2/ao_alsa.c')
-rw-r--r-- | libao2/ao_alsa.c | 21 |
1 files changed, 0 insertions, 21 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; |