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gstreamer/ext/alsa/gstalsasrc.c
Nicolas Dufresne d64a4b7a69 Revert "alsa: Implement a DeviceProvider" 
This reverts commit 69c3c31608.

All devices have the same name, they are duplicated with pulseaudio one
and the provided does not respond to HW being plugged/unplugged. I think
it's not ready for 1.16.
2019-01-18 11:39:02 -05:00

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/* GStreamer
* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
*
* gstalsasrc.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-alsasrc
* @title: alsasrc
* @see_also: alsasink
*
* This element reads data from an audio card using the ALSA API.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v alsasrc ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
* ]|
* Record from a sound card using ALSA and encode to Ogg/Vorbis.
*
*/
#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 "gstalsasrc.h"
#include "gstalsadeviceprobe.h"
#include <gst/gst-i18n-plugin.h>
#ifndef ESTRPIPE
#define ESTRPIPE EPIPE
#endif
#define DEFAULT_PROP_DEVICE "default"
#define DEFAULT_PROP_DEVICE_NAME ""
#define DEFAULT_PROP_CARD_NAME ""
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_CARD_NAME,
PROP_LAST
};
#define gst_alsasrc_parent_class parent_class
G_DEFINE_TYPE (GstAlsaSrc, gst_alsasrc, GST_TYPE_AUDIO_SRC);
static void gst_alsasrc_finalize (GObject * object);
static void gst_alsasrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_alsasrc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_alsasrc_change_state (GstElement * element,
GstStateChange transition);
static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter);
static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
GstAudioRingBufferSpec * spec);
static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
static guint gst_alsasrc_read
(GstAudioSrc * asrc, gpointer data, guint length, GstClockTime * timestamp);
static guint gst_alsasrc_delay (GstAudioSrc * asrc);
static void gst_alsasrc_reset (GstAudioSrc * asrc);
/* AlsaSrc signals and args */
enum
{
LAST_SIGNAL
};
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
#endif
static GstStaticPadTemplate alsasrc_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_FORMATS_ALL ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static void
gst_alsasrc_finalize (GObject * object)
{
GstAlsaSrc *src = GST_ALSA_SRC (object);
g_free (src->device);
g_mutex_clear (&src->alsa_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_alsasrc_class_init (GstAlsaSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
gobject_class->finalize = gst_alsasrc_finalize;
gobject_class->get_property = gst_alsasrc_get_property;
gobject_class->set_property = gst_alsasrc_set_property;
gst_element_class_set_static_metadata (gstelement_class,
"Audio source (ALSA)", "Source/Audio",
"Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
gst_element_class_add_static_pad_template (gstelement_class,
&alsasrc_src_factory);
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
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_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CARD_NAME,
g_param_spec_string ("card-name", "Card name",
"Human-readable name of the sound card",
DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
}
static void
gst_alsasrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAlsaSrc *src;
src = GST_ALSA_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (src->device);
src->device = g_value_dup_string (value);
if (src->device == NULL) {
src->device = g_strdup (DEFAULT_PROP_DEVICE);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_alsasrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAlsaSrc *src;
src = GST_ALSA_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, src->device);
break;
case PROP_DEVICE_NAME:
g_value_take_string (value,
gst_alsa_find_device_name (GST_OBJECT_CAST (src),
src->device, src->handle, SND_PCM_STREAM_CAPTURE));
break;
case PROP_CARD_NAME:
g_value_take_string (value,
gst_alsa_find_card_name (GST_OBJECT_CAST (src),
src->device, SND_PCM_STREAM_CAPTURE));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstAlsaSrc *alsa = GST_ALSA_SRC (element);
GstClock *clk;
switch (transition) {
/* show the compiler that we care */
case GST_STATE_CHANGE_NULL_TO_READY:
case GST_STATE_CHANGE_READY_TO_PAUSED:
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
case GST_STATE_CHANGE_READY_TO_NULL:
case GST_STATE_CHANGE_NULL_TO_NULL:
case GST_STATE_CHANGE_READY_TO_READY:
case GST_STATE_CHANGE_PAUSED_TO_PAUSED:
case GST_STATE_CHANGE_PLAYING_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
alsa->driver_timestamps = FALSE;
clk = gst_element_get_clock (element);
if (clk != NULL) {
if (GST_IS_SYSTEM_CLOCK (clk)) {
gint clocktype;
g_object_get (clk, "clock-type", &clocktype, NULL);
if (clocktype == GST_CLOCK_TYPE_MONOTONIC) {
GST_INFO ("Using driver timestamps !");
alsa->driver_timestamps = TRUE;
}
}
gst_object_unref (clk);
}
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return ret;
}
static void
gst_alsasrc_init (GstAlsaSrc * alsasrc)
{
GST_DEBUG_OBJECT (alsasrc, "initializing");
alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
alsasrc->cached_caps = NULL;
alsasrc->driver_timestamps = FALSE;
g_mutex_init (&alsasrc->alsa_lock);
}
#define CHECK(call, error) \
G_STMT_START { \
if ((err = call) < 0) \
goto error; \
} G_STMT_END;
static GstCaps *
gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstAlsaSrc *src;
GstCaps *caps, *templ_caps;
src = GST_ALSA_SRC (bsrc);
if (src->handle == NULL) {
GST_DEBUG_OBJECT (src, "device not open, using template caps");
return GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
}
if (src->cached_caps) {
GST_LOG_OBJECT (src, "Returning cached caps");
if (filter)
return gst_caps_intersect_full (filter, src->cached_caps,
GST_CAPS_INTERSECT_FIRST);
else
return gst_caps_ref (src->cached_caps);
}
element_class = GST_ELEMENT_GET_CLASS (src);
pad_template = gst_element_class_get_pad_template (element_class, "src");
g_return_val_if_fail (pad_template != NULL, NULL);
templ_caps = gst_pad_template_get_caps (pad_template);
GST_INFO_OBJECT (src, "template caps %" GST_PTR_FORMAT, templ_caps);
caps = gst_alsa_probe_supported_formats (GST_OBJECT (src),
src->device, src->handle, templ_caps);
gst_caps_unref (templ_caps);
if (caps) {
src->cached_caps = gst_caps_ref (caps);
}
GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
return intersection;
} else {
return caps;
}
}
static int
set_hwparams (GstAlsaSrc * alsa)
{
guint rrate;
gint err;
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_malloc (&params);
/* 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 */
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;
#ifndef GST_DISABLE_GST_DEBUG
/* get and dump some limits */
{
guint min, max;
snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);
snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
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, NULL);
snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
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, NULL);
snd_pcm_hw_params_get_periods_max (params, &max, NULL);
GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
}
#endif
if (alsa->buffer_time != -1) {
/* set the buffer time */
CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
&alsa->buffer_time, NULL), buffer_time);
GST_DEBUG_OBJECT (alsa, "buffer time %u", alsa->buffer_time);
}
if (alsa->period_time != -1) {
/* set the period time */
CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
&alsa->period_time, NULL), period_time);
GST_DEBUG_OBJECT (alsa, "period time %u", alsa->period_time);
}
/* write the parameters to device */
CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
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, NULL),
period_size);
snd_pcm_hw_params_free (params);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Broken configuration for recording: 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 recording: %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 recording: %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 recording in mono mode."));
if ((alsa->channels) == 2)
msg = g_strdup (_("Could not open device for recording in stereo mode."));
if ((alsa->channels) > 2)
msg =
g_strdup_printf (_
("Could not open device for recording in %d-channel mode"),
alsa->channels);
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
("%s", 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 recording: %s",
alsa->rate, snd_strerror (err)));
snd_pcm_hw_params_free (params);
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_time:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set buffer time %i for recording: %s",
alsa->buffer_time, snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
buffer_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get buffer size for recording: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
period_time:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set period time %i for recording: %s", alsa->period_time,
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 recording: %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 recording: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
}
static int
set_swparams (GstAlsaSrc * alsa)
{
int err;
snd_pcm_sw_params_t *params;
snd_pcm_sw_params_malloc (&params);
/* get the current swparams */
CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
/* 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);
/* start the transfer on first read */
CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
0), start_threshold);
/* use monotonic timestamping */
CHECK (snd_pcm_sw_params_set_tstamp_mode (alsa->handle, params,
SND_PCM_TSTAMP_MMAP), tstamp_mode);
#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 recording 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;
}
tstamp_mode:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set tstamp mode 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
alsasrc_parse_spec (GstAlsaSrc * alsa, GstAudioRingBufferSpec * spec)
{
switch (spec->type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
case GST_AUDIO_FORMAT_U8:
alsa->format = SND_PCM_FORMAT_U8;
break;
case GST_AUDIO_FORMAT_S8:
alsa->format = SND_PCM_FORMAT_S8;
break;
case GST_AUDIO_FORMAT_S16LE:
alsa->format = SND_PCM_FORMAT_S16_LE;
break;
case GST_AUDIO_FORMAT_S16BE:
alsa->format = SND_PCM_FORMAT_S16_BE;
break;
case GST_AUDIO_FORMAT_U16LE:
alsa->format = SND_PCM_FORMAT_U16_LE;
break;
case GST_AUDIO_FORMAT_U16BE:
alsa->format = SND_PCM_FORMAT_U16_BE;
break;
case GST_AUDIO_FORMAT_S24_32LE:
alsa->format = SND_PCM_FORMAT_S24_LE;
break;
case GST_AUDIO_FORMAT_S24_32BE:
alsa->format = SND_PCM_FORMAT_S24_BE;
break;
case GST_AUDIO_FORMAT_U24_32LE:
alsa->format = SND_PCM_FORMAT_U24_LE;
break;
case GST_AUDIO_FORMAT_U24_32BE:
alsa->format = SND_PCM_FORMAT_U24_BE;
break;
case GST_AUDIO_FORMAT_S32LE:
alsa->format = SND_PCM_FORMAT_S32_LE;
break;
case GST_AUDIO_FORMAT_S32BE:
alsa->format = SND_PCM_FORMAT_S32_BE;
break;
case GST_AUDIO_FORMAT_U32LE:
alsa->format = SND_PCM_FORMAT_U32_LE;
break;
case GST_AUDIO_FORMAT_U32BE:
alsa->format = SND_PCM_FORMAT_U32_BE;
break;
case GST_AUDIO_FORMAT_S24LE:
alsa->format = SND_PCM_FORMAT_S24_3LE;
break;
case GST_AUDIO_FORMAT_S24BE:
alsa->format = SND_PCM_FORMAT_S24_3BE;
break;
case GST_AUDIO_FORMAT_U24LE:
alsa->format = SND_PCM_FORMAT_U24_3LE;
break;
case GST_AUDIO_FORMAT_U24BE:
alsa->format = SND_PCM_FORMAT_U24_3BE;
break;
case GST_AUDIO_FORMAT_S20LE:
alsa->format = SND_PCM_FORMAT_S20_3LE;
break;
case GST_AUDIO_FORMAT_S20BE:
alsa->format = SND_PCM_FORMAT_S20_3BE;
break;
case GST_AUDIO_FORMAT_U20LE:
alsa->format = SND_PCM_FORMAT_U20_3LE;
break;
case GST_AUDIO_FORMAT_U20BE:
alsa->format = SND_PCM_FORMAT_U20_3BE;
break;
case GST_AUDIO_FORMAT_S18LE:
alsa->format = SND_PCM_FORMAT_S18_3LE;
break;
case GST_AUDIO_FORMAT_S18BE:
alsa->format = SND_PCM_FORMAT_S18_3BE;
break;
case GST_AUDIO_FORMAT_U18LE:
alsa->format = SND_PCM_FORMAT_U18_3LE;
break;
case GST_AUDIO_FORMAT_U18BE:
alsa->format = SND_PCM_FORMAT_U18_3BE;
break;
case GST_AUDIO_FORMAT_F32LE:
alsa->format = SND_PCM_FORMAT_FLOAT_LE;
break;
case GST_AUDIO_FORMAT_F32BE:
alsa->format = SND_PCM_FORMAT_FLOAT_BE;
break;
case GST_AUDIO_FORMAT_F64LE:
alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
break;
case GST_AUDIO_FORMAT_F64BE:
alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
break;
default:
goto error;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
alsa->format = SND_PCM_FORMAT_A_LAW;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
alsa->format = SND_PCM_FORMAT_MU_LAW;
break;
default:
goto error;
}
alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
alsa->buffer_time = spec->buffer_time;
alsa->period_time = spec->latency_time;
alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9)
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
return TRUE;
/* ERRORS */
error:
{
return FALSE;
}
}
static gboolean
gst_alsasrc_open (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
(alsa->driver_timestamps) ? 0 : SND_PCM_NONBLOCK), open_error);
return TRUE;
/* ERRORS */
open_error:
{
if (err == -EBUSY) {
GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
(_("Could not open audio device for recording. "
"Device is being used by another application.")),
("Device '%s' is busy", alsa->device));
} else {
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
(_("Could not open audio device for recording.")),
("Recording open error on device '%s': %s", alsa->device,
snd_strerror (err)));
}
return FALSE;
}
}
static gboolean
gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
if (!alsasrc_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);
CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
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);
}
#ifdef SND_CHMAP_API_VERSION
alsa_detect_channels_mapping (GST_OBJECT (alsa), alsa->handle, spec,
alsa->channels, GST_AUDIO_BASE_SRC (alsa)->ringbuffer);
#endif /* SND_CHMAP_API_VERSION */
return TRUE;
/* ERRORS */
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;
}
prepare_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Prepare failed: %s", snd_strerror (err)));
return FALSE;
}
}
static gboolean
gst_alsasrc_unprepare (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
alsa = GST_ALSA_SRC (asrc);
snd_pcm_drop (alsa->handle);
snd_pcm_hw_free (alsa->handle);
snd_pcm_nonblock (alsa->handle, 1);
return TRUE;
}
static gboolean
gst_alsasrc_close (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
snd_pcm_close (alsa->handle);
alsa->handle = NULL;
gst_caps_replace (&alsa->cached_caps, NULL);
return TRUE;
}
/*
* Underrun and suspend recovery
*/
static gint
xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
{
GST_WARNING_OBJECT (alsa, "xrun recovery %d: %s", err, g_strerror (-err));
if (err == -EPIPE) { /* under-run */
err = snd_pcm_prepare (handle);
if (err < 0)
GST_WARNING_OBJECT (alsa,
"Can't recover 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 recover from suspend, prepare failed: %s",
snd_strerror (err));
}
return 0;
}
return err;
}
static GstClockTime
gst_alsasrc_get_timestamp (GstAlsaSrc * asrc)
{
snd_pcm_status_t *status;
snd_htimestamp_t tstamp;
GstClockTime timestamp;
snd_pcm_uframes_t avail;
gint err = -EPIPE;
if (G_UNLIKELY (!asrc)) {
GST_ERROR_OBJECT (asrc, "No alsa handle created yet !");
return GST_CLOCK_TIME_NONE;
}
if (G_UNLIKELY (snd_pcm_status_malloc (&status) != 0)) {
GST_ERROR_OBJECT (asrc, "snd_pcm_status_malloc failed");
return GST_CLOCK_TIME_NONE;
}
if (G_UNLIKELY (snd_pcm_status (asrc->handle, status) != 0)) {
GST_ERROR_OBJECT (asrc, "snd_pcm_status failed");
return GST_CLOCK_TIME_NONE;
}
/* in case an xrun condition has occured we need to handle this */
if (snd_pcm_status_get_state (status) != SND_PCM_STATE_RUNNING) {
if (xrun_recovery (asrc, asrc->handle, err) < 0) {
GST_WARNING_OBJECT (asrc, "Could not recover from xrun condition !");
}
/* reload the status alsa status object, since recovery made it invalid */
if (G_UNLIKELY (snd_pcm_status (asrc->handle, status) != 0)) {
GST_ERROR_OBJECT (asrc, "snd_pcm_status failed");
}
}
/* get high resolution time stamp from driver */
snd_pcm_status_get_htstamp (status, &tstamp);
timestamp = GST_TIMESPEC_TO_TIME (tstamp);
/* max available frames sets the depth of the buffer */
avail = snd_pcm_status_get_avail (status);
/* calculate the timestamp of the next sample to be read */
timestamp -= gst_util_uint64_scale_int (avail, GST_SECOND, asrc->rate);
/* compensate for the fact that we really need the timestamp of the
* previously read data segment */
timestamp -= asrc->period_time * 1000;
snd_pcm_status_free (status);
GST_LOG_OBJECT (asrc, "ALSA timestamp : %" GST_TIME_FORMAT
", delay %lu", GST_TIME_ARGS (timestamp), avail);
return timestamp;
}
static guint
gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
GstAlsaSrc *alsa;
gint err;
gint cptr;
guint8 *ptr = data;
alsa = GST_ALSA_SRC (asrc);
cptr = length / alsa->bpf;
GST_ALSA_SRC_LOCK (asrc);
while (cptr > 0) {
if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
if (err == -EAGAIN) {
GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
continue;
} else if (err == -ENODEV) {
goto device_disappeared;
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
goto read_error;
}
continue;
}
ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
cptr -= err;
}
GST_ALSA_SRC_UNLOCK (asrc);
/* if driver timestamps are enabled we need to return this here */
if (alsa->driver_timestamps && timestamp)
*timestamp = gst_alsasrc_get_timestamp (alsa);
return length - (cptr * alsa->bpf);
read_error:
{
GST_ALSA_SRC_UNLOCK (asrc);
return length; /* skip one period */
}
device_disappeared:
{
GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
(_("Error recording from audio device. "
"The device has been disconnected.")), (NULL));
GST_ALSA_SRC_UNLOCK (asrc);
return (guint) - 1;
}
}
static guint
gst_alsasrc_delay (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
snd_pcm_sframes_t delay;
int res;
alsa = GST_ALSA_SRC (asrc);
res = snd_pcm_delay (alsa->handle, &delay);
if (G_UNLIKELY (res < 0)) {
GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
delay = 0;
}
return CLAMP (delay, 0, alsa->buffer_size);
}
static void
gst_alsasrc_reset (GstAudioSrc * asrc)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
GST_ALSA_SRC_LOCK (asrc);
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_SRC_UNLOCK (asrc);
return;
/* ERRORS */
drop_error:
{
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
snd_strerror (err));
GST_ALSA_SRC_UNLOCK (asrc);
return;
}
prepare_error:
{
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
snd_strerror (err));
GST_ALSA_SRC_UNLOCK (asrc);
return;
}
}
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