mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-06 01:19:38 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-base/ext/alsa/gstalsa.c
Carlos Rafael Giani fd25e24217 alsa: Add support for DSD audio 
Code is partially based on the DSD of Robert Tiemann <rtie@gmx.de>:
https://gitlab.freedesktop.org/rtiemann/gstreamer/-/tree/dsd

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3901>
2023-06-23 01:27:04 +00:00

1064 lines
32 KiB
C
Raw Blame History

/* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
*
* 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.
*/
#include "gstalsa.h"
#include <gst/audio/audio.h>
#include <gst/audio/gstdsd.h>
static GstCaps *
gst_alsa_detect_rates (GstObject * obj, snd_pcm_hw_params_t * hw_params,
GstCaps * in_caps)
{
GstCaps *caps;
guint min, max;
gint err, dir, min_rate, max_rate, i;
GST_LOG_OBJECT (obj, "probing sample rates ...");
if ((err = snd_pcm_hw_params_get_rate_min (hw_params, &min, &dir)) < 0)
goto min_rate_err;
if ((err = snd_pcm_hw_params_get_rate_max (hw_params, &max, &dir)) < 0)
goto max_rate_err;
min_rate = min;
max_rate = max;
if (min_rate < 4000)
min_rate = 4000; /* random 'sensible minimum' */
if (max_rate <= 0)
max_rate = G_MAXINT; /* or maybe just use 192400 or so? */
else if (max_rate > 0 && max_rate < 4000)
max_rate = MAX (4000, min_rate);
GST_DEBUG_OBJECT (obj, "Min. rate = %u (%d)", min_rate, min);
GST_DEBUG_OBJECT (obj, "Max. rate = %u (%d)", max_rate, max);
caps = gst_caps_make_writable (in_caps);
for (i = 0; i < gst_caps_get_size (caps); ++i) {
GstStructure *s;
s = gst_caps_get_structure (caps, i);
if (min_rate == max_rate) {
gst_structure_set (s, "rate", G_TYPE_INT, min_rate, NULL);
} else {
gst_structure_set (s, "rate", GST_TYPE_INT_RANGE,
min_rate, max_rate, NULL);
}
}
return caps;
/* ERRORS */
min_rate_err:
{
GST_ERROR_OBJECT (obj, "failed to query minimum sample rate: %s",
snd_strerror (err));
gst_caps_unref (in_caps);
return NULL;
}
max_rate_err:
{
GST_ERROR_OBJECT (obj, "failed to query maximum sample rate: %s",
snd_strerror (err));
gst_caps_unref (in_caps);
return NULL;
}
}
static snd_pcm_format_t
gst_alsa_get_pcm_format (GstAudioFormat fmt)
{
switch (fmt) {
case GST_AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
case GST_AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
/* 16 bit */
case GST_AUDIO_FORMAT_S16LE:
return SND_PCM_FORMAT_S16_LE;
case GST_AUDIO_FORMAT_S16BE:
return SND_PCM_FORMAT_S16_BE;
case GST_AUDIO_FORMAT_U16LE:
return SND_PCM_FORMAT_U16_LE;
case GST_AUDIO_FORMAT_U16BE:
return SND_PCM_FORMAT_U16_BE;
/* 24 bit in low 3 bytes of 32 bits */
case GST_AUDIO_FORMAT_S24_32LE:
return SND_PCM_FORMAT_S24_LE;
case GST_AUDIO_FORMAT_S24_32BE:
return SND_PCM_FORMAT_S24_BE;
case GST_AUDIO_FORMAT_U24_32LE:
return SND_PCM_FORMAT_U24_LE;
case GST_AUDIO_FORMAT_U24_32BE:
return SND_PCM_FORMAT_U24_BE;
/* 24 bit in 3 bytes */
case GST_AUDIO_FORMAT_S24LE:
return SND_PCM_FORMAT_S24_3LE;
case GST_AUDIO_FORMAT_S24BE:
return SND_PCM_FORMAT_S24_3BE;
case GST_AUDIO_FORMAT_U24LE:
return SND_PCM_FORMAT_U24_3LE;
case GST_AUDIO_FORMAT_U24BE:
return SND_PCM_FORMAT_U24_3BE;
/* 32 bit */
case GST_AUDIO_FORMAT_S32LE:
return SND_PCM_FORMAT_S32_LE;
case GST_AUDIO_FORMAT_S32BE:
return SND_PCM_FORMAT_S32_BE;
case GST_AUDIO_FORMAT_U32LE:
return SND_PCM_FORMAT_U32_LE;
case GST_AUDIO_FORMAT_U32BE:
return SND_PCM_FORMAT_U32_BE;
case GST_AUDIO_FORMAT_F32LE:
return SND_PCM_FORMAT_FLOAT_LE;
case GST_AUDIO_FORMAT_F32BE:
return SND_PCM_FORMAT_FLOAT_BE;
case GST_AUDIO_FORMAT_F64LE:
return SND_PCM_FORMAT_FLOAT64_LE;
case GST_AUDIO_FORMAT_F64BE:
return SND_PCM_FORMAT_FLOAT64_BE;
default:
break;
}
return SND_PCM_FORMAT_UNKNOWN;
}
static gboolean
format_supported (const GValue * format_val, snd_pcm_format_mask_t * mask,
int endianness)
{
const GstAudioFormatInfo *finfo;
snd_pcm_format_t pcm_format;
GstAudioFormat format;
if (!G_VALUE_HOLDS_STRING (format_val))
return FALSE;
format = gst_audio_format_from_string (g_value_get_string (format_val));
if (format == GST_AUDIO_FORMAT_UNKNOWN)
return FALSE;
finfo = gst_audio_format_get_info (format);
if (GST_AUDIO_FORMAT_INFO_ENDIANNESS (finfo) != endianness
&& GST_AUDIO_FORMAT_INFO_ENDIANNESS (finfo) != 0)
return FALSE;
pcm_format = gst_alsa_get_pcm_format (format);
if (pcm_format == SND_PCM_FORMAT_UNKNOWN)
return FALSE;
return snd_pcm_format_mask_test (mask, pcm_format);
}
static GstCaps *
gst_alsa_detect_formats (GstObject * obj, snd_pcm_hw_params_t * hw_params,
GstCaps * in_caps, int endianness)
{
snd_pcm_format_mask_t *mask;
GstStructure *s;
GstCaps *caps;
gint i;
snd_pcm_format_mask_malloc (&mask);
snd_pcm_hw_params_get_format_mask (hw_params, mask);
caps = NULL;
for (i = 0; i < gst_caps_get_size (in_caps); ++i) {
const GValue *format;
GValue list = G_VALUE_INIT;
s = gst_caps_get_structure (in_caps, i);
if (!gst_structure_has_name (s, "audio/x-raw")) {
GST_DEBUG_OBJECT (obj, "skipping non-raw format");
continue;
}
format = gst_structure_get_value (s, "format");
if (format == NULL)
continue;
g_value_init (&list, GST_TYPE_LIST);
if (GST_VALUE_HOLDS_LIST (format)) {
gint i, len;
len = gst_value_list_get_size (format);
for (i = 0; i < len; i++) {
const GValue *val;
val = gst_value_list_get_value (format, i);
if (format_supported (val, mask, endianness))
gst_value_list_append_value (&list, val);
}
} else if (G_VALUE_HOLDS_STRING (format)) {
if (format_supported (format, mask, endianness))
gst_value_list_append_value (&list, format);
}
if (gst_value_list_get_size (&list) > 1) {
if (caps == NULL)
caps = gst_caps_new_empty ();
s = gst_structure_copy (s);
gst_structure_take_value (s, "format", &list);
gst_caps_append_structure (caps, s);
} else if (gst_value_list_get_size (&list) == 1) {
if (caps == NULL)
caps = gst_caps_new_empty ();
format = gst_value_list_get_value (&list, 0);
s = gst_structure_copy (s);
gst_structure_set_value (s, "format", format);
gst_caps_append_structure (caps, s);
g_value_unset (&list);
} else {
g_value_unset (&list);
}
}
snd_pcm_format_mask_free (mask);
gst_caps_unref (in_caps);
return caps;
}
/* Notes about what the "rate" means in DSD:
*
* In DSD, "sample formats" don't actually exist. There is only the DSD bit;
* this is what could be considered the closest equivalent to a "sample format".
* But since it is impractical to deal with individual bits in software, the
* bits are typically grouped into words (8/16/32 bit words). These are the
* DSDU8, DSDU16LE etc. "grouping formats".
*
* The "rate" in DSD information refers to the number of DSD _bytes_ per second
* (not bits per second, because, as said, per-bit handling in software does
* not usually make sense). ALSA however interprets "rate" as the number of
* DSD _words_ per minute. If the word format is DSDU8, then there's no difference.
* But if for example it is DSDU16LE, then ALSA's rate is half of the rate
* from GstDsdInfo. For this reason, before setting the rate in the ALSA
* hw params, it is essential to divide the rate from the DSD info by the
* word length (in bytes).
*/
typedef struct
{
snd_pcm_format_t alsa_format;
const char *gstreamer_format_name;
} DsdFormatInfo;
static GstCaps *
gst_alsa_detect_dsd_formats (GstObject * obj, snd_pcm_hw_params_t * hw_params)
{
snd_pcm_format_mask_t *mask;
GValue format_list_value = G_VALUE_INIT;
gint table_idx;
gboolean dsd_is_supported = FALSE;
GstCaps *caps = NULL;
const DsdFormatInfo format_table[] = {
{SND_PCM_FORMAT_DSD_U8, "DSDU8"},
{SND_PCM_FORMAT_DSD_U16_LE, "DSDU16LE"},
{SND_PCM_FORMAT_DSD_U16_BE, "DSDU16BE"},
{SND_PCM_FORMAT_DSD_U32_LE, "DSDU32LE"},
{SND_PCM_FORMAT_DSD_U32_BE, "DSDU32BE"}
};
const gint format_table_size = sizeof (format_table) / sizeof (DsdFormatInfo);
g_value_init (&format_list_value, GST_TYPE_LIST);
snd_pcm_format_mask_malloc (&mask);
snd_pcm_hw_params_get_format_mask (hw_params, mask);
for (table_idx = 0; table_idx < format_table_size; ++table_idx) {
const DsdFormatInfo *format_info = &(format_table[table_idx]);
gboolean format_supported = snd_pcm_format_mask_test (mask,
format_info->alsa_format);
GST_DEBUG_OBJECT (obj, "%s supported: %s",
format_info->gstreamer_format_name, format_supported ? "yes" : "no");
if (format_supported) {
GValue format_value = G_VALUE_INIT;
g_value_init (&format_value, G_TYPE_STRING);
g_value_set_string (&format_value, format_info->gstreamer_format_name);
gst_value_list_append_and_take_value (&format_list_value, &format_value);
dsd_is_supported = TRUE;
}
}
if (dsd_is_supported) {
GstStructure *structure;
structure = gst_structure_new_empty ("audio/x-dsd");
/* As a small optimization, if we only support exactly one
* format, store it directly instead of an 1-item list. */
if (gst_value_list_get_size (&format_list_value) == 1) {
const GValue *supported_format_value =
gst_value_list_get_value (&format_list_value, 0);
gst_structure_set_value (structure, "format", supported_format_value);
g_value_unset (&format_list_value);
} else
gst_structure_take_value (structure, "format", &format_list_value);
caps = gst_caps_new_full (structure, NULL);
} else {
g_value_unset (&format_list_value);
}
snd_pcm_format_mask_free (mask);
return caps;
}
static GstCaps *
gst_alsa_detect_dsd_rates (GstObject * obj, snd_pcm_t * handle,
snd_pcm_hw_params_t * hw_params, GstCaps * in_caps)
{
GstCaps *caps = NULL;
guint min_rate, max_rate;
gint err, dir, caps_idx;
int cur_dsd_multiplier;
gboolean keep_testing_rates;
GValue rate_list_value = G_VALUE_INIT;
GValue rate_value = G_VALUE_INIT;
GST_LOG_OBJECT (obj, "probing DSD sample rates ...");
g_value_init (&rate_list_value, GST_TYPE_LIST);
g_value_init (&rate_value, G_TYPE_INT);
if ((err = snd_pcm_hw_params_get_rate_min (hw_params, &min_rate, &dir)) < 0)
goto min_rate_err;
if ((err = snd_pcm_hw_params_get_rate_max (hw_params, &max_rate, &dir)) < 0)
goto max_rate_err;
/* In DSD, valid rates are an integer multiple of 44100 (DSD-44x) or
* 48000 (DSD-48x), and those multipliers must themselves be a power of
* 2. For example, "DSD64-44x" means 64*44100 = 2822400 bits per second.
* In software, we use bytes, so DSD64-44x equals 2822400/8 = 352800 bytes
* per second. DSD64 is the lowest valid rate. The next higher valid rate
* would be DSD128-4x, and DSD256-44x after that etc. DSD200-44x is not
* valid, for example. For this reason, it makes sense to check for the
* individual valid rates that lie within the range defined by min_rate
* and max_rate. */
cur_dsd_multiplier = ((gint64) min_rate) * 8 / 44100;
/* Multipliers below 64 are not valid. If the hardware can't handle
* at least DSD64-44x, we can't play DSD, so this is a good starting
* point for the rate tests below. */
if (cur_dsd_multiplier < 64)
cur_dsd_multiplier = 64;
keep_testing_rates = TRUE;
while (keep_testing_rates) {
const int rates_to_test[] = {
GST_DSD_MAKE_DSD_RATE_44x (cur_dsd_multiplier),
GST_DSD_MAKE_DSD_RATE_48x (cur_dsd_multiplier)
};
const gchar *rates_desc[] = { "44x", "48x" };
int i;
for (i = 0; i < G_N_ELEMENTS (rates_to_test); ++i) {
int rate_to_test = rates_to_test[i];
if (rate_to_test > max_rate) {
keep_testing_rates = FALSE;
break;
}
if (snd_pcm_hw_params_test_rate (handle, hw_params, rate_to_test, 0) == 0) {
GST_DEBUG_OBJECT (obj,
"DSD%d-%s available (equals rate of %d DSD bytes per second)",
cur_dsd_multiplier, rates_desc[i], rate_to_test);
g_value_set_int (&rate_value, rate_to_test);
gst_value_list_append_value (&rate_list_value, &rate_value);
}
}
cur_dsd_multiplier *= 2;
}
caps = gst_caps_make_writable (in_caps);
if (gst_value_list_get_size (&rate_list_value) == 1) {
/* As a small optimization, if we only support exactly one
* rate, store it directly instead of an 1-item list. */
const GValue *supported_rate_value =
gst_value_list_get_value (&rate_list_value, 0);
for (caps_idx = 0; caps_idx < gst_caps_get_size (caps); ++caps_idx) {
GstStructure *structure = gst_caps_get_structure (caps, caps_idx);
gst_structure_set_value (structure, "rate", supported_rate_value);
}
} else {
for (caps_idx = 0; caps_idx < gst_caps_get_size (caps); ++caps_idx) {
GstStructure *structure = gst_caps_get_structure (caps, caps_idx);
gst_structure_set_value (structure, "rate", &rate_list_value);
}
}
finish:
g_value_unset (&rate_list_value);
g_value_unset (&rate_value);
return caps;
/* ERRORS */
min_rate_err:
{
GST_ERROR_OBJECT (obj, "failed to query minimum sample rate: %s",
snd_strerror (err));
gst_caps_unref (in_caps);
goto finish;
}
max_rate_err:
{
GST_ERROR_OBJECT (obj, "failed to query maximum sample rate: %s",
snd_strerror (err));
gst_caps_unref (in_caps);
goto finish;
}
}
/* we don't have channel mappings for more than this many channels */
#define GST_ALSA_MAX_CHANNELS 8
static GstStructure *
get_channel_free_structure (const GstStructure * in_structure)
{
GstStructure *s = gst_structure_copy (in_structure);
gst_structure_remove_field (s, "channels");
return s;
}
#define ONE_64 G_GUINT64_CONSTANT (1)
#define CHANNEL_MASK_STEREO ((ONE_64<<GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT) | (ONE_64<<GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT))
#define CHANNEL_MASK_2_1 (CHANNEL_MASK_STEREO | (ONE_64<<GST_AUDIO_CHANNEL_POSITION_LFE1))
#define CHANNEL_MASK_4_0 (CHANNEL_MASK_STEREO | (ONE_64<<GST_AUDIO_CHANNEL_POSITION_REAR_LEFT) | (ONE_64<<GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT))
#define CHANNEL_MASK_5_1 (CHANNEL_MASK_4_0 | (ONE_64<<GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER) | (ONE_64<<GST_AUDIO_CHANNEL_POSITION_LFE1))
#define CHANNEL_MASK_7_1 (CHANNEL_MASK_5_1 | (ONE_64<<GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT) | (ONE_64<<GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT))
static GstCaps *
caps_add_channel_configuration (GstCaps * caps,
const GstStructure * in_structure, gint min_chans, gint max_chans)
{
GstStructure *s = NULL;
gint c;
if (min_chans == max_chans && max_chans == 1) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
caps = gst_caps_merge_structure (caps, s);
return caps;
}
g_assert (min_chans >= 1);
/* mono and stereo don't need channel configurations */
if (min_chans == 2) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, 2, "channel-mask",
GST_TYPE_BITMASK, CHANNEL_MASK_STEREO, NULL);
caps = gst_caps_merge_structure (caps, s);
} else if (min_chans == 1 && max_chans >= 2) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, 2, "channel-mask",
GST_TYPE_BITMASK, CHANNEL_MASK_STEREO, NULL);
caps = gst_caps_merge_structure (caps, s);
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
caps = gst_caps_merge_structure (caps, s);
}
/* don't know whether to use 2.1 or 3.0 here - but I suspect
* alsa might work around that/fix it somehow. Can we tell alsa
* what our channel layout is like? */
if (max_chans >= 3 && min_chans <= 3) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, 3, "channel-mask",
GST_TYPE_BITMASK, CHANNEL_MASK_2_1, NULL);
caps = gst_caps_merge_structure (caps, s);
}
/* everything else (4, 6, 8 channels) needs a channel layout */
for (c = MAX (4, min_chans); c <= 8; c += 2) {
if (max_chans >= c) {
guint64 channel_mask;
s = get_channel_free_structure (in_structure);
switch (c) {
case 4:
channel_mask = CHANNEL_MASK_4_0;
break;
case 6:
channel_mask = CHANNEL_MASK_5_1;
break;
case 8:
channel_mask = CHANNEL_MASK_7_1;
break;
default:
channel_mask = 0;
g_assert_not_reached ();
break;
}
gst_structure_set (s, "channels", G_TYPE_INT, c, "channel-mask",
GST_TYPE_BITMASK, channel_mask, NULL);
caps = gst_caps_merge_structure (caps, s);
}
}
/* NONE layouts for everything else */
for (c = MAX (9, min_chans); c <= max_chans; ++c) {
s = get_channel_free_structure (in_structure);
gst_structure_set (s, "channels", G_TYPE_INT, c, "channel-mask",
GST_TYPE_BITMASK, G_GUINT64_CONSTANT (0), NULL);
caps = gst_caps_merge_structure (caps, s);
}
return caps;
}
static GstCaps *
gst_alsa_detect_channels (GstObject * obj, snd_pcm_hw_params_t * hw_params,
GstCaps * in_caps)
{
GstCaps *caps;
guint min, max;
gint min_chans, max_chans;
gint err, i;
GST_LOG_OBJECT (obj, "probing channels ...");
if ((err = snd_pcm_hw_params_get_channels_min (hw_params, &min)) < 0)
goto min_chan_error;
if ((err = snd_pcm_hw_params_get_channels_max (hw_params, &max)) < 0)
goto max_chan_error;
/* note: the above functions may return (guint) -1 */
min_chans = min;
max_chans = max;
if (min_chans < 0) {
min_chans = 1;
max_chans = GST_ALSA_MAX_CHANNELS;
} else if (max_chans < 0) {
max_chans = GST_ALSA_MAX_CHANNELS;
}
if (min_chans > max_chans) {
gint temp;
GST_WARNING_OBJECT (obj, "minimum channels > maximum channels (%d > %d), "
"please fix your soundcard drivers", min, max);
temp = min_chans;
min_chans = max_chans;
max_chans = temp;
}
/* pro cards seem to return large numbers for min_channels */
if (min_chans > GST_ALSA_MAX_CHANNELS) {
GST_DEBUG_OBJECT (obj, "min_chans = %u, looks like a pro card", min_chans);
if (max_chans < min_chans) {
max_chans = min_chans;
} else {
/* only support [max_chans; max_chans] for these cards for now
* to avoid inflating the source caps with loads of structures ... */
min_chans = max_chans;
}
} else {
min_chans = MAX (min_chans, 1);
max_chans = MIN (GST_ALSA_MAX_CHANNELS, max_chans);
}
GST_DEBUG_OBJECT (obj, "Min. channels = %d (%d)", min_chans, min);
GST_DEBUG_OBJECT (obj, "Max. channels = %d (%d)", max_chans, max);
caps = gst_caps_new_empty ();
for (i = 0; i < gst_caps_get_size (in_caps); ++i) {
GstStructure *s;
GType field_type;
gint c_min = min_chans;
gint c_max = max_chans;
s = gst_caps_get_structure (in_caps, i);
/* the template caps might limit the number of channels (like alsasrc),
* in which case we don't want to return a superset, so hack around this
* for the two common cases where the channels are either a fixed number
* or a min/max range). Example: alsasrc template has channels = [1,2] and
* the detection will claim to support 8 channels for device 'plughw:0' */
field_type = gst_structure_get_field_type (s, "channels");
if (field_type == G_TYPE_INT) {
gst_structure_get_int (s, "channels", &c_min);
gst_structure_get_int (s, "channels", &c_max);
} else if (field_type == GST_TYPE_INT_RANGE) {
const GValue *val;
val = gst_structure_get_value (s, "channels");
c_min = CLAMP (gst_value_get_int_range_min (val), min_chans, max_chans);
c_max = CLAMP (gst_value_get_int_range_max (val), min_chans, max_chans);
} else {
c_min = min_chans;
c_max = max_chans;
}
caps = caps_add_channel_configuration (caps, s, c_min, c_max);
}
gst_caps_unref (in_caps);
return caps;
/* ERRORS */
min_chan_error:
{
GST_ERROR_OBJECT (obj, "failed to query minimum channel count: %s",
snd_strerror (err));
return NULL;
}
max_chan_error:
{
GST_ERROR_OBJECT (obj, "failed to query maximum channel count: %s",
snd_strerror (err));
return NULL;
}
}
snd_pcm_t *
gst_alsa_open_iec958_pcm (GstObject * obj, gchar * device)
{
char *iec958_pcm_name = NULL;
snd_pcm_t *pcm = NULL;
int res;
char devstr[256]; /* Storage for local 'default' device string */
/*
* Try and open our default iec958 device. Fall back to searching on card x
* if this fails, which should only happen on older alsa setups
*/
/* The string will be one of these:
* SPDIF_CON: Non-audio flag not set:
* spdif:{AES0 0x0 AES1 0x82 AES2 0x0 AES3 0x2}
* SPDIF_CON: Non-audio flag set:
* spdif:{AES0 0x2 AES1 0x82 AES2 0x0 AES3 0x2}
*/
sprintf (devstr,
"%s:{AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x}",
device,
IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO,
IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
0, IEC958_AES3_CON_FS_48000);
GST_DEBUG_OBJECT (obj, "Generated device string \"%s\"", devstr);
iec958_pcm_name = devstr;
res = snd_pcm_open (&pcm, iec958_pcm_name, SND_PCM_STREAM_PLAYBACK, 0);
if (G_UNLIKELY (res < 0)) {
GST_DEBUG_OBJECT (obj, "failed opening IEC958 device: %s",
snd_strerror (res));
pcm = NULL;
}
return pcm;
}
/*
* gst_alsa_probe_supported_formats:
*
* Takes the template caps and returns the subset which is actually
* supported by this device.
*
*/
GstCaps *
gst_alsa_probe_supported_formats (GstObject * obj, gchar * device,
snd_pcm_t * handle, const GstCaps * template_caps)
{
snd_pcm_hw_params_t *hw_params;
snd_pcm_stream_t stream_type;
GstCaps *caps;
GstCaps *dsd_caps;
gint err;
snd_pcm_hw_params_malloc (&hw_params);
if ((err = snd_pcm_hw_params_any (handle, hw_params)) < 0)
goto error;
stream_type = snd_pcm_stream (handle);
/* Try detecting PCM */
caps = gst_alsa_detect_formats (obj, hw_params,
gst_caps_copy (template_caps), G_BYTE_ORDER);
/* if there are no formats in native endianness, try non-native as well */
if (caps == NULL) {
GST_INFO_OBJECT (obj, "no PCM formats in native endianness detected");
caps = gst_alsa_detect_formats (obj, hw_params,
gst_caps_copy (template_caps),
(G_BYTE_ORDER == G_LITTLE_ENDIAN) ? G_BIG_ENDIAN : G_LITTLE_ENDIAN);
if (caps == NULL) {
GST_ERROR_OBJECT (obj, "failed to detect PCM formats");
goto subroutine_error;
}
}
if (!(caps = gst_alsa_detect_rates (obj, hw_params, caps))) {
GST_ERROR_OBJECT (obj, "failed to detect PCM rates");
goto subroutine_error;
}
if (!(caps = gst_alsa_detect_channels (obj, hw_params, caps))) {
GST_ERROR_OBJECT (obj, "failed to detect PCM channels");
goto subroutine_error;
}
/* Try detecting DSD */
dsd_caps = gst_alsa_detect_dsd_formats (obj, hw_params);
if (dsd_caps != NULL) {
GST_INFO_OBJECT (obj, "DSD support detected");
if (!(dsd_caps =
gst_alsa_detect_dsd_rates (obj, handle, hw_params, dsd_caps))) {
GST_ERROR_OBJECT (obj, "failed to detect DSD rates");
goto subroutine_error;
}
if (!(dsd_caps = gst_alsa_detect_channels (obj, hw_params, dsd_caps))) {
GST_ERROR_OBJECT (obj, "failed to detect DSD channels");
goto subroutine_error;
}
gst_caps_append (caps, dsd_caps);
} else {
GST_INFO_OBJECT (obj, "DSD support not detected");
}
/* Try opening IEC958 device to see if we can support that format (playback
* only for now but we could add SPDIF capture later) */
if (stream_type == SND_PCM_STREAM_PLAYBACK) {
snd_pcm_t *pcm = gst_alsa_open_iec958_pcm (obj, device);
if (G_LIKELY (pcm)) {
gst_caps_append (caps, gst_caps_from_string (PASSTHROUGH_CAPS));
snd_pcm_close (pcm);
}
}
snd_pcm_hw_params_free (hw_params);
return caps;
/* ERRORS */
error:
{
GST_ERROR_OBJECT (obj, "failed to query formats: %s", snd_strerror (err));
snd_pcm_hw_params_free (hw_params);
return NULL;
}
subroutine_error:
{
GST_ERROR_OBJECT (obj, "failed to query formats");
snd_pcm_hw_params_free (hw_params);
gst_caps_replace (&caps, NULL);
return NULL;
}
}
/* returns the card name when the device number is unknown or -1 */
static gchar *
gst_alsa_find_device_name_no_handle (GstObject * obj, const gchar * devcard,
gint device_num, snd_pcm_stream_t stream)
{
snd_ctl_card_info_t *info = NULL;
snd_ctl_t *ctl = NULL;
gchar *ret = NULL;
gint dev = -1;
GST_LOG_OBJECT (obj, "[%s] device=%d", devcard, device_num);
if (snd_ctl_open (&ctl, devcard, 0) < 0)
return NULL;
snd_ctl_card_info_malloc (&info);
if (snd_ctl_card_info (ctl, info) < 0)
goto done;
if (device_num != -1) {
while (snd_ctl_pcm_next_device (ctl, &dev) == 0 && dev >= 0) {
if (dev == device_num) {
snd_pcm_info_t *pcminfo;
snd_pcm_info_malloc (&pcminfo);
snd_pcm_info_set_device (pcminfo, dev);
snd_pcm_info_set_subdevice (pcminfo, 0);
snd_pcm_info_set_stream (pcminfo, stream);
if (snd_ctl_pcm_info (ctl, pcminfo) < 0) {
snd_pcm_info_free (pcminfo);
break;
}
ret = (gchar *) snd_pcm_info_get_name (pcminfo);
if (ret) {
ret = g_strdup (ret);
GST_LOG_OBJECT (obj, "name from pcminfo: %s", ret);
}
snd_pcm_info_free (pcminfo);
if (ret)
break;
}
}
}
if (ret == NULL) {
char *name = NULL;
gint card;
GST_LOG_OBJECT (obj, "trying card name");
card = snd_ctl_card_info_get_card (info);
snd_card_get_name (card, &name);
ret = g_strdup (name);
free (name);
}
done:
snd_ctl_card_info_free (info);
snd_ctl_close (ctl);
return ret;
}
gchar *
gst_alsa_find_card_name (GstObject * obj, const gchar * devcard,
snd_pcm_stream_t stream)
{
return gst_alsa_find_device_name_no_handle (obj, devcard, -1, stream);
}
gchar *
gst_alsa_find_device_name (GstObject * obj, const gchar * device,
snd_pcm_t * handle, snd_pcm_stream_t stream)
{
gchar *ret = NULL;
if (device != NULL) {
gchar *dev, *comma;
gint devnum;
GST_LOG_OBJECT (obj, "Trying to get device name from string '%s'", device);
/* only want name:card bit, but not devices and subdevices */
dev = g_strdup (device);
if ((comma = strchr (dev, ','))) {
*comma = '\0';
devnum = atoi (comma + 1);
ret = gst_alsa_find_device_name_no_handle (obj, dev, devnum, stream);
}
g_free (dev);
}
if (ret == NULL && handle != NULL) {
snd_pcm_info_t *info;
GST_LOG_OBJECT (obj, "Trying to get device name from open handle");
snd_pcm_info_malloc (&info);
snd_pcm_info (handle, info);
ret = g_strdup (snd_pcm_info_get_name (info));
snd_pcm_info_free (info);
}
GST_LOG_OBJECT (obj, "Device name for device '%s': %s",
GST_STR_NULL (device), GST_STR_NULL (ret));
return ret;
}
/* ALSA channel positions */
const GstAudioChannelPosition alsa_position[][8] = {
{
GST_AUDIO_CHANNEL_POSITION_MONO},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE1},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
{
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1},
{
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID,
GST_AUDIO_CHANNEL_POSITION_INVALID},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}
};
#ifdef SND_CHMAP_API_VERSION
/* +1 is to make zero as holes */
#define ITEM(x, y) \
[SND_CHMAP_ ## x] = GST_AUDIO_CHANNEL_POSITION_ ## y + 1
static const GstAudioChannelPosition gst_pos[SND_CHMAP_LAST + 1] = {
ITEM (MONO, MONO),
ITEM (FL, FRONT_LEFT),
ITEM (FR, FRONT_RIGHT),
ITEM (FC, FRONT_CENTER),
ITEM (RL, REAR_LEFT),
ITEM (RR, REAR_RIGHT),
ITEM (RC, REAR_CENTER),
ITEM (LFE, LFE1),
ITEM (SL, SIDE_LEFT),
ITEM (SR, SIDE_RIGHT),
ITEM (FLC, FRONT_LEFT_OF_CENTER),
ITEM (FRC, FRONT_RIGHT_OF_CENTER),
ITEM (FLW, WIDE_LEFT),
ITEM (FRW, WIDE_RIGHT),
ITEM (TC, TOP_CENTER),
ITEM (TFL, TOP_FRONT_LEFT),
ITEM (TFR, TOP_FRONT_RIGHT),
ITEM (TFC, TOP_FRONT_CENTER),
ITEM (TRL, TOP_REAR_LEFT),
ITEM (TRR, TOP_REAR_RIGHT),
ITEM (TRC, TOP_REAR_CENTER),
ITEM (LLFE, LFE1),
ITEM (RLFE, LFE2),
ITEM (BC, BOTTOM_FRONT_CENTER),
ITEM (BLC, BOTTOM_FRONT_LEFT),
ITEM (BRC, BOTTOM_FRONT_LEFT),
};
#undef ITEM
gboolean
alsa_chmap_to_channel_positions (const snd_pcm_chmap_t * chmap,
GstAudioChannelPosition * pos)
{
int c;
gboolean all_mono = TRUE;
for (c = 0; c < chmap->channels; c++) {
if (chmap->pos[c] > SND_CHMAP_LAST)
return FALSE;
pos[c] = gst_pos[chmap->pos[c]];
if (!pos[c])
return FALSE;
pos[c]--;
if (pos[c] != GST_AUDIO_CHANNEL_POSITION_MONO)
all_mono = FALSE;
}
if (all_mono && chmap->channels > 1) {
/* GST_AUDIO_CHANNEL_POSITION_MONO can only be used with 1 channel and
* GST_AUDIO_CHANNEL_POSITION_NONE is meant to be used for position-less
* multi channels.
* Converting as ALSA can only express such configuration by using an array
* full of SND_CHMAP_MONO.
*/
for (c = 0; c < chmap->channels; c++)
pos[c] = GST_AUDIO_CHANNEL_POSITION_NONE;
}
return TRUE;
}
void
alsa_detect_channels_mapping (GstObject * obj, snd_pcm_t * handle,
GstAudioRingBufferSpec * spec, guint channels, GstAudioRingBuffer * buf)
{
snd_pcm_chmap_t *chmap;
GstAudioChannelPosition pos[8];
if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW || channels >= 9)
return;
chmap = snd_pcm_get_chmap (handle);
if (!chmap) {
GST_LOG_OBJECT (obj, "ALSA driver does not implement channels mapping API");
return;
}
if (chmap->channels != channels) {
GST_LOG_OBJECT (obj,
"got channels mapping for %u channels but stream has %u channels; ignoring",
chmap->channels, channels);
goto out;
}
if (alsa_chmap_to_channel_positions (chmap, pos)) {
#ifndef GST_DISABLE_GST_DEBUG
{
gchar *tmp = gst_audio_channel_positions_to_string (pos, channels);
GST_LOG_OBJECT (obj, "got channels mapping %s", tmp);
g_free (tmp);
}
#endif /* GST_DISABLE_GST_DEBUG */
gst_audio_ring_buffer_set_channel_positions (buf, pos);
} else {
GST_LOG_OBJECT (obj, "failed to convert ALSA channels mapping");
}
out:
free (chmap);
}
#endif /* SND_CHMAP_API_VERSION */
Reference in a new issue View git blame Copy permalink