[Mplayer-cvslog] CVS: main/libmpdemux ai_alsa.c,NONE,1.1 ai_oss.c,NONE,1.1 audio_in.c,NONE,1.1 audio_in.h,NONE,1.1

[Arpi of Ize]

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 at 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(&params);
    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 */