mirror of https://github.com/mpv-player/mpv
mp_msg transition of unmaintained audio output drivers.
Patch by Reynaldo H. Verdejo Pinochet <reynaldo at opendot dot cl> git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@13384 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
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parent
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commit
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AUTHORS
1
AUTHORS
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@ -671,6 +671,7 @@ Vajna, Mikl
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Verdejo Pinochet, Reynaldo H. (reynaldo) <reynaldo@opendot.cl>
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* improved EDL support
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* mp_msg transition on unmantained libao2 drivers
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Wigren, Per <wigren@home.se>
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* bmovl - Bitmap Overlay video filter
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@ -781,3 +781,101 @@ static char help_text[]=
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#define MSGTR_VO_JPEG_BaselineJPEG "Baseline JPEG enabled."
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#define MSGTR_VO_JPEG_NoBaselineJPEG "Baseline JPEG disabled."
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// ======================= AO Audio Output drivers ========================
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// libao2
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// ao_oss.c
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#define MSGTR_AO_OSS_CantOpenMixer "[AO OSS] audio_setup: Can't open mixer device %s: %s\n"
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#define MSGTR_AO_OSS_ChanNotFound "[AO OSS] audio_setup: Audio card mixer does not have channel '%s' using default.\n"
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#define MSGTR_AO_OSS_CantOpenDev "[AO OSS] audio_setup: Can't open audio device %s: %s\n"
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#define MSGTR_AO_OSS_CantMakeFd "[AO OSS] audio_setup: Can't make filedescriptor blocking: %s\n"
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#define MSGTR_AO_OSS_CantSetAC3 "[AO OSS] Can't set audio device %s to AC3 output, trying S16...\n"
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#define MSGTR_AO_OSS_CantSetChans "[AO OSS] audio_setup: Failed to set audio device to %d channels.\n"
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#define MSGTR_AO_OSS_CantUseGetospace "[AO OSS] audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n"
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#define MSGTR_AO_OSS_CantUseSelect "[AO OSS]\n *** Your audio driver DOES NOT support select() ***\n Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n"
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#define MSGTR_AO_OSS_CantReopen "[AO OSS]\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n"
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// ao_arts.c
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#define MSGTR_AO_ARTS_CantInit "[AO ARTS] %s\n"
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#define MSGTR_AO_ARTS_ServerConnect "[AO ARTS] Connected to sound server.\n"
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#define MSGTR_AO_ARTS_CantOpenStream "[AO ARTS] Unable to open a stream.\n"
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#define MSGTR_AO_ARTS_StreamOpen "[AO ARTS] Stream opened.\n"
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#define MSGTR_AO_ARTS_BufferSize "[AO ARTS] buffer size: %d\n"
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// ao_dxr2.c
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#define MSGTR_AO_DXR2_SetVolFailed "[AO DXR2] Setting volume to %d failed.\n"
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#define MSGTR_AO_DXR2_UnsupSamplerate "[AO DXR2] dxr2: %d Hz not supported, try \"-aop list=resample\"\n"
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// ao_esd.c
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#define MSGTR_AO_ESD_CantOpenSound "[AO ESD] esd_open_sound failed: %s\n"
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#define MSGTR_AO_ESD_LatencyInfo "[AO ESD] latency: [server: %0.2fs, net: %0.2fs] (adjust %0.2fs)\n"
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#define MSGTR_AO_ESD_CantOpenPBStream "[AO ESD] failed to open esd playback stream: %s\n"
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// ao_mpegpes.c
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#define MSGTR_AO_MPEGPES_CantSetMixer "[AO MPEGPES] DVB audio set mixer failed: %s\n"
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#define MSGTR_AO_MPEGPES_UnsupSamplerate "[AO MPEGPES] %d Hz not supported, try to resample...\n"
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// ao_null.c
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// This one desn't even have any mp_msg nor printf's?? [CHECK]
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// ao_pcm.c
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#define MSGTR_AO_PCM_FileInfo "[AO PCM] File: %s (%s)\nPCM: Samplerate: %iHz Channels: %s Format %s\n"
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#define MSGTR_AO_PCM_HintInfo "[AO PCM] Info: fastest dumping is achieved with -vc dummy -vo null\nPCM: Info: to write WAVE files use -waveheader (default)."
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#define MSGTR_AO_PCM_CantOpenOutputFile "[AO PCM] Failed to open %s for writing!\n"
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// ao_sdl.c
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#define MSGTR_AO_SDL_INFO "[AO SDL] Samplerate: %iHz Channels: %s Format %s\n"
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#define MSGTR_AO_SDL_DriverInfo "[AO SDL] using %s audio driver.\n"
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#define MSGTR_AO_SDL_UnsupportedAudioFmt "[AO SDL] Unsupported audio format: 0x%x.\n"
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#define MSGTR_AO_SDL_CantInit "[AO SDL] Initializing of SDL Audio failed: %s\n"
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#define MSGTR_AO_SDL_CantOpenAudio "[AO SDL] Unable to open audio: %s\n"
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// ao_sgi.c
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#define MSGTR_AO_SGI_INFO "[AO SGI] control.\n"
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#define MSGTR_AO_SGI_InitInfo "[AO SGI] init: Samplerate: %iHz Channels: %s Format %s\n"
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#define MSGTR_AO_SGI_InvalidDevice "[AO SGI] play: invalid device.\n"
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#define MSGTR_AO_SGI_CantSetParms_Samplerate "[AO SGI] init: setparams failed: %s\nCould not set desired samplerate.\n"
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#define MSGTR_AO_SGI_CantSetAlRate "[AO SGI] init: AL_RATE was not accepted on the given resource.\n"
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#define MSGTR_AO_SGI_CantGetParms "[AO SGI] init: getparams failed: %s\n"
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#define MSGTR_AO_SGI_SampleRateInfo "[AO SGI] init: samplerate is now %lf (desired rate is %lf)\n"
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#define MSGTR_AO_SGI_InitConfigError "[AO SGI] init: %s\n"
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#define MSGTR_AO_SGI_InitOpenAudioFailed "[AO SGI] init: Unable to open audio channel: %s\n"
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#define MSGTR_AO_SGI_Uninit "[AO SGI] uninit: ...\n"
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#define MSGTR_AO_SGI_Reset "[AO SGI] reset: ...\n"
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#define MSGTR_AO_SGI_PauseInfo "[AO SGI] audio_pause: ...\n"
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#define MSGTR_AO_SGI_ResumeInfo "[AO SGI] audio_resume: ...\n"
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// ao_sun.c
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#define MSGTR_AO_SUN_RtscSetinfoFailed "[AO SUN] rtsc: SETINFO failed.\n"
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#define MSGTR_AO_SUN_RtscWriteFailed "[AO SUN] rtsc: write failed."
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#define MSGTR_AO_SUN_CantOpenAudioDev "[AO SUN] Can't open audio device %s, %s -> nosound.\n"
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#define MSGTR_AO_SUN_UnsupSampleRate "[AO SUN] audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate.\n"
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#define MSGTR_AO_SUN_CantUseSelect "[AO SUN]\n *** Your audio driver DOES NOT support select() ***\nRecompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n"
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#define MSGTR_AO_SUN_CantReopenReset "[AO SUN]\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n"
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// ao_alsa5.c
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#define MSGTR_AO_ALSA5_InitInfo "[AO ALSA5] alsa-init: requested format: %d Hz, %d channels, %s\n"
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#define MSGTR_AO_ALSA5_SoundCardNotFound "[AO ALSA5] alsa-init: no soundcards found.\n"
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#define MSGTR_AO_ALSA5_InvalidFormatReq "[AO ALSA5] alsa-init: invalid format (%s) requested - output disabled.\n"
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#define MSGTR_AO_ALSA5_PlayBackError "[AO ALSA5] alsa-init: playback open error: %s\n"
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#define MSGTR_AO_ALSA5_PcmInfoError "[AO ALSA5] alsa-init: pcm info error: %s\n"
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#define MSGTR_AO_ALSA5_SoundcardsFound "[AO ALSA5] alsa-init: %d soundcard(s) found, using: %s\n"
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#define MSGTR_AO_ALSA5_PcmChanInfoError "[AO ALSA5] alsa-init: pcm channel info error: %s\n"
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#define MSGTR_AO_ALSA5_CantSetParms "[AO ALSA5] alsa-init: error setting parameters: %s\n"
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#define MSGTR_AO_ALSA5_CantSetChan "[AO ALSA5] alsa-init: error setting up channel: %s\n"
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#define MSGTR_AO_ALSA5_ChanPrepareError "[AO ALSA5] alsa-init: channel prepare error: %s\n"
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#define MSGTR_AO_ALSA5_DrainError "[AO ALSA5] alsa-uninit: playback drain error: %s\n"
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#define MSGTR_AO_ALSA5_FlushError "[AO ALSA5] alsa-uninit: playback flush error: %s\n"
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#define MSGTR_AO_ALSA5_PcmCloseError "[AO ALSA5] alsa-uninit: pcm close error: %s\n"
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#define MSGTR_AO_ALSA5_ResetDrainError "[AO ALSA5] alsa-reset: playback drain error: %s\n"
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#define MSGTR_AO_ALSA5_ResetFlushError "[AO ALSA5] alsa-reset: playback flush error: %s\n"
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#define MSGTR_AO_ALSA5_ResetChanPrepareError "[AO ALSA5] alsa-reset: channel prepare error: %s\n"
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#define MSGTR_AO_ALSA5_PauseDrainError "[AO ALSA5] alsa-pause: playback drain error: %s\n"
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#define MSGTR_AO_ALSA5_PauseFlushError "[AO ALSA5] alsa-pause: playback flush error: %s\n"
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#define MSGTR_AO_ALSA5_ResumePrepareError "[AO ALSA5] alsa-resume: channel prepare error: %s\n"
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#define MSGTR_AO_ALSA5_Underrun "[AO ALSA5] alsa-play: alsa underrun, resetting stream.\n"
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#define MSGTR_AO_ALSA5_PlaybackPrepareError "[AO ALSA5] alsa-play: playback prepare error: %s\n"
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#define MSGTR_AO_ALSA5_WriteErrorAfterReset "[AO ALSA5] alsa-play: write error after reset: %s - giving up.\n"
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#define MSGTR_AO_ALSA5_OutPutError "[AO ALSA5] alsa-play: output error: %s\n"
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@ -16,6 +16,7 @@
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#include "afmt.h"
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#include "../mp_msg.h"
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#include "../help_mp.h"
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static ao_info_t info =
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{
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@ -50,7 +51,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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snd_pcm_info_t info;
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snd_pcm_channel_info_t chninfo;
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mp_msg(MSGT_AO, MSGL_INFO, "alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz,
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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_InitInfo, rate_hz,
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channels, audio_out_format_name(format));
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alsa_handler = NULL;
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@ -60,7 +61,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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if ((cards = snd_cards()) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: no soundcards found\n");
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_SoundCardNotFound);
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return(0);
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}
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@ -110,7 +111,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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ao_data.bps *= 2;
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break;
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case -1:
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: invalid format (%s) requested - output disabled\n",
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,
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audio_out_format_name(format));
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return(0);
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default:
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@ -163,17 +164,17 @@ static int init(int rate_hz, int channels, int format, int flags)
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if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: playback open error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlayBackError, snd_strerror(err));
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return(0);
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}
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if ((err = snd_pcm_info(alsa_handler, &info)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: pcm info error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmInfoError, snd_strerror(err));
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return(0);
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}
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mp_msg(MSGT_AO, MSGL_INFO, "alsa-init: %d soundcard(s) found, using: %s\n",
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mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_SoundcardsFound,
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cards, info.name);
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if (info.flags & SND_PCM_INFO_PLAYBACK)
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@ -182,7 +183,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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chninfo.channel = SND_PCM_CHANNEL_PLAYBACK;
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if ((err = snd_pcm_channel_info(alsa_handler, &chninfo)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: pcm channel info error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmChanInfoError, snd_strerror(err));
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return(0);
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}
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@ -206,7 +207,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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if ((err = snd_pcm_channel_params(alsa_handler, ¶ms)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: error setting parameters: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetParms, snd_strerror(err));
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return(0);
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}
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@ -219,13 +220,13 @@ static int init(int rate_hz, int channels, int format, int flags)
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if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: error setting up channel: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetChan, snd_strerror(err));
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return(0);
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}
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if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: channel prepare error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ChanPrepareError, snd_strerror(err));
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return(0);
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}
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@ -242,19 +243,19 @@ static void uninit(int immed)
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if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: playback drain error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_DrainError, snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: playback flush error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_FlushError, snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_close(alsa_handler)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: pcm close error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmCloseError, snd_strerror(err));
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return;
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}
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}
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@ -266,19 +267,19 @@ static void reset()
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if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: playback drain error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetDrainError, snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: playback flush error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetFlushError, snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: channel prepare error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetChanPrepareError, snd_strerror(err));
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return;
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}
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}
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@ -290,13 +291,13 @@ static void audio_pause()
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if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-pause: playback drain error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseDrainError, snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-pause: playback flush error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseFlushError, snd_strerror(err));
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return;
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}
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}
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@ -307,7 +308,7 @@ static void audio_resume()
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int err;
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if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-resume: channel prepare error: %s\n", snd_strerror(err));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResumePrepareError, snd_strerror(err));
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return;
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}
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}
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@ -327,21 +328,21 @@ static int play(void* data, int len, int flags)
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{
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if (got_len == -EPIPE) /* underrun? */
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: alsa underrun, resetting stream\n");
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_Underrun);
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if ((got_len = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlaybackPrepareError, snd_strerror(got_len));
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return(0);
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}
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if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: write error after reset: %s - giving up\n",
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mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_WriteErrorAfterReset,
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snd_strerror(got_len));
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return(0);
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}
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return(got_len); /* 2nd write was ok */
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}
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: output error: %s\n", snd_strerror(got_len));
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_OutPutError, snd_strerror(got_len));
|
||||
return(0);
|
||||
}
|
||||
return(got_len);
|
||||
|
|
|
|||
|
|
@ -15,6 +15,7 @@
|
|||
#include "afmt.h"
|
||||
#include "../config.h"
|
||||
#include "../mp_msg.h"
|
||||
#include "../help_mp.h"
|
||||
|
||||
#define OBTAIN_BITRATE(a) (((a != AFMT_U8) && (a != AFMT_S8)) ? 16 : 8)
|
||||
|
||||
|
|
@ -45,10 +46,10 @@ static int init(int rate_hz, int channels, int format, int flags)
|
|||
int frag_spec;
|
||||
|
||||
if( (err=arts_init()) ) {
|
||||
mp_msg(MSGT_AO, MSGL_ERR, "AO: [arts] %s\n", arts_error_text(err));
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantInit, arts_error_text(err));
|
||||
return 0;
|
||||
}
|
||||
mp_msg(MSGT_AO, MSGL_INFO, "AO: [arts] Connected to sound server\n");
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_ServerConnect);
|
||||
|
||||
/*
|
||||
* arts supports 8bit unsigned and 16bit signed sample formats
|
||||
|
|
@ -79,7 +80,7 @@ static int init(int rate_hz, int channels, int format, int flags)
|
|||
stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer");
|
||||
|
||||
if(stream == NULL) {
|
||||
mp_msg(MSGT_AO, MSGL_ERR, "AO: [arts] Unable to open a stream\n");
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantOpenStream);
|
||||
arts_free();
|
||||
return 0;
|
||||
}
|
||||
|
|
@ -90,11 +91,11 @@ static int init(int rate_hz, int channels, int format, int flags)
|
|||
frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16;
|
||||
arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec);
|
||||
ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE);
|
||||
mp_msg(MSGT_AO, MSGL_INFO, "AO: [arts] Stream opened\n");
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_StreamOpen);
|
||||
|
||||
mp_msg(MSGT_AO, MSGL_INFO,"AO: [arts] buffer size: %d\n",
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize,
|
||||
ao_data.buffersize);
|
||||
mp_msg(MSGT_AO, MSGL_INFO,"AO: [arts] packet size: %d\n",
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize,
|
||||
arts_stream_get(stream, ARTS_P_PACKET_SIZE));
|
||||
|
||||
return 1;
|
||||
|
|
|
|||
|
|
@ -50,7 +50,7 @@ static int control(int cmd,void *arg){
|
|||
if(v.arg != volume) {
|
||||
volume = v.arg;
|
||||
if( ioctl(dxr2_fd,DXR2_IOC_SET_AUDIO_VOLUME,&v) < 0) {
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"DXR2 : Setting volume to %d failed\n",volume);
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_SetVolFailed,volume);
|
||||
return CONTROL_ERROR;
|
||||
}
|
||||
}
|
||||
|
|
@ -110,7 +110,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
break;
|
||||
#endif
|
||||
default:
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"[AO] dxr2: %d Hz not supported, try \"-aop list=resample\"\n",rate);
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_UnsupSamplerate,rate);
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -37,6 +37,7 @@
|
|||
#include "afmt.h"
|
||||
#include "../config.h"
|
||||
#include "../mp_msg.h"
|
||||
#include "../help_mp.h"
|
||||
|
||||
|
||||
#undef ESD_DEBUG
|
||||
|
|
@ -152,8 +153,7 @@ static int init(int rate_hz, int channels, int format, int flags)
|
|||
if (esd_fd < 0) {
|
||||
esd_fd = esd_open_sound(server);
|
||||
if (esd_fd < 0) {
|
||||
mp_msg(MSGT_AO, MSGL_ERR,
|
||||
"AO: [esd] esd_open_sound failed: %s\n",
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ESD_CantOpenSound,
|
||||
strerror(errno));
|
||||
return 0;
|
||||
}
|
||||
|
|
@ -230,17 +230,14 @@ static int init(int rate_hz, int channels, int format, int flags)
|
|||
lag_serv = (esd_latency * 4.0f) / (bytes_per_sample * rate_hz);
|
||||
lag_seconds = lag_net + lag_serv;
|
||||
audio_delay += lag_seconds;
|
||||
mp_msg(MSGT_AO, MSGL_INFO,
|
||||
"AO: [esd] latency: [server: %0.2fs, net: %0.2fs] "
|
||||
"(adjust %0.2fs)\n", lag_serv, lag_net, lag_seconds);
|
||||
mp_msg(MSGT_AO, MSGL_INFO,MSGTR_AO_ESD_LatencyInfo,
|
||||
lag_serv, lag_net, lag_seconds);
|
||||
}
|
||||
|
||||
esd_play_fd = esd_play_stream_fallback(esd_fmt, rate_hz,
|
||||
server, ESD_CLIENT_NAME);
|
||||
if (esd_play_fd < 0) {
|
||||
mp_msg(MSGT_AO, MSGL_ERR,
|
||||
"AO: [esd] failed to open esd playback stream: %s\n",
|
||||
strerror(errno));
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ESD_CantOpenPBStream, strerror(errno));
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -18,6 +18,7 @@
|
|||
#include "afmt.h"
|
||||
|
||||
#include "../mp_msg.h"
|
||||
#include "../help_mp.h"
|
||||
|
||||
#ifdef HAVE_DVB
|
||||
#ifndef HAVE_DVB_HEAD
|
||||
|
|
@ -67,7 +68,7 @@ static int control(int cmd,void *arg){
|
|||
if(dvb_mixer.volume_right>255) dvb_mixer.volume_right=255;
|
||||
// printf("Setting DVB volume: %d ; %d \n",dvb_mixer.volume_left,dvb_mixer.volume_right);
|
||||
if ( (ioctl(vo_mpegpes_fd2,AUDIO_SET_MIXER, &dvb_mixer) < 0)){
|
||||
mp_msg(MSGT_AO, MSGL_ERR, "DVB audio set mixer failed: %s\n",
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_MPEGPES_CantSetMixer,
|
||||
strerror(errno));
|
||||
return CONTROL_ERROR;
|
||||
}
|
||||
|
|
@ -112,7 +113,7 @@ retry:
|
|||
case 44100: freq_id=2;break;
|
||||
case 32000: freq_id=3;break;
|
||||
default:
|
||||
mp_msg(MSGT_AO, MSGL_ERR, "ao_mpegpes: %d Hz not supported, try to resample...\n",rate);
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_MPEGPES_UnsupSamplerate, rate);
|
||||
#if 0
|
||||
if(rate>48000) rate=96000; else
|
||||
if(rate>44100) rate=48000; else
|
||||
|
|
|
|||
|
|
@ -110,9 +110,3 @@ static float get_delay(){
|
|||
drain();
|
||||
return (float) buffer / (float) ao_data.bps;
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -14,6 +14,7 @@
|
|||
#include "../config.h"
|
||||
#include "../mp_msg.h"
|
||||
#include "../mixer.h"
|
||||
#include "../help_mp.h"
|
||||
|
||||
#include "afmt.h"
|
||||
|
||||
|
|
@ -108,7 +109,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
int fd, devs, i;
|
||||
|
||||
if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't open mixer device %s: %s\n",
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenMixer,
|
||||
oss_mixer_device, strerror(errno));
|
||||
}else{
|
||||
ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
|
||||
|
|
@ -117,7 +118,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
|
||||
if(!strcasecmp(mixer_channels[i], mixer_channel)){
|
||||
if(!(devs & (1 << i))){
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Audio card mixer does not have channel '%s' using default\n",
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,
|
||||
mixer_channel);
|
||||
i = SOUND_MIXER_NRDEVICES+1;
|
||||
break;
|
||||
|
|
@ -127,7 +128,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
}
|
||||
}
|
||||
if(i==SOUND_MIXER_NRDEVICES){
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't find mixer channel '%s' using default\n",
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,
|
||||
mixer_channel);
|
||||
}
|
||||
}
|
||||
|
|
@ -143,14 +144,14 @@ static int init(int rate,int channels,int format,int flags){
|
|||
audio_fd=open(dsp, O_WRONLY);
|
||||
#endif
|
||||
if(audio_fd<0){
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't open audio device %s: %s\n", dsp, strerror(errno));
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenDev, dsp, strerror(errno));
|
||||
return 0;
|
||||
}
|
||||
|
||||
#ifdef __linux__
|
||||
/* Remove the non-blocking flag */
|
||||
if(fcntl(audio_fd, F_SETFL, 0) < 0) {
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Can't make filedescriptor blocking: %s\n", strerror(errno));
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantMakeFd, strerror(errno));
|
||||
return 0;
|
||||
}
|
||||
#endif
|
||||
|
|
@ -168,7 +169,7 @@ ac3_retry:
|
|||
ao_data.format=format;
|
||||
if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format)<0 ||
|
||||
ao_data.format != format) if(format == AFMT_AC3){
|
||||
mp_msg(MSGT_AO,MSGL_WARN,"Can't set audio device %s to AC3 output, trying S16...\n", dsp);
|
||||
mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSetAC3, dsp);
|
||||
#ifdef WORDS_BIGENDIAN
|
||||
format=AFMT_S16_BE;
|
||||
#else
|
||||
|
|
@ -189,14 +190,14 @@ ac3_retry:
|
|||
if (ao_data.channels > 2) {
|
||||
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
|
||||
ao_data.channels != channels ) {
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Failed to set audio device to %d channels\n", channels);
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, channels);
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
else {
|
||||
int c = ao_data.channels-1;
|
||||
if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"audio_setup: Failed to set audio device to %d channels\n", ao_data.channels);
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, ao_data.channels);
|
||||
return 0;
|
||||
}
|
||||
ao_data.channels=c+1;
|
||||
|
|
@ -214,7 +215,7 @@ ac3_retry:
|
|||
|
||||
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
|
||||
int r=0;
|
||||
mp_msg(MSGT_AO,MSGL_WARN,"audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
|
||||
mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_OSS_CantUseGetospace);
|
||||
if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
|
||||
mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
|
||||
} else {
|
||||
|
|
@ -245,8 +246,7 @@ ac3_retry:
|
|||
}
|
||||
free(data);
|
||||
if(ao_data.buffersize==0){
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"\n *** Your audio driver DOES NOT support select() ***\n"
|
||||
"Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantUseSelect);
|
||||
return 0;
|
||||
}
|
||||
#endif
|
||||
|
|
@ -283,7 +283,7 @@ static void reset(){
|
|||
uninit(1);
|
||||
audio_fd=open(dsp, O_WRONLY);
|
||||
if(audio_fd < 0){
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno));
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantReopen, strerror(errno));
|
||||
return;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -8,6 +8,9 @@
|
|||
#include "afmt.h"
|
||||
#include "audio_out.h"
|
||||
#include "audio_out_internal.h"
|
||||
#include "../mp_msg.h"
|
||||
#include "../help_mp.h"
|
||||
|
||||
|
||||
static ao_info_t info =
|
||||
{
|
||||
|
|
@ -111,13 +114,10 @@ static int init(int rate,int channels,int format,int flags){
|
|||
wavhdr.data_length=le2me_32(0x7ffff000);
|
||||
wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
|
||||
|
||||
printf("PCM: File: %s (%s)\n"
|
||||
"PCM: Samplerate: %iHz Channels: %s Format %s\n",
|
||||
ao_outputfilename, (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
|
||||
(ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
|
||||
(channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
|
||||
printf("PCM: Info: fastest dumping is achieved with -vc dummy -vo null\n"
|
||||
"PCM: Info: to write WAVE files use -waveheader (default); "
|
||||
"for RAW PCM -nowaveheader.\n");
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
|
||||
|
||||
fp = fopen(ao_outputfilename, "wb");
|
||||
if(fp) {
|
||||
|
|
@ -127,7 +127,8 @@ static int init(int rate,int channels,int format,int flags){
|
|||
}
|
||||
return 1;
|
||||
}
|
||||
printf("PCM: Failed to open %s for writing!\n", ao_outputfilename);
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile,
|
||||
ao_outputfilename);
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -16,6 +16,7 @@
|
|||
|
||||
#include "../config.h"
|
||||
#include "../mp_msg.h"
|
||||
#include "../help_mp.h"
|
||||
|
||||
#include "audio_out.h"
|
||||
#include "audio_out_internal.h"
|
||||
|
|
@ -171,11 +172,11 @@ static int init(int rate,int channels,int format,int flags){
|
|||
/* Allocate ring-buffer memory */
|
||||
for(i=0;i<NUM_BUFS;i++) buffer[i]=(unsigned char *) malloc(BUFFSIZE);
|
||||
|
||||
mp_msg(MSGT_AO,MSGL_INFO,"SDL: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
|
||||
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
|
||||
|
||||
if(ao_subdevice) {
|
||||
setenv("SDL_AUDIODRIVER", ao_subdevice, 1);
|
||||
mp_msg(MSGT_AO,MSGL_INFO,"SDL: using %s audio driver\n", ao_subdevice);
|
||||
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_DriverInfo, ao_subdevice);
|
||||
}
|
||||
|
||||
ao_data.channels=channels;
|
||||
|
|
@ -209,7 +210,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
default:
|
||||
aspec.format = AUDIO_S16LSB;
|
||||
ao_data.format = AFMT_S16_LE;
|
||||
mp_msg(MSGT_AO,MSGL_WARN,"SDL: Unsupported audio format: 0x%x.\n", format);
|
||||
mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, format);
|
||||
}
|
||||
|
||||
/* The desired audio frequency in samples-per-second. */
|
||||
|
|
@ -230,13 +231,13 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
|
|||
|
||||
/* initialize the SDL Audio system */
|
||||
if (SDL_Init (SDL_INIT_AUDIO/*|SDL_INIT_NOPARACHUTE*/)) {
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"SDL: Initializing of SDL Audio failed: %s.\n", SDL_GetError());
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantInit, SDL_GetError());
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Open the audio device and start playing sound! */
|
||||
if(SDL_OpenAudio(&aspec, &obtained) < 0) {
|
||||
mp_msg(MSGT_AO,MSGL_ERR,"SDL: Unable to open audio: %s\n", SDL_GetError());
|
||||
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantOpenAudio, SDL_GetError());
|
||||
return(0);
|
||||
}
|
||||
|
||||
|
|
@ -264,7 +265,7 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
|
|||
ao_data.format = AFMT_U16_BE;
|
||||
break;
|
||||
default:
|
||||
mp_msg(MSGT_AO,MSGL_WARN,"SDL: Unsupported SDL audio format: 0x%x.\n", obtained.format);
|
||||
mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, obtained.format);
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -11,6 +11,8 @@
|
|||
|
||||
#include "audio_out.h"
|
||||
#include "audio_out_internal.h"
|
||||
#include "../mp_msg.h"
|
||||
#include "../help_mp.h"
|
||||
|
||||
static ao_info_t info =
|
||||
{
|
||||
|
|
@ -31,7 +33,7 @@ static int queue_size;
|
|||
// to set/get/query special features/parameters
|
||||
static int control(int cmd, void *arg){
|
||||
|
||||
printf("ao_sgi, control\n");
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_INFO);
|
||||
|
||||
return -1;
|
||||
}
|
||||
|
|
@ -40,7 +42,7 @@ static int control(int cmd, void *arg){
|
|||
// return: 1=success 0=fail
|
||||
static int init(int rate, int channels, int format, int flags) {
|
||||
|
||||
printf("ao_sgi, init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
|
||||
|
||||
{ /* from /usr/share/src/dmedia/audio/setrate.c */
|
||||
|
||||
|
|
@ -51,7 +53,7 @@ static int init(int rate, int channels, int format, int flags) {
|
|||
|
||||
rv = alGetResourceByName(AL_SYSTEM, "out.analog", AL_DEVICE_TYPE);
|
||||
if (!rv) {
|
||||
printf("ao_sgi, play: invalid device\n");
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice);
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
@ -63,20 +65,19 @@ static int init(int rate, int channels, int format, int flags) {
|
|||
x[1].value.i = AL_CRYSTAL_MCLK_TYPE;
|
||||
|
||||
if (alSetParams(rv,x, 2)<0) {
|
||||
printf("ao_sgi, init: setparams failed: %s\n", alGetErrorString(oserror()));
|
||||
printf("ao_sgi, init: could not set desired samplerate\n");
|
||||
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror()));
|
||||
}
|
||||
|
||||
if (x[0].sizeOut < 0) {
|
||||
printf("ao_sgi, init: AL_RATE was not accepted on the given resource\n");
|
||||
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate);
|
||||
}
|
||||
|
||||
if (alGetParams(rv,x, 1)<0) {
|
||||
printf("ao_sgi, init: getparams failed: %s\n", alGetErrorString(oserror()));
|
||||
mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror()));
|
||||
}
|
||||
|
||||
if (frate != alFixedToDouble(x[0].value.ll)) {
|
||||
printf("ao_sgi, init: samplerate is now %lf (desired rate is %lf)\n", alFixedToDouble(x[0].value.ll), frate);
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, alFixedToDouble(x[0].value.ll), frate);
|
||||
}
|
||||
sample_rate = (int)frate;
|
||||
}
|
||||
|
|
@ -88,7 +89,7 @@ static int init(int rate, int channels, int format, int flags) {
|
|||
ao_config = alNewConfig();
|
||||
|
||||
if (!ao_config) {
|
||||
printf("ao_sgi, init: %s\n", alGetErrorString(oserror()));
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
@ -100,14 +101,14 @@ static int init(int rate, int channels, int format, int flags) {
|
|||
alSetQueueSize(ao_config, 48000);
|
||||
|
||||
if (alSetDevice(ao_config, AL_DEFAULT_OUTPUT) < 0) {
|
||||
printf("ao_sgi, init: %s\n", alGetErrorString(oserror()));
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
|
||||
return 0;
|
||||
}
|
||||
|
||||
ao_port = alOpenPort("mplayer", "w", ao_config);
|
||||
|
||||
if (!ao_port) {
|
||||
printf("ao_sgi, init: Unable to open audio channel: %s\n", alGetErrorString(oserror()));
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, alGetErrorString(oserror()));
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
@ -122,7 +123,7 @@ static void uninit(int immed) {
|
|||
|
||||
/* TODO: samplerate should be set back to the value before mplayer was started! */
|
||||
|
||||
printf("ao_sgi, uninit: ...\n");
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Uninit);
|
||||
|
||||
if (ao_port) {
|
||||
while(alGetFilled(ao_port) > 0) sginap(1);
|
||||
|
|
@ -135,21 +136,21 @@ static void uninit(int immed) {
|
|||
// stop playing and empty buffers (for seeking/pause)
|
||||
static void reset() {
|
||||
|
||||
printf("ao_sgi, reset: ...\n");
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Reset);
|
||||
|
||||
}
|
||||
|
||||
// stop playing, keep buffers (for pause)
|
||||
static void audio_pause() {
|
||||
|
||||
printf("ao_sgi, audio_pause: ...\n");
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_PauseInfo);
|
||||
|
||||
}
|
||||
|
||||
// resume playing, after audio_pause()
|
||||
static void audio_resume() {
|
||||
|
||||
printf("ao_sgi, audio_resume: ...\n");
|
||||
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_ResumeInfo);
|
||||
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -26,6 +26,8 @@
|
|||
#include "audio_out.h"
|
||||
#include "audio_out_internal.h"
|
||||
#include "afmt.h"
|
||||
#include "../mp_msg.h"
|
||||
#include "../help_mp.h"
|
||||
|
||||
static ao_info_t info =
|
||||
{
|
||||
|
|
@ -126,13 +128,13 @@ static int realtime_samplecounter_available(char *dev)
|
|||
info.play.samples = 0;
|
||||
if (ioctl(fd, AUDIO_SETINFO, &info)) {
|
||||
if (verbose>0)
|
||||
printf("rtsc: SETINFO failed\n");
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed);
|
||||
goto error;
|
||||
}
|
||||
|
||||
if (write(fd, silence, len) != len) {
|
||||
if (verbose>0)
|
||||
printf("rtsc: write failed");
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed);
|
||||
goto error;
|
||||
}
|
||||
|
||||
|
|
@ -482,7 +484,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
|
||||
audio_fd=open(audio_dev, O_WRONLY);
|
||||
if(audio_fd<0){
|
||||
printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno));
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantOpenAudioDev, audio_dev, strerror(errno));
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
@ -556,7 +558,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
}
|
||||
|
||||
if (!ok) {
|
||||
printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n",
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate,
|
||||
channels, audio_out_format_name(format), rate);
|
||||
return 0;
|
||||
}
|
||||
|
|
@ -588,8 +590,7 @@ static int init(int rate,int channels,int format,int flags){
|
|||
}
|
||||
free(data);
|
||||
if(ao_data.buffersize==0){
|
||||
printf("\n *** Your audio driver DOES NOT support select() ***\n");
|
||||
printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
|
||||
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantUseSelect);
|
||||
return 0;
|
||||
}
|
||||
#ifdef __svr4__
|
||||
|
|
@ -631,7 +632,7 @@ static void reset(){
|
|||
uninit(1);
|
||||
audio_fd=open(audio_dev, O_WRONLY);
|
||||
if(audio_fd<0){
|
||||
printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno));
|
||||
mp_msg(MSGT_AO, MSGL_FATAL, MSGTR_AO_SUN_CantReopenReset, strerror(errno));
|
||||
return;
|
||||
}
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue