--- a/gst_plugins_base/ext/alsa/gstalsasink.c Wed Mar 31 22:03:18 2010 +0300
+++ /dev/null Thu Jan 01 00:00:00 1970 +0000
@@ -1,983 +0,0 @@
-/* GStreamer
- * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
- * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
- *
- * gstalsasink.c:
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library 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
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-alsasink
- * @short_description: play audio to an ALSA device
- * @see_also: alsasrc, alsamixer
- *
- * <refsect2>
- * <para>
- * This element renders raw audio samples using the ALSA api.
- * </para>
- * <title>Example pipelines</title>
- * <para>
- * Play an Ogg/Vorbis file.
- * </para>
- * <programlisting>
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink
- * </programlisting>
- * </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-#include <sys/ioctl.h>
-#include <fcntl.h>
-#include <errno.h>
-#include <unistd.h>
-#include <string.h>
-#include <getopt.h>
-#include <alsa/asoundlib.h>
-
-#include "gstalsa.h"
-#include "gstalsasink.h"
-#include "gstalsadeviceprobe.h"
-
-#include <gst/gst-i18n-plugin.h>
-
-/* elementfactory information */
-static const GstElementDetails gst_alsasink_details =
-GST_ELEMENT_DETAILS ("Audio sink (ALSA)",
- "Sink/Audio",
- "Output to a sound card via ALSA",
- "Wim Taymans <wim@fluendo.com>");
-
-#define DEFAULT_DEVICE "default"
-#define DEFAULT_DEVICE_NAME ""
-#define SPDIF_PERIOD_SIZE 1536
-#define SPDIF_BUFFER_SIZE 15360
-
-enum
-{
- PROP_0,
- PROP_DEVICE,
- PROP_DEVICE_NAME
-};
-
-static void gst_alsasink_init_interfaces (GType type);
-
-GST_BOILERPLATE_FULL (GstAlsaSink, gst_alsasink, GstAudioSink,
- GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces);
-
-static void gst_alsasink_finalise (GObject * object);
-static void gst_alsasink_set_property (GObject * object,
- guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_alsasink_get_property (GObject * object,
- guint prop_id, GValue * value, GParamSpec * pspec);
-
-static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink);
-
-static gboolean gst_alsasink_open (GstAudioSink * asink);
-static gboolean gst_alsasink_prepare (GstAudioSink * asink,
- GstRingBufferSpec * spec);
-static gboolean gst_alsasink_unprepare (GstAudioSink * asink);
-static gboolean gst_alsasink_close (GstAudioSink * asink);
-static guint gst_alsasink_write (GstAudioSink * asink, gpointer data,
- guint length);
-static guint gst_alsasink_delay (GstAudioSink * asink);
-static void gst_alsasink_reset (GstAudioSink * asink);
-
-static gint output_ref; /* 0 */
-static snd_output_t *output; /* NULL */
-static GStaticMutex output_mutex = G_STATIC_MUTEX_INIT;
-
-
-#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-# define ALSA_SINK_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
-#else
-# define ALSA_SINK_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
-#endif
-
-static GstStaticPadTemplate alsasink_sink_factory =
- GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
- "signed = (boolean) { TRUE, FALSE }, "
- "width = (int) 32, "
- "depth = (int) 32, "
- "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
- "audio/x-raw-int, "
- "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
- "signed = (boolean) { TRUE, FALSE }, "
- "width = (int) 24, "
- "depth = (int) 24, "
- "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
- "audio/x-raw-int, "
- "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
- "signed = (boolean) { TRUE, FALSE }, "
- "width = (int) 32, "
- "depth = (int) 24, "
- "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
- "audio/x-raw-int, "
- "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
- "signed = (boolean) { TRUE, FALSE }, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
- "audio/x-raw-int, "
- "signed = (boolean) { TRUE, FALSE }, "
- "width = (int) 8, "
- "depth = (int) 8, "
- "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ];"
- "audio/x-iec958")
- );
-
-static void
-gst_alsasink_finalise (GObject * object)
-{
- GstAlsaSink *sink = GST_ALSA_SINK (object);
-
- g_free (sink->device);
- g_mutex_free (sink->alsa_lock);
-
- g_static_mutex_lock (&output_mutex);
- --output_ref;
- if (output_ref == 0) {
- snd_output_close (output);
- output = NULL;
- }
- g_static_mutex_unlock (&output_mutex);
-
- G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static void
-gst_alsasink_init_interfaces (GType type)
-{
- gst_alsa_type_add_device_property_probe_interface (type);
-}
-
-static void
-gst_alsasink_base_init (gpointer g_class)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_set_details (element_class, &gst_alsasink_details);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&alsasink_sink_factory));
-}
-static void
-gst_alsasink_class_init (GstAlsaSinkClass * klass)
-{
- GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
- GstBaseSinkClass *gstbasesink_class;
- GstBaseAudioSinkClass *gstbaseaudiosink_class;
- GstAudioSinkClass *gstaudiosink_class;
-
- gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
- gstbasesink_class = (GstBaseSinkClass *) klass;
- gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
- gstaudiosink_class = (GstAudioSinkClass *) klass;
-
- parent_class = g_type_class_peek_parent (klass);
-
- gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasink_finalise);
- gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasink_get_property);
- gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasink_set_property);
-
- gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);
-
- gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open);
- gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare);
- gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare);
- gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close);
- gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write);
- gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay);
- gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset);
-
- g_object_class_install_property (gobject_class, PROP_DEVICE,
- g_param_spec_string ("device", "Device",
- "ALSA device, as defined in an asound configuration file",
- DEFAULT_DEVICE, G_PARAM_READWRITE));
-
- g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
- g_param_spec_string ("device-name", "Device name",
- "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
- G_PARAM_READABLE));
-}
-
-static void
-gst_alsasink_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstAlsaSink *sink;
-
- sink = GST_ALSA_SINK (object);
-
- switch (prop_id) {
- case PROP_DEVICE:
- g_free (sink->device);
- sink->device = g_value_dup_string (value);
- /* setting NULL restores the default device */
- if (sink->device == NULL) {
- sink->device = g_strdup (DEFAULT_DEVICE);
- }
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_alsasink_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstAlsaSink *sink;
-
- sink = GST_ALSA_SINK (object);
-
- switch (prop_id) {
- case PROP_DEVICE:
- g_value_set_string (value, sink->device);
- break;
- case PROP_DEVICE_NAME:
- g_value_take_string (value,
- gst_alsa_find_device_name (GST_OBJECT_CAST (sink),
- sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK));
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_alsasink_init (GstAlsaSink * alsasink, GstAlsaSinkClass * g_class)
-{
- GST_DEBUG_OBJECT (alsasink, "initializing alsasink");
-
- alsasink->device = g_strdup (DEFAULT_DEVICE);
- alsasink->handle = NULL;
- alsasink->cached_caps = NULL;
- alsasink->alsa_lock = g_mutex_new ();
-
- g_static_mutex_lock (&output_mutex);
- if (output_ref == 0) {
- snd_output_stdio_attach (&output, stdout, 0);
- ++output_ref;
- }
- g_static_mutex_unlock (&output_mutex);
-}
-
-#define CHECK(call, error) \
-G_STMT_START { \
-if ((err = call) < 0) \
- goto error; \
-} G_STMT_END;
-
-static GstCaps *
-gst_alsasink_getcaps (GstBaseSink * bsink)
-{
- GstElementClass *element_class;
- GstPadTemplate *pad_template;
- GstAlsaSink *sink = GST_ALSA_SINK (bsink);
- GstCaps *caps;
-
- if (sink->handle == NULL) {
- GST_DEBUG_OBJECT (sink, "device not open, using template caps");
- return NULL; /* base class will get template caps for us */
- }
-
- if (sink->cached_caps) {
- GST_LOG_OBJECT (sink, "Returning cached caps");
- return gst_caps_ref (sink->cached_caps);
- }
-
- element_class = GST_ELEMENT_GET_CLASS (sink);
- pad_template = gst_element_class_get_pad_template (element_class, "sink");
- g_return_val_if_fail (pad_template != NULL, NULL);
-
- caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->handle,
- gst_pad_template_get_caps (pad_template));
-
- if (caps) {
- sink->cached_caps = gst_caps_ref (caps);
- }
-
- GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);
-
- return caps;
-}
-
-static int
-set_hwparams (GstAlsaSink * alsa)
-{
- guint rrate;
- gint err, dir;
- snd_pcm_hw_params_t *params;
- guint period_time, buffer_time;
-
- snd_pcm_hw_params_malloc (¶ms);
-
- GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "
- "SPDIF (%d)", alsa->channels, alsa->rate,
- snd_pcm_format_name (alsa->format), alsa->iec958);
-
- /* start with requested values, if we cannot configure alsa for those values,
- * we set these values to -1, which will leave the default alsa values */
- buffer_time = alsa->buffer_time;
- period_time = alsa->period_time;
-
-retry:
- /* choose all parameters */
- CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
- /* set the interleaved read/write format */
- CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
- wrong_access);
- /* set the sample format */
- if (alsa->iec958) {
- /* Try to use big endian first else fallback to le and swap bytes */
- if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {
- alsa->format = SND_PCM_FORMAT_S16_LE;
- alsa->need_swap = TRUE;
- GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");
- } else {
- alsa->need_swap = FALSE;
- }
- }
- CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
- no_sample_format);
- /* set the count of channels */
- CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
- no_channels);
- /* set the stream rate */
- rrate = alsa->rate;
- CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
- no_rate);
- if (rrate != alsa->rate)
- goto rate_match;
-
- /* get and dump some limits */
- {
- guint min, max;
-
- snd_pcm_hw_params_get_buffer_time_min (params, &min, &dir);
- snd_pcm_hw_params_get_buffer_time_max (params, &max, &dir);
-
- GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",
- alsa->buffer_time, min, max);
-
- snd_pcm_hw_params_get_period_time_min (params, &min, &dir);
- snd_pcm_hw_params_get_period_time_max (params, &max, &dir);
-
- GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",
- alsa->period_time, min, max);
-
- snd_pcm_hw_params_get_periods_min (params, &min, &dir);
- snd_pcm_hw_params_get_periods_max (params, &max, &dir);
-
- GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
- }
-
- /* now try to configure the buffer time and period time, if one
- * of those fail, we fall back to the defaults and emit a warning. */
- if (buffer_time != -1 && !alsa->iec958) {
- /* set the buffer time */
- if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
- &buffer_time, &dir)) < 0) {
- GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to set buffer time %i for playback: %s",
- buffer_time, snd_strerror (err)));
- /* disable buffer_time the next round */
- buffer_time = -1;
- goto retry;
- }
- GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);
- }
- if (period_time != -1 && !alsa->iec958) {
- /* set the period time */
- if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
- &period_time, &dir)) < 0) {
- GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to set period time %i for playback: %s",
- period_time, snd_strerror (err)));
- /* disable period_time the next round */
- period_time = -1;
- goto retry;
- }
- GST_DEBUG_OBJECT (alsa, "period time %u", period_time);
- }
-
- /* Set buffer size and period size manually for SPDIF */
- if (G_UNLIKELY (alsa->iec958)) {
- snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;
- snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;
-
- CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,
- &buffer_size), buffer_size);
- CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,
- &period_size, NULL), period_size);
- }
-
- /* write the parameters to device */
- CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
-
- /* now get the configured values */
- CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
- buffer_size);
- CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, &dir),
- period_size);
-
- GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
- alsa->period_size);
-
- snd_pcm_hw_params_free (params);
- return 0;
-
- /* ERRORS */
-no_config:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Broken configuration for playback: no configurations available: %s",
- snd_strerror (err)));
- snd_pcm_hw_params_free (params);
- return err;
- }
-wrong_access:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Access type not available for playback: %s", snd_strerror (err)));
- snd_pcm_hw_params_free (params);
- return err;
- }
-no_sample_format:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Sample format not available for playback: %s", snd_strerror (err)));
- snd_pcm_hw_params_free (params);
- return err;
- }
-no_channels:
- {
- gchar *msg = NULL;
-
- if ((alsa->channels) == 1)
- msg = g_strdup (_("Could not open device for playback in mono mode."));
- if ((alsa->channels) == 2)
- msg = g_strdup (_("Could not open device for playback in stereo mode."));
- if ((alsa->channels) > 2)
- msg =
- g_strdup_printf (_
- ("Could not open device for playback in %d-channel mode."),
- alsa->channels);
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (msg), (snd_strerror (err)));
- g_free (msg);
- snd_pcm_hw_params_free (params);
- return err;
- }
-no_rate:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Rate %iHz not available for playback: %s",
- alsa->rate, snd_strerror (err)));
- return err;
- }
-rate_match:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
- snd_pcm_hw_params_free (params);
- return -EINVAL;
- }
-buffer_size:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to get buffer size for playback: %s", snd_strerror (err)));
- snd_pcm_hw_params_free (params);
- return err;
- }
-period_size:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to get period size for playback: %s", snd_strerror (err)));
- snd_pcm_hw_params_free (params);
- return err;
- }
-set_hw_params:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to set hw params for playback: %s", snd_strerror (err)));
- snd_pcm_hw_params_free (params);
- return err;
- }
-}
-
-static int
-set_swparams (GstAlsaSink * alsa)
-{
- int err;
- snd_pcm_sw_params_t *params;
-
- snd_pcm_sw_params_malloc (¶ms);
-
- /* get the current swparams */
- CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
- /* start the transfer when the buffer is almost full: */
- /* (buffer_size / avail_min) * avail_min */
- CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
- (alsa->buffer_size / alsa->period_size) * alsa->period_size),
- start_threshold);
-
- /* allow the transfer when at least period_size samples can be processed */
- CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
- alsa->period_size), set_avail);
-
-#if GST_CHECK_ALSA_VERSION(1,0,16)
- /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
-#else
- /* align all transfers to 1 sample */
- CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
-#endif
-
- /* write the parameters to the playback device */
- CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
-
- snd_pcm_sw_params_free (params);
- return 0;
-
- /* ERRORS */
-no_config:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to determine current swparams for playback: %s",
- snd_strerror (err)));
- snd_pcm_sw_params_free (params);
- return err;
- }
-start_threshold:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to set start threshold mode for playback: %s",
- snd_strerror (err)));
- snd_pcm_sw_params_free (params);
- return err;
- }
-set_avail:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to set avail min for playback: %s", snd_strerror (err)));
- snd_pcm_sw_params_free (params);
- return err;
- }
-#if !GST_CHECK_ALSA_VERSION(1,0,16)
-set_align:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to set transfer align for playback: %s", snd_strerror (err)));
- snd_pcm_sw_params_free (params);
- return err;
- }
-#endif
-set_sw_params:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Unable to set sw params for playback: %s", snd_strerror (err)));
- snd_pcm_sw_params_free (params);
- return err;
- }
-}
-
-static gboolean
-alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec)
-{
- /* Initialize our boolean */
- alsa->iec958 = FALSE;
-
- switch (spec->type) {
- case GST_BUFTYPE_LINEAR:
- GST_DEBUG_OBJECT (alsa,
- "Linear format : depth=%d, width=%d, sign=%d, bigend=%d", spec->depth,
- spec->width, spec->sign, spec->bigend);
-
- alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
- spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
- break;
- case GST_BUFTYPE_FLOAT:
- switch (spec->format) {
- case GST_FLOAT32_LE:
- alsa->format = SND_PCM_FORMAT_FLOAT_LE;
- break;
- case GST_FLOAT32_BE:
- alsa->format = SND_PCM_FORMAT_FLOAT_BE;
- break;
- case GST_FLOAT64_LE:
- alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
- break;
- case GST_FLOAT64_BE:
- alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
- break;
- default:
- goto error;
- }
- break;
- case GST_BUFTYPE_A_LAW:
- alsa->format = SND_PCM_FORMAT_A_LAW;
- break;
- case GST_BUFTYPE_MU_LAW:
- alsa->format = SND_PCM_FORMAT_MU_LAW;
- break;
- case GST_BUFTYPE_IEC958:
- alsa->format = SND_PCM_FORMAT_S16_BE;
- alsa->iec958 = TRUE;
- break;
- default:
- goto error;
-
- }
- alsa->rate = spec->rate;
- alsa->channels = spec->channels;
- alsa->buffer_time = spec->buffer_time;
- alsa->period_time = spec->latency_time;
- alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
-
- return TRUE;
-
- /* ERRORS */
-error:
- {
- return FALSE;
- }
-}
-
-static gboolean
-gst_alsasink_open (GstAudioSink * asink)
-{
- GstAlsaSink *alsa;
- gint err;
-
- alsa = GST_ALSA_SINK (asink);
-
- CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK), open_error);
- GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device);
-
- return TRUE;
-
- /* ERRORS */
-open_error:
- {
- if (err == -EBUSY) {
- GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
- (_("Could not open audio device for playback. "
- "Device is being used by another application.")),
- ("Device '%s' is busy", alsa->device));
- } else {
- GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
- (_("Could not open audio device for playback.")),
- ("Playback open error on device '%s': %s", alsa->device,
- snd_strerror (err)));
- }
- return FALSE;
- }
-}
-
-static gboolean
-gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
-{
- GstAlsaSink *alsa;
- gint err;
-
- alsa = GST_ALSA_SINK (asink);
-
- if (spec->format == GST_IEC958) {
- snd_pcm_close (alsa->handle);
- alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa));
- if (G_UNLIKELY (!alsa->handle)) {
- goto no_iec958;
- }
- }
-
- if (!alsasink_parse_spec (alsa, spec))
- goto spec_parse;
-
- CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
-
- CHECK (set_hwparams (alsa), hw_params_failed);
- CHECK (set_swparams (alsa), sw_params_failed);
-
- alsa->bytes_per_sample = spec->bytes_per_sample;
- spec->segsize = alsa->period_size * spec->bytes_per_sample;
- spec->segtotal = alsa->buffer_size / alsa->period_size;
-
- {
- snd_output_t *out_buf = NULL;
- char *msg = NULL;
-
- snd_output_buffer_open (&out_buf);
- snd_pcm_dump_hw_setup (alsa->handle, out_buf);
- snd_output_buffer_string (out_buf, &msg);
- GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
- snd_output_close (out_buf);
- snd_output_buffer_open (&out_buf);
- snd_pcm_dump_sw_setup (alsa->handle, out_buf);
- snd_output_buffer_string (out_buf, &msg);
- GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
- snd_output_close (out_buf);
- }
-
- return TRUE;
-
- /* ERRORS */
-no_iec958:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL),
- ("Could not open IEC958 (SPDIF) device for playback"));
- return FALSE;
- }
-spec_parse:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Error parsing spec"));
- return FALSE;
- }
-non_block:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Could not set device to blocking: %s", snd_strerror (err)));
- return FALSE;
- }
-hw_params_failed:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Setting of hwparams failed: %s", snd_strerror (err)));
- return FALSE;
- }
-sw_params_failed:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Setting of swparams failed: %s", snd_strerror (err)));
- return FALSE;
- }
-}
-
-static gboolean
-gst_alsasink_unprepare (GstAudioSink * asink)
-{
- GstAlsaSink *alsa;
- gint err;
-
- alsa = GST_ALSA_SINK (asink);
-
- CHECK (snd_pcm_drop (alsa->handle), drop);
-
- CHECK (snd_pcm_hw_free (alsa->handle), hw_free);
-
- CHECK (snd_pcm_nonblock (alsa->handle, 1), non_block);
-
- return TRUE;
-
- /* ERRORS */
-drop:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Could not drop samples: %s", snd_strerror (err)));
- return FALSE;
- }
-hw_free:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Could not free hw params: %s", snd_strerror (err)));
- return FALSE;
- }
-non_block:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
- ("Could not set device to nonblocking: %s", snd_strerror (err)));
- return FALSE;
- }
-}
-
-static gboolean
-gst_alsasink_close (GstAudioSink * asink)
-{
- GstAlsaSink *alsa = GST_ALSA_SINK (asink);
- gint err;
-
- if (alsa->handle) {
- CHECK (snd_pcm_close (alsa->handle), close_error);
- alsa->handle = NULL;
- }
- gst_caps_replace (&alsa->cached_caps, NULL);
-
- return TRUE;
-
- /* ERRORS */
-close_error:
- {
- GST_ELEMENT_ERROR (alsa, RESOURCE, CLOSE, (NULL),
- ("Playback close error: %s", snd_strerror (err)));
- return FALSE;
- }
-}
-
-
-/*
- * Underrun and suspend recovery
- */
-static gint
-xrun_recovery (GstAlsaSink * alsa, snd_pcm_t * handle, gint err)
-{
- GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
-
- if (err == -EPIPE) { /* under-run */
- err = snd_pcm_prepare (handle);
- if (err < 0)
- GST_WARNING_OBJECT (alsa,
- "Can't recovery from underrun, prepare failed: %s",
- snd_strerror (err));
- return 0;
- } else if (err == -ESTRPIPE) {
- while ((err = snd_pcm_resume (handle)) == -EAGAIN)
- g_usleep (100); /* wait until the suspend flag is released */
-
- if (err < 0) {
- err = snd_pcm_prepare (handle);
- if (err < 0)
- GST_WARNING_OBJECT (alsa,
- "Can't recovery from suspend, prepare failed: %s",
- snd_strerror (err));
- }
- return 0;
- }
- return err;
-}
-
-static guint
-gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
-{
- GstAlsaSink *alsa;
- gint err;
- gint cptr;
- gint16 *ptr = data;
-
- alsa = GST_ALSA_SINK (asink);
-
- if (alsa->iec958 && alsa->need_swap) {
- guint i;
-
- GST_DEBUG_OBJECT (asink, "swapping bytes");
- for (i = 0; i < length / 2; i++) {
- ptr[i] = GUINT16_SWAP_LE_BE (ptr[i]);
- }
- }
-
- GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);
-
- cptr = length / alsa->bytes_per_sample;
-
- GST_ALSA_SINK_LOCK (asink);
- while (cptr > 0) {
- err = snd_pcm_writei (alsa->handle, ptr, cptr);
-
- GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);
- if (err < 0) {
- GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err));
- if (err == -EAGAIN) {
- continue;
- } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
- goto write_error;
- }
- continue;
- }
-
- ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
- cptr -= err;
- }
- GST_ALSA_SINK_UNLOCK (asink);
-
- return length - (cptr * alsa->bytes_per_sample);
-
-write_error:
- {
- GST_ALSA_SINK_UNLOCK (asink);
- return length; /* skip one period */
- }
-}
-
-static guint
-gst_alsasink_delay (GstAudioSink * asink)
-{
- GstAlsaSink *alsa;
- snd_pcm_sframes_t delay;
- int res;
-
- alsa = GST_ALSA_SINK (asink);
-
- res = snd_pcm_delay (alsa->handle, &delay);
- if (G_UNLIKELY (res < 0)) {
- /* on errors, report 0 delay */
- GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
- delay = 0;
- }
- if (G_UNLIKELY (delay < 0)) {
- /* make sure we never return a negative delay */
- GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay");
- delay = 0;
- }
-
- return delay;
-}
-
-static void
-gst_alsasink_reset (GstAudioSink * asink)
-{
- GstAlsaSink *alsa;
- gint err;
-
- alsa = GST_ALSA_SINK (asink);
-
- GST_ALSA_SINK_LOCK (asink);
- GST_DEBUG_OBJECT (alsa, "drop");
- CHECK (snd_pcm_drop (alsa->handle), drop_error);
- GST_DEBUG_OBJECT (alsa, "prepare");
- CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
- GST_DEBUG_OBJECT (alsa, "reset done");
- GST_ALSA_SINK_UNLOCK (asink);
-
- return;
-
- /* ERRORS */
-drop_error:
- {
- GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
- snd_strerror (err));
- GST_ALSA_SINK_UNLOCK (asink);
- return;
- }
-prepare_error:
- {
- GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
- snd_strerror (err));
- GST_ALSA_SINK_UNLOCK (asink);
- return;
- }
-}