sdl_alsa_audio.c

SDL库 在进行视频显示程序spcaview安装时必须的库文件

 *  and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
 */
#define SWIZ6(T) \
    T *ptr = (T *) mixbuf; \
    const Uint32 count = (this->spec.samples / 6); \
    Uint32 i; \
    for (i = 0; i < count; i++, ptr += 6) { \
        T tmp; \
        tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
        tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
    }

static __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); }
static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); }
static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); }
static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); }

#undef SWIZ6


/*
 * Called right before feeding this->mixbuf to the hardware. Swizzle channels
 *  from Windows/Mac order to the format alsalib will want.
 */
static __inline__ void swizzle_alsa_channels(_THIS)
{
    if (this->spec.channels == 6) {
        const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
        if (fmtsize == 16)
            swizzle_alsa_channels_6_16bit(this);
        else if (fmtsize == 8)
            swizzle_alsa_channels_6_8bit(this);
        else if (fmtsize == 32)
            swizzle_alsa_channels_6_32bit(this);
        else if (fmtsize == 64)
            swizzle_alsa_channels_6_64bit(this);
    }

    /* !!! FIXME: update this for 7.1 if needed, later. */
}


static void ALSA_PlayAudio(_THIS)
{
	int           status;
	int           sample_len;
	signed short *sample_buf;

	swizzle_alsa_channels(this);

	sample_len = this->spec.samples;
	sample_buf = (signed short *)mixbuf;

	while ( sample_len > 0 ) {
		status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len);
		if ( status < 0 ) {
			if ( status == -EAGAIN ) {
				SDL_Delay(1);
				continue;
			}
			if ( status == -ESTRPIPE ) {
				do {
					SDL_Delay(1);
					status = SDL_NAME(snd_pcm_resume)(pcm_handle);
				} while ( status == -EAGAIN );
			}
			if ( status < 0 ) {
				status = SDL_NAME(snd_pcm_prepare)(pcm_handle);
			}
			if ( status < 0 ) {
				/* Hmm, not much we can do - abort */
				this->enabled = 0;
				return;
			}
			continue;
		}
		sample_buf += status * this->spec.channels;
		sample_len -= status;
	}
}

static Uint8 *ALSA_GetAudioBuf(_THIS)
{
	return(mixbuf);
}

static void ALSA_CloseAudio(_THIS)
{
	if ( mixbuf != NULL ) {
		SDL_FreeAudioMem(mixbuf);
		mixbuf = NULL;
	}
	if ( pcm_handle ) {
		SDL_NAME(snd_pcm_drain)(pcm_handle);
		SDL_NAME(snd_pcm_close)(pcm_handle);
		pcm_handle = NULL;
	}
}

static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
	int                  status;
	snd_pcm_hw_params_t *hwparams;
	snd_pcm_sw_params_t *swparams;
	snd_pcm_format_t     format;
	snd_pcm_uframes_t    frames;
	Uint16               test_format;

	/* Open the audio device */
	/* Name of device should depend on # channels in spec */
	status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);

	if ( status < 0 ) {
		SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
		return(-1);
	}

	/* Figure out what the hardware is capable of */
	snd_pcm_hw_params_alloca(&hwparams);
	status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams);
	if ( status < 0 ) {
		SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* SDL only uses interleaved sample output */
	status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* Try for a closest match on audio format */
	status = -1;
	for ( test_format = SDL_FirstAudioFormat(spec->format);
	      test_format && (status < 0); ) {
		switch ( test_format ) {
			case AUDIO_U8:
				format = SND_PCM_FORMAT_U8;
				break;
			case AUDIO_S8:
				format = SND_PCM_FORMAT_S8;
				break;
			case AUDIO_S16LSB:
				format = SND_PCM_FORMAT_S16_LE;
				break;
			case AUDIO_S16MSB:
				format = SND_PCM_FORMAT_S16_BE;
				break;
			case AUDIO_U16LSB:
				format = SND_PCM_FORMAT_U16_LE;
				break;
			case AUDIO_U16MSB:
				format = SND_PCM_FORMAT_U16_BE;
				break;
			default:
				format = 0;
				break;
		}
		if ( format != 0 ) {
			status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format);
		}
		if ( status < 0 ) {
			test_format = SDL_NextAudioFormat();
		}
	}
	if ( status < 0 ) {
		SDL_SetError("Couldn't find any hardware audio formats");
		ALSA_CloseAudio(this);
		return(-1);
	}
	spec->format = test_format;

	/* Set the number of channels */
	status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels);
	if ( status < 0 ) {
		status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams);
		if ( (status <= 0) || (status > 2) ) {
			SDL_SetError("Couldn't set audio channels");
			ALSA_CloseAudio(this);
			return(-1);
		}
		spec->channels = status;
	}

	/* Set the audio rate */
	status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, spec->freq, NULL);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
	spec->freq = status;

	/* Set the buffer size, in samples */
	frames = spec->samples;
	frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, frames, NULL);
	spec->samples = frames;
	SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, 2, NULL);

	/* "set" the hardware with the desired parameters */
	status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

/* This is useful for debugging... */
/*
{ snd_pcm_sframes_t bufsize; int fragments;
   bufsize = SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams);
   fragments = SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams);

   fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments);
}
*/

	/* Set the software parameters */
	snd_pcm_sw_params_alloca(&swparams);
	status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams);
	if ( status < 0 ) {
		SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
	status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 0);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
	status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, frames);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set avail min: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
	status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* Calculate the final parameters for this audio specification */
	SDL_CalculateAudioSpec(spec);

	/* Allocate mixing buffer */
	mixlen = spec->size;
	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
	if ( mixbuf == NULL ) {
		ALSA_CloseAudio(this);
		return(-1);
	}
	SDL_memset(mixbuf, spec->silence, spec->size);

	/* Get the parent process id (we're the parent of the audio thread) */
	parent = getpid();

	/* Switch to blocking mode for playback */
	SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);

	/* We're ready to rock and roll. :-) */
	return(0);
}