From d4bdd0473d6f43132257c9fb3848d829755167a3 Mon Sep 17 00:00:00 2001 From: wm4 Date: Mon, 5 Nov 2012 17:02:04 +0100 Subject: Rename directories, move files (step 1 of 2) (does not compile) Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code. --- audio/out/ao_oss.c | 560 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 560 insertions(+) create mode 100644 audio/out/ao_oss.c (limited to 'audio/out/ao_oss.c') diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c new file mode 100644 index 0000000000..9d4dde4837 --- /dev/null +++ b/audio/out/ao_oss.c @@ -0,0 +1,560 @@ +/* + * OSS audio output driver + * + * This file is part of MPlayer. + * + * MPlayer is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * MPlayer is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with MPlayer; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "config.h" +#include "mp_msg.h" +#include "mixer.h" + +#ifdef HAVE_SYS_SOUNDCARD_H +#include +#else +#ifdef HAVE_SOUNDCARD_H +#include +#endif +#endif + +#include "libaf/format.h" + +#include "audio_out.h" +#include "audio_out_internal.h" + +static const ao_info_t info = +{ + "OSS/ioctl audio output", + "oss", + "A'rpi", + "" +}; + +/* Support for >2 output channels added 2001-11-25 - Steve Davies */ + +LIBAO_EXTERN(oss) + +static int format2oss(int format) +{ + switch(format) + { + case AF_FORMAT_U8: return AFMT_U8; + case AF_FORMAT_S8: return AFMT_S8; + case AF_FORMAT_U16_LE: return AFMT_U16_LE; + case AF_FORMAT_U16_BE: return AFMT_U16_BE; + case AF_FORMAT_S16_LE: return AFMT_S16_LE; + case AF_FORMAT_S16_BE: return AFMT_S16_BE; +#ifdef AFMT_S24_PACKED + case AF_FORMAT_S24_LE: return AFMT_S24_PACKED; +#endif +#ifdef AFMT_U32_LE + case AF_FORMAT_U32_LE: return AFMT_U32_LE; +#endif +#ifdef AFMT_U32_BE + case AF_FORMAT_U32_BE: return AFMT_U32_BE; +#endif +#ifdef AFMT_S32_LE + case AF_FORMAT_S32_LE: return AFMT_S32_LE; +#endif +#ifdef AFMT_S32_BE + case AF_FORMAT_S32_BE: return AFMT_S32_BE; +#endif +#ifdef AFMT_FLOAT + case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT; +#endif + // SPECIALS + case AF_FORMAT_MU_LAW: return AFMT_MU_LAW; + case AF_FORMAT_A_LAW: return AFMT_A_LAW; + case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM; +#ifdef AFMT_MPEG + case AF_FORMAT_MPEG2: return AFMT_MPEG; +#endif +#ifdef AFMT_AC3 + case AF_FORMAT_AC3_NE: return AFMT_AC3; +#endif + } + mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format)); + return -1; +} + +static int oss2format(int format) +{ + switch(format) + { + case AFMT_U8: return AF_FORMAT_U8; + case AFMT_S8: return AF_FORMAT_S8; + case AFMT_U16_LE: return AF_FORMAT_U16_LE; + case AFMT_U16_BE: return AF_FORMAT_U16_BE; + case AFMT_S16_LE: return AF_FORMAT_S16_LE; + case AFMT_S16_BE: return AF_FORMAT_S16_BE; +#ifdef AFMT_S24_PACKED + case AFMT_S24_PACKED: return AF_FORMAT_S24_LE; +#endif +#ifdef AFMT_U32_LE + case AFMT_U32_LE: return AF_FORMAT_U32_LE; +#endif +#ifdef AFMT_U32_BE + case AFMT_U32_BE: return AF_FORMAT_U32_BE; +#endif +#ifdef AFMT_S32_LE + case AFMT_S32_LE: return AF_FORMAT_S32_LE; +#endif +#ifdef AFMT_S32_BE + case AFMT_S32_BE: return AF_FORMAT_S32_BE; +#endif +#ifdef AFMT_FLOAT + case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE; +#endif + // SPECIALS + case AFMT_MU_LAW: return AF_FORMAT_MU_LAW; + case AFMT_A_LAW: return AF_FORMAT_A_LAW; + case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM; +#ifdef AFMT_MPEG + case AFMT_MPEG: return AF_FORMAT_MPEG2; +#endif +#ifdef AFMT_AC3 + case AFMT_AC3: return AF_FORMAT_AC3_NE; +#endif + } + mp_tmsg(MSGT_GLOBAL,MSGL_ERR,"[AO OSS] Unknown/Unsupported OSS format: %x.\n", format); + return -1; +} + +static char *dsp=PATH_DEV_DSP; +static audio_buf_info zz; +static int audio_fd=-1; +static int prepause_space; + +static const char *oss_mixer_device = PATH_DEV_MIXER; +static int oss_mixer_channel = SOUND_MIXER_PCM; + +#ifdef SNDCTL_DSP_GETPLAYVOL +static int volume_oss4(ao_control_vol_t *vol, int cmd) { + int v; + + if (audio_fd < 0) + return CONTROL_ERROR; + + if (cmd == AOCONTROL_GET_VOLUME) { + if (ioctl(audio_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1) + return CONTROL_ERROR; + vol->right = (v & 0xff00) >> 8; + vol->left = v & 0x00ff; + return CONTROL_OK; + } else if (cmd == AOCONTROL_SET_VOLUME) { + v = ((int) vol->right << 8) | (int) vol->left; + if (ioctl(audio_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1) + return CONTROL_ERROR; + return CONTROL_OK; + } else + return CONTROL_UNKNOWN; +} +#endif + +// to set/get/query special features/parameters +static int control(int cmd,void *arg){ + switch(cmd){ + case AOCONTROL_GET_VOLUME: + case AOCONTROL_SET_VOLUME: + { + ao_control_vol_t *vol = (ao_control_vol_t *)arg; + int fd, v, devs; + +#ifdef SNDCTL_DSP_GETPLAYVOL + // Try OSS4 first + if (volume_oss4(vol, cmd) == CONTROL_OK) + return CONTROL_OK; +#endif + + if(AF_FORMAT_IS_AC3(ao_data.format)) + return CONTROL_TRUE; + + if ((fd = open(oss_mixer_device, O_RDONLY)) != -1) + { + ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); + if (devs & (1 << oss_mixer_channel)) + { + if (cmd == AOCONTROL_GET_VOLUME) + { + ioctl(fd, MIXER_READ(oss_mixer_channel), &v); + vol->right = (v & 0xFF00) >> 8; + vol->left = v & 0x00FF; + } + else + { + v = ((int)vol->right << 8) | (int)vol->left; + ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v); + } + } + else + { + close(fd); + return CONTROL_ERROR; + } + close(fd); + return CONTROL_OK; + } + } + return CONTROL_ERROR; + } + return CONTROL_UNKNOWN; +} + +// open & setup audio device +// return: 1=success 0=fail +static int init(int rate,int channels,int format,int flags){ + char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES; + int oss_format; + char *mdev = mixer_device, *mchan = mixer_channel; + + mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels, + af_fmt2str_short(format)); + + if (ao_subdevice) { + char *m,*c; + m = strchr(ao_subdevice,':'); + if(m) { + c = strchr(m+1,':'); + if(c) { + mchan = c+1; + c[0] = '\0'; + } + mdev = m+1; + m[0] = '\0'; + } + dsp = ao_subdevice; + } + + if(mdev) + oss_mixer_device=mdev; + else + oss_mixer_device=PATH_DEV_MIXER; + + if(mchan){ + int fd, devs, i; + + if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){ + mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open mixer device %s: %s\n", + oss_mixer_device, strerror(errno)); + }else{ + ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); + close(fd); + + for (i=0; i2 channels, in case some drivers don't have it + if (ao_data.channels > 2) { + if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 || + ao_data.channels != channels ) { + mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", channels); + return 0; + } + } + else { + int c = ao_data.channels-1; + if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) { + mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", ao_data.channels); + return 0; + } + ao_data.channels=c+1; + } + mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels); + // set rate + ao_data.samplerate=rate; + ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); + mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate); + } + + if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){ + int r=0; + mp_tmsg(MSGT_AO,MSGL_WARN,"[AO OSS] audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n"); + 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 { + ao_data.outburst=r; + mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst); + } + } else { + mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n", + zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes); + if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes; + ao_data.outburst=zz.fragsize; + } + + if(ao_data.buffersize==-1){ + // Measuring buffer size: + void* data; + ao_data.buffersize=0; +#ifdef HAVE_AUDIO_SELECT + data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst); + while(ao_data.buffersize<0x40000){ + fd_set rfds; + struct timeval tv; + FD_ZERO(&rfds); FD_SET(audio_fd,&rfds); + tv.tv_sec=0; tv.tv_usec = 0; + if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break; + write(audio_fd,data,ao_data.outburst); + ao_data.buffersize+=ao_data.outburst; + } + free(data); + if(ao_data.buffersize==0){ + mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\n *** Your audio driver DOES NOT support select() ***\n Recompile mpv with #undef HAVE_AUDIO_SELECT in config.h !\n\n"); + return 0; + } +#endif + } + + ao_data.bps=ao_data.channels; + switch (ao_data.format & AF_FORMAT_BITS_MASK) { + case AF_FORMAT_8BIT: + break; + case AF_FORMAT_16BIT: + ao_data.bps*=2; + break; + case AF_FORMAT_24BIT: + ao_data.bps*=3; + break; + case AF_FORMAT_32BIT: + ao_data.bps*=4; + break; + } + + ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down + ao_data.bps*=ao_data.samplerate; + + return 1; +} + +// close audio device +static void uninit(int immed){ + if(audio_fd == -1) return; +#ifdef SNDCTL_DSP_SYNC + // to get the buffer played + if (!immed) + ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL); +#endif +#ifdef SNDCTL_DSP_RESET + if (immed) + ioctl(audio_fd, SNDCTL_DSP_RESET, NULL); +#endif + close(audio_fd); + audio_fd = -1; +} + +// stop playing and empty buffers (for seeking/pause) +static void reset(void){ + int oss_format; + uninit(1); + audio_fd=open(dsp, O_WRONLY); + if(audio_fd < 0){ + mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno)); + return; + } + +#if defined(FD_CLOEXEC) && defined(F_SETFD) + fcntl(audio_fd, F_SETFD, FD_CLOEXEC); +#endif + + oss_format = format2oss(ao_data.format); + if(AF_FORMAT_IS_AC3(ao_data.format)) + ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); + ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format); + if(!AF_FORMAT_IS_AC3(ao_data.format)) { + if (ao_data.channels > 2) + ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels); + else { + int c = ao_data.channels-1; + ioctl (audio_fd, SNDCTL_DSP_STEREO, &c); + } + ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); + } +} + +// stop playing, keep buffers (for pause) +static void audio_pause(void) +{ + prepause_space = get_space(); + uninit(1); +} + +// resume playing, after audio_pause() +static void audio_resume(void) +{ + int fillcnt; + reset(); + fillcnt = get_space() - prepause_space; + if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) { + void *silence = calloc(fillcnt, 1); + play(silence, fillcnt, 0); + free(silence); + } +} + + +// return: how many bytes can be played without blocking +static int get_space(void){ + int playsize=ao_data.outburst; + +#ifdef SNDCTL_DSP_GETOSPACE + if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){ + // calculate exact buffer space: + playsize = zz.fragments*zz.fragsize; + return playsize; + } +#endif + + // check buffer +#ifdef HAVE_AUDIO_SELECT + { fd_set rfds; + struct timeval tv; + FD_ZERO(&rfds); + FD_SET(audio_fd, &rfds); + tv.tv_sec = 0; + tv.tv_usec = 0; + if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block! + } +#endif + + return ao_data.outburst; +} + +// plays 'len' bytes of 'data' +// it should round it down to outburst*n +// return: number of bytes played +static int play(void* data,int len,int flags){ + if(len==0) + return len; + if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) { + len/=ao_data.outburst; + len*=ao_data.outburst; + } + len=write(audio_fd,data,len); + return len; +} + +static int audio_delay_method=2; + +// return: delay in seconds between first and last sample in buffer +static float get_delay(void){ + /* Calculate how many bytes/second is sent out */ + if(audio_delay_method==2){ +#ifdef SNDCTL_DSP_GETODELAY + int r=0; + if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1) + return ((float)r)/(float)ao_data.bps; +#endif + audio_delay_method=1; // fallback if not supported + } + if(audio_delay_method==1){ + // SNDCTL_DSP_GETOSPACE + if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1) + return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps; + audio_delay_method=0; // fallback if not supported + } + return ((float)ao_data.buffersize)/(float)ao_data.bps; +} -- cgit v1.2.3