Update of /cvsroot/mplayer/main/libmpdemux In directory mail:/var/tmp.root/cvs-serv3543 Added Files: ai_alsa.c ai_oss.c audio_in.c audio_in.h Log Message: new v4l capture patch by Jindrich Makovicka <makovick@kmlinux.fjfi.cvut.cz>: - multithreaded audio/video buffering (I know mplayer crew hates threads but it seems to me as the only way of doing reliable a/v capture) - a/v timebase synchronization (sample count vs. gettimeofday) - "immediate" mode support for mplayer - fixed colorspace stuff - RGB?? and YUY2 modes now work as expected - native ALSA audio capture - separated audio input layer --- NEW FILE --- #include "config.h" #ifdef HAVE_ALSA9 #include <alsa/asoundlib.h> #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; size_t buffer_size; int err; size_t n; unsigned int rate; snd_pcm_uframes_t start_threshold, stop_threshold; 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_msg(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_msg(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_msg(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_msg(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_msg(MSGT_TV, MSGL_ERR, "Unable to install hw params:"); 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_msg(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_msg(MSGT_TV, MSGL_ERR, "unable to install sw params:\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_msg(MSGT_TV, MSGL_ERR, "Error opening audio"); return -1; } err = snd_output_stdio_attach(&ai->alsa.log, stderr, 0); if (err < 0) { return -1; } err = ai_alsa_setup(ai); return err; } #endif /* HAVE_ALSA9 */ --- NEW FILE --- #include "config.h" #include <linux/soundcard.h> #include <fcntl.h> #include <errno.h> #include "audio_in.h" #include "mp_msg.h" int ai_oss_set_samplerate(audio_in_t *ai) { int tmp = ai->req_samplerate; if (ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &tmp) == -1) return -1; ai->samplerate = ai->req_samplerate; return 0; } int ai_oss_set_channels(audio_in_t *ai) { int err; int ioctl_param; if (ai->req_channels > 2) { ioctl_param = ai->req_channels; mp_msg(MSGT_TV, MSGL_V, "ioctl dsp channels: %d\n", err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_CHANNELS, &ioctl_param)); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Unable to set channel count: %d\n", ai->req_channels); return -1; } } else { ioctl_param = (ai->req_channels == 2); mp_msg(MSGT_TV, MSGL_V, "ioctl dsp stereo: %d (req: %d)\n", err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_STEREO, &ioctl_param), ioctl_param); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Unable to set stereo: %d\n", ai->req_channels == 2); return -1; } } ai->channels = ai->req_channels; return 0; } int ai_oss_init(audio_in_t *ai) { int err; int ioctl_param; ai->oss.audio_fd = open(ai->oss.device, O_RDONLY); if (ai->oss.audio_fd < 0) { mp_msg(MSGT_TV, MSGL_ERR, "unable to open '%s': %s\n", ai->oss.device, strerror(errno)); return -1; } ioctl_param = 0 ; mp_msg(MSGT_TV, MSGL_V, "ioctl dsp getfmt: %d\n", ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETFMTS, &ioctl_param)); mp_msg(MSGT_TV, MSGL_V, "Supported formats: %x\n", ioctl_param); if (!(ioctl_param & AFMT_S16_LE)) mp_msg(MSGT_TV, MSGL_ERR, "notsupported format\n"); ioctl_param = AFMT_S16_LE; mp_msg(MSGT_TV, MSGL_V, "ioctl dsp setfmt: %d\n", err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETFMT, &ioctl_param)); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Unable to set audio format."); return -1; } if (ai_oss_set_channels(ai) < 0) return -1; ioctl_param = ai->req_samplerate; mp_msg(MSGT_TV, MSGL_V, "ioctl dsp speed: %d\n", err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &ioctl_param)); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Unable to set samplerate: %d\n", ai->req_samplerate); return -1; } ai->samplerate = ai->req_samplerate; mp_msg(MSGT_TV, MSGL_V, "ioctl dsp trigger: %d\n", ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETTRIGGER, &ioctl_param)); mp_msg(MSGT_TV, MSGL_V, "trigger: %x\n", ioctl_param); ioctl_param = PCM_ENABLE_INPUT; mp_msg(MSGT_TV, MSGL_V, "ioctl dsp trigger: %d\n", err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETTRIGGER, &ioctl_param)); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Unable to set trigger: %d\n", PCM_ENABLE_INPUT); return -1; } ai->blocksize = 0; mp_msg(MSGT_TV, MSGL_V, "ioctl dsp getblocksize: %d\n", err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETBLKSIZE, &ai->blocksize)); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Unable to get block size!\n"); } mp_msg(MSGT_TV, MSGL_V, "blocksize: %d\n", ai->blocksize); // correct the blocksize to a reasonable value if (ai->blocksize <= 0) { ai->blocksize = 4096*ai->channels*2; mp_msg(MSGT_TV, MSGL_ERR, "audio block size is zero, setting to %d!\n", ai->blocksize); } else if (ai->blocksize < 4096*ai->channels*2) { ai->blocksize *= 4096*ai->channels*2/ai->blocksize; mp_msg(MSGT_TV, MSGL_ERR, "audio block size too low, setting to %d!\n", ai->blocksize); } ai->samplesize = 16; ai->bytes_per_sample = 2; return 0; } --- NEW FILE --- #include "config.h" #include "audio_in.h" #include "mp_msg.h" #include <stdio.h> #include <stdlib.h> #include <string.h> #include <errno.h> // sanitizes ai structure before calling other functions int audio_in_init(audio_in_t *ai, int type) { ai->type = type; ai->setup = 0; ai->channels = -1; ai->samplerate = -1; ai->blocksize = -1; ai->bytes_per_sample = -1; ai->samplesize = -1; switch (ai->type) { #ifdef HAVE_ALSA9 case AUDIO_IN_ALSA: ai->alsa.handle = NULL; ai->alsa.log = NULL; ai->alsa.device = strdup("default"); return 0; #endif case AUDIO_IN_OSS: ai->oss.audio_fd = -1; ai->oss.device = strdup("/dev/dsp"); return 0; default: return -1; } } int audio_in_setup(audio_in_t *ai) { int err; switch (ai->type) { #ifdef HAVE_ALSA9 case AUDIO_IN_ALSA: if (ai_alsa_init(ai) < 0) return -1; ai->setup = 1; return 0; #endif case AUDIO_IN_OSS: if (ai_oss_init(ai) < 0) return -1; ai->setup = 1; return 0; default: return -1; } } int audio_in_set_samplerate(audio_in_t *ai, int rate) { switch (ai->type) { #ifdef HAVE_ALSA9 case AUDIO_IN_ALSA: ai->req_samplerate = rate; if (!ai->setup) return 0; if (ai_alsa_setup(ai) < 0) return -1; return ai->samplerate; #endif case AUDIO_IN_OSS: ai->req_samplerate = rate; if (!ai->setup) return 0; if (ai_oss_set_samplerate(ai) < 0) return -1; return ai->samplerate; default: return -1; } } int audio_in_set_channels(audio_in_t *ai, int channels) { switch (ai->type) { #ifdef HAVE_ALSA9 case AUDIO_IN_ALSA: ai->req_channels = channels; if (!ai->setup) return 0; if (ai_alsa_setup(ai) < 0) return -1; return ai->channels; #endif case AUDIO_IN_OSS: ai->req_channels = channels; if (!ai->setup) return 0; if (ai_oss_set_channels(ai) < 0) return -1; return ai->channels; default: return -1; } } int audio_in_set_device(audio_in_t *ai, char *device) { int i; if (ai->setup) return -1; switch (ai->type) { #ifdef HAVE_ALSA9 case AUDIO_IN_ALSA: if (ai->alsa.device) free(ai->alsa.device); ai->alsa.device = strdup(device); /* mplayer cannot handle colons in arguments */ for (i = 0; i < strlen(ai->alsa.device); i++) { if (ai->alsa.device[i] == ',') ai->alsa.device[i] = ':'; } return 0; #endif case AUDIO_IN_OSS: if (ai->oss.device) free(ai->oss.device); ai->oss.device = strdup(device); return 0; default: return -1; } } int audio_in_uninit(audio_in_t *ai) { if (ai->setup) { switch (ai->type) { #ifdef HAVE_ALSA9 case AUDIO_IN_ALSA: if (ai->alsa.log) snd_output_close(ai->alsa.log); if (ai->alsa.handle) { snd_pcm_close(ai->alsa.handle); } ai->setup = 0; return 0; #endif case AUDIO_IN_OSS: close(ai->oss.audio_fd); ai->setup = 0; return 0; default: return -1; } } } int audio_in_start_capture(audio_in_t *ai) { switch (ai->type) { #ifdef HAVE_ALSA9 case AUDIO_IN_ALSA: return snd_pcm_start(ai->alsa.handle); #endif case AUDIO_IN_OSS: return 0; default: return -1; } } int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) { int ret; switch (ai->type) { #ifdef HAVE_ALSA9 case AUDIO_IN_ALSA: ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); if (ret != ai->alsa.chunk_size) { if (ret < 0) { mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", snd_strerror(ret)); } else { mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); } return -1; } return ret; #endif case AUDIO_IN_OSS: ret = read(ai->oss.audio_fd, buffer, ai->blocksize); if (ret != ai->blocksize) { if (ret < 0) { mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", strerror(errno)); } else { mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); } return -1; } return ret; default: return -1; } } --- NEW FILE --- #ifndef _audio_in_h #define _audio_in_h #define AUDIO_IN_ALSA 1 #define AUDIO_IN_OSS 2 #include "config.h" #ifdef HAVE_ALSA9 #include <alsa/asoundlib.h> typedef struct { char *device; snd_pcm_t *handle; snd_output_t *log; int buffer_time, period_time, chunk_size; size_t bits_per_sample, bits_per_frame; } ai_alsa_t; #endif typedef struct { char *device; int audio_fd; } ai_oss_t; typedef struct { int type; int setup; /* requested values */ int req_channels; int req_samplerate; /* real values read-only */ int channels; int samplerate; int blocksize; int bytes_per_sample; int samplesize; #ifdef HAVE_ALSA9 ai_alsa_t alsa; #endif ai_oss_t oss; } audio_in_t; int audio_in_init(audio_in_t *ai, int type); int audio_in_setup(audio_in_t *ai); int audio_in_set_device(audio_in_t *ai, char *device); int audio_in_set_samplerate(audio_in_t *ai, int rate); int audio_in_set_channels(audio_in_t *ai, int channels); int audio_in_uninit(audio_in_t *ai); int audio_in_start_capture(audio_in_t *ai); int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer); #ifdef HAVE_ALSA9 int ai_alsa_setup(audio_in_t *ai); int ai_alsa_init(audio_in_t *ai); #endif int ai_oss_set_samplerate(audio_in_t *ai); int ai_oss_set_channels(audio_in_t *ai); int ai_oss_init(audio_in_t *ai); #endif /* _audio_in_h */