renamed src/audioOutputs/ to src/output/

Again, no CamelCase in the directory name.

1 files changed, 0 insertions, 444 deletions

diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
deleted file mode 100644
index 1845f1b76..000000000
--- a/src/audioOutputs/audioOutput_alsa.c
+++ /dev/null
@@ -1,444 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program 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.
- *
- * This program 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 this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_ALSA
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-static const char default_device[] = "default";
-
-#define MPD_ALSA_RETRY_NR 5
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <alsa/asoundlib.h>
-
-typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
-					snd_pcm_uframes_t size);
-
-typedef struct _AlsaData {
-	const char *device;
-
-	/** the mode flags passed to snd_pcm_open */
-	int mode;
-
-	snd_pcm_t *pcmHandle;
-	alsa_writei_t *writei;
-	unsigned int buffer_time;
-	unsigned int period_time;
-	int sampleSize;
-	int useMmap;
-} AlsaData;
-
-static AlsaData *newAlsaData(void)
-{
-	AlsaData *ret = xmalloc(sizeof(AlsaData));
-
-	ret->device = default_device;
-	ret->mode = 0;
-	ret->pcmHandle = NULL;
-	ret->writei = snd_pcm_writei;
-	ret->useMmap = 0;
-	ret->buffer_time = 0;
-	ret->period_time = 0;
-
-	return ret;
-}
-
-static void freeAlsaData(AlsaData * ad)
-{
-	if (ad->device && ad->device != default_device)
-		xfree(ad->device);
-	free(ad);
-}
-
-static void
-alsa_configure(AlsaData *ad, ConfigParam *param)
-{
-	BlockParam *bp;
-
-	if ((bp = getBlockParam(param, "device")))
-		ad->device = xstrdup(bp->value);
-	ad->useMmap = getBoolBlockParam(param, "use_mmap", 1);
-	if (ad->useMmap == CONF_BOOL_UNSET)
-		ad->useMmap = 0;
-	if ((bp = getBlockParam(param, "buffer_time")))
-		ad->buffer_time = atoi(bp->value);
-	if ((bp = getBlockParam(param, "period_time")))
-		ad->period_time = atoi(bp->value);
-
-#ifdef SND_PCM_NO_AUTO_RESAMPLE
-	if (!getBoolBlockParam(param, "auto_resample", true))
-		ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_CHANNELS
-	if (!getBoolBlockParam(param, "auto_channels", true))
-		ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_FORMAT
-	if (!getBoolBlockParam(param, "auto_format", true))
-		ad->mode |= SND_PCM_NO_AUTO_FORMAT;
-#endif
-}
-
-static void *alsa_initDriver(mpd_unused struct audio_output *ao,
-			     mpd_unused const struct audio_format *audio_format,
-			     ConfigParam * param)
-{
-	/* no need for pthread_once thread-safety when reading config */
-	static int free_global_registered;
-	AlsaData *ad = newAlsaData();
-
-	if (!free_global_registered) {
-		atexit((void(*)(void))snd_config_update_free_global);
-		free_global_registered = 1;
-	}
-
-	if (param)
-		alsa_configure(ad, param);
-
-	return ad;
-}
-
-static void alsa_finishDriver(void *data)
-{
-	AlsaData *ad = data;
-
-	freeAlsaData(ad);
-}
-
-static int alsa_testDefault(void)
-{
-	snd_pcm_t *handle;
-
-	int ret = snd_pcm_open(&handle, default_device,
-	                       SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
-	if (ret) {
-		WARNING("Error opening default ALSA device: %s\n",
-			snd_strerror(-ret));
-		return -1;
-	} else
-		snd_pcm_close(handle);
-
-	return 0;
-}
-
-static snd_pcm_format_t get_bitformat(const struct audio_format *af)
-{
-	switch (af->bits) {
-	case 8: return SND_PCM_FORMAT_S8;
-	case 16: return SND_PCM_FORMAT_S16;
-	case 24: return SND_PCM_FORMAT_S24;
-	case 32: return SND_PCM_FORMAT_S32;
-	}
-	return SND_PCM_FORMAT_UNKNOWN;
-}
-
-static int alsa_openDevice(void *data, struct audio_format *audioFormat)
-{
-	AlsaData *ad = data;
-	snd_pcm_format_t bitformat;
-	snd_pcm_hw_params_t *hwparams;
-	snd_pcm_sw_params_t *swparams;
-	unsigned int sample_rate = audioFormat->sample_rate;
-	unsigned int channels = audioFormat->channels;
-	snd_pcm_uframes_t alsa_buffer_size;
-	snd_pcm_uframes_t alsa_period_size;
-	int err;
-	const char *cmd = NULL;
-	int retry = MPD_ALSA_RETRY_NR;
-	unsigned int period_time, period_time_ro;
-	unsigned int buffer_time;
-
-	if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN)
-		ERROR("ALSA device \"%s\" doesn't support %u bit audio\n",
-		      ad->device, audioFormat->bits);
-
-	err = snd_pcm_open(&ad->pcmHandle, ad->device,
-			   SND_PCM_STREAM_PLAYBACK, ad->mode);
-	if (err < 0) {
-		ad->pcmHandle = NULL;
-		goto error;
-	}
-
-	period_time_ro = period_time = ad->period_time;
-configure_hw:
-	/* configure HW params */
-	snd_pcm_hw_params_alloca(&hwparams);
-
-	cmd = "snd_pcm_hw_params_any";
-	err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
-	if (err < 0)
-		goto error;
-
-	if (ad->useMmap) {
-		err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
-						   SND_PCM_ACCESS_MMAP_INTERLEAVED);
-		if (err < 0) {
-			ERROR("Cannot set mmap'ed mode on ALSA device \"%s\": "
-			      " %s\n", ad->device, snd_strerror(-err));
-			ERROR("Falling back to direct write mode\n");
-			ad->useMmap = 0;
-		} else
-			ad->writei = snd_pcm_mmap_writei;
-	}
-
-	if (!ad->useMmap) {
-		cmd = "snd_pcm_hw_params_set_access";
-		err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
-						   SND_PCM_ACCESS_RW_INTERLEAVED);
-		if (err < 0)
-			goto error;
-		ad->writei = snd_pcm_writei;
-	}
-
-	err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
-	if (err == -EINVAL && audioFormat->bits != 16) {
-		/* fall back to 16 bit, let pcm_utils.c do the conversion */
-		err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams,
-						   SND_PCM_FORMAT_S16);
-		if (err == 0) {
-			DEBUG("ALSA device \"%s\": converting %u bit to 16 bit\n",
-			      ad->device, audioFormat->bits);
-			audioFormat->bits = 16;
-		}
-	}
-
-	if (err < 0) {
-		ERROR("ALSA device \"%s\" does not support %u bit audio: "
-		      "%s\n", ad->device, audioFormat->bits, snd_strerror(-err));
-		goto fail;
-	}
-
-	err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
-						  &channels);
-	if (err < 0) {
-		ERROR("ALSA device \"%s\" does not support %i channels: "
-		      "%s\n", ad->device, (int)audioFormat->channels,
-		      snd_strerror(-err));
-		goto fail;
-	}
-	audioFormat->channels = (int8_t)channels;
-
-	err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
-					      &sample_rate, NULL);
-	if (err < 0 || sample_rate == 0) {
-		ERROR("ALSA device \"%s\" does not support %u Hz audio\n",
-		      ad->device, audioFormat->sample_rate);
-		goto fail;
-	}
-	audioFormat->sample_rate = sample_rate;
-
-	if (ad->buffer_time > 0) {
-		buffer_time = ad->buffer_time;
-		cmd = "snd_pcm_hw_params_set_buffer_time_near";
-		err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
-							     &buffer_time, NULL);
-		if (err < 0)
-			goto error;
-	}
-
-	if (period_time_ro > 0) {
-		period_time = period_time_ro;
-		cmd = "snd_pcm_hw_params_set_period_time_near";
-		err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
-							     &period_time, NULL);
-		if (err < 0)
-			goto error;
-	}
-
-	cmd = "snd_pcm_hw_params";
-	err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
-	if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
-		period_time_ro = period_time_ro >> 1;
-		goto configure_hw;
-	} else if (err < 0)
-		goto error;
-	if (retry != MPD_ALSA_RETRY_NR)
-		DEBUG("ALSA period_time set to %d\n", period_time);
-
-	cmd = "snd_pcm_hw_params_get_buffer_size";
-	err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_hw_params_get_period_size";
-	err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
-						NULL);
-	if (err < 0)
-		goto error;
-
-	/* configure SW params */
-	snd_pcm_sw_params_alloca(&swparams);
-
-	cmd = "snd_pcm_sw_params_current";
-	err = snd_pcm_sw_params_current(ad->pcmHandle, swparams);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_start_threshold";
-	err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
-						    alsa_buffer_size -
-						    alsa_period_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_avail_min";
-	err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams,
-					      alsa_period_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params";
-	err = snd_pcm_sw_params(ad->pcmHandle, swparams);
-	if (err < 0)
-		goto error;
-
-	ad->sampleSize = audio_format_frame_size(audioFormat);
-
-	DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at "
-	      "%u Hz\n", ad->device, audioFormat->bits,
-	      channels, sample_rate);
-
-	return 0;
-
-error:
-	if (cmd) {
-		ERROR("Error opening ALSA device \"%s\" (%s): %s\n",
-		      ad->device, cmd, snd_strerror(-err));
-	} else {
-		ERROR("Error opening ALSA device \"%s\": %s\n", ad->device,
-		      snd_strerror(-err));
-	}
-fail:
-	if (ad->pcmHandle)
-		snd_pcm_close(ad->pcmHandle);
-	ad->pcmHandle = NULL;
-	return -1;
-}
-
-static int alsa_errorRecovery(AlsaData * ad, int err)
-{
-	if (err == -EPIPE) {
-		DEBUG("Underrun on ALSA device \"%s\"\n", ad->device);
-	} else if (err == -ESTRPIPE) {
-		DEBUG("ALSA device \"%s\" was suspended\n", ad->device);
-	}
-
-	switch (snd_pcm_state(ad->pcmHandle)) {
-	case SND_PCM_STATE_PAUSED:
-		err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
-		break;
-	case SND_PCM_STATE_SUSPENDED:
-		err = snd_pcm_resume(ad->pcmHandle);
-		if (err == -EAGAIN)
-			return 0;
-		/* fall-through to snd_pcm_prepare: */
-	case SND_PCM_STATE_SETUP:
-	case SND_PCM_STATE_XRUN:
-		err = snd_pcm_prepare(ad->pcmHandle);
-		break;
-	case SND_PCM_STATE_DISCONNECTED:
-		/* so alsa_closeDevice won't try to drain: */
-		snd_pcm_close(ad->pcmHandle);
-		ad->pcmHandle = NULL;
-		break;
-	/* this is no error, so just keep running */
-	case SND_PCM_STATE_RUNNING:
-		err = 0;
-		break;
-	default:
-		/* unknown state, do nothing */
-		break;
-	}
-
-	return err;
-}
-
-static void alsa_dropBufferedAudio(void *data)
-{
-	AlsaData *ad = data;
-
-	alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
-}
-
-static void alsa_closeDevice(void *data)
-{
-	AlsaData *ad = data;
-
-	if (ad->pcmHandle) {
-		if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) {
-			snd_pcm_drain(ad->pcmHandle);
-		}
-		snd_pcm_close(ad->pcmHandle);
-		ad->pcmHandle = NULL;
-	}
-}
-
-static int alsa_playAudio(void *data, const char *playChunk, size_t size)
-{
-	AlsaData *ad = data;
-	int ret;
-
-	size /= ad->sampleSize;
-
-	while (size > 0) {
-		ret = ad->writei(ad->pcmHandle, playChunk, size);
-
-		if (ret == -EAGAIN || ret == -EINTR)
-			continue;
-
-		if (ret < 0) {
-			if (alsa_errorRecovery(ad, ret) < 0) {
-				ERROR("closing ALSA device \"%s\" due to write "
-				      "error: %s\n", ad->device,
-				      snd_strerror(-errno));
-				alsa_closeDevice(ad);
-				return -1;
-			}
-			continue;
-		}
-
-		playChunk += ret * ad->sampleSize;
-		size -= ret;
-	}
-
-	return 0;
-}
-
-const struct audio_output_plugin alsaPlugin = {
-	.name = "alsa",
-	.test_default_device = alsa_testDefault,
-	.init = alsa_initDriver,
-	.finish = alsa_finishDriver,
-	.open = alsa_openDevice,
-	.play = alsa_playAudio,
-	.cancel = alsa_dropBufferedAudio,
-	.close = alsa_closeDevice,
-};
-
-#else /* HAVE ALSA */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
-#endif /* HAVE_ALSA */