gstreamer/subprojects/gst-plugins-bad/gst/inter/gstinteraudiosrc.c
Philippe Normand acd4202bc0 interaudiosrc: Add audio meta to buffers containing non-interleaved samples
Without this a downstream audioconverter wouldn't be able to map the
GstAudioBuffer prior to conversion.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5416>
2023-09-28 15:14:38 +02:00

495 lines
16 KiB
C

/* GStreamer
* Copyright (C) 2011 David A. Schleef <ds@schleef.org>
*
* 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 Street, Suite 500,
* Boston, MA 02110-1335, USA.
*/
/**
* SECTION:element-interaudiosrc
* @title: gstinteraudiosrc
*
* The interaudiosrc element is an audio source element. It is used
* in connection with a interaudiosink element in a different pipeline.
*
* ## Example launch line
* |[
* gst-launch-1.0 -v interaudiosrc ! queue ! autoaudiosink
* ]|
*
* The interaudiosrc element cannot be used effectively with gst-launch-1.0,
* as it requires a second pipeline in the application to send audio.
* See the gstintertest.c example in the gst-plugins-bad source code for
* more details.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstinteraudiosrc.h"
#include <gst/gst.h>
#include <gst/base/gstbasesrc.h>
#include <gst/audio/audio.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_src_debug_category);
#define GST_CAT_DEFAULT gst_inter_audio_src_debug_category
/* prototypes */
static void gst_inter_audio_src_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_inter_audio_src_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_inter_audio_src_finalize (GObject * object);
static GstCaps *gst_inter_audio_src_get_caps (GstBaseSrc * src,
GstCaps * filter);
static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps);
static gboolean gst_inter_audio_src_start (GstBaseSrc * src);
static gboolean gst_inter_audio_src_stop (GstBaseSrc * src);
static void
gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end);
static GstFlowReturn
gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf);
static gboolean gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query);
static GstCaps *gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps);
enum
{
PROP_0,
PROP_CHANNEL,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
PROP_PERIOD_TIME
};
#define DEFAULT_CHANNEL ("default")
/* pad templates */
static GstStaticPadTemplate gst_inter_audio_src_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
", layout = (string) interleaved")
);
/* class initialization */
#define parent_class gst_inter_audio_src_parent_class
G_DEFINE_TYPE (GstInterAudioSrc, gst_inter_audio_src, GST_TYPE_BASE_SRC);
GST_ELEMENT_REGISTER_DEFINE (interaudiosrc, "interaudiosrc",
GST_RANK_NONE, GST_TYPE_INTER_AUDIO_SRC);
static void
gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_inter_audio_src_debug_category, "interaudiosrc",
0, "debug category for interaudiosrc element");
gst_element_class_add_static_pad_template (element_class,
&gst_inter_audio_src_src_template);
gst_element_class_set_static_metadata (element_class,
"Internal audio source",
"Source/Audio",
"Virtual audio source for internal process communication",
"David Schleef <ds@schleef.org>");
gobject_class->set_property = gst_inter_audio_src_set_property;
gobject_class->get_property = gst_inter_audio_src_get_property;
gobject_class->finalize = gst_inter_audio_src_finalize;
base_src_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_caps);
base_src_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_set_caps);
base_src_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_src_start);
base_src_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_src_stop);
base_src_class->get_times = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_times);
base_src_class->create = GST_DEBUG_FUNCPTR (gst_inter_audio_src_create);
base_src_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_src_query);
base_src_class->fixate = GST_DEBUG_FUNCPTR (gst_inter_audio_src_fixate);
g_object_class_install_property (gobject_class, PROP_CHANNEL,
g_param_spec_string ("channel", "Channel",
"Channel name to match inter src and sink elements",
DEFAULT_CHANNEL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_uint64 ("buffer-time", "Buffer Time",
"Size of audio buffer", 1, G_MAXUINT64, DEFAULT_AUDIO_BUFFER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
g_param_spec_uint64 ("latency-time", "Latency Time",
"Latency as reported by the source",
1, G_MAXUINT64, DEFAULT_AUDIO_LATENCY_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PERIOD_TIME,
g_param_spec_uint64 ("period-time", "Period Time",
"The minimum amount of data to read in each iteration",
1, G_MAXUINT64, DEFAULT_AUDIO_PERIOD_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_inter_audio_src_init (GstInterAudioSrc * interaudiosrc)
{
gst_base_src_set_format (GST_BASE_SRC (interaudiosrc), GST_FORMAT_TIME);
gst_base_src_set_live (GST_BASE_SRC (interaudiosrc), TRUE);
gst_base_src_set_blocksize (GST_BASE_SRC (interaudiosrc), -1);
interaudiosrc->channel = g_strdup (DEFAULT_CHANNEL);
interaudiosrc->buffer_time = DEFAULT_AUDIO_BUFFER_TIME;
interaudiosrc->latency_time = DEFAULT_AUDIO_LATENCY_TIME;
interaudiosrc->period_time = DEFAULT_AUDIO_PERIOD_TIME;
}
void
gst_inter_audio_src_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
switch (property_id) {
case PROP_CHANNEL:
g_free (interaudiosrc->channel);
interaudiosrc->channel = g_value_dup_string (value);
break;
case PROP_BUFFER_TIME:
interaudiosrc->buffer_time = g_value_get_uint64 (value);
break;
case PROP_LATENCY_TIME:
interaudiosrc->latency_time = g_value_get_uint64 (value);
break;
case PROP_PERIOD_TIME:
interaudiosrc->period_time = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_src_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
switch (property_id) {
case PROP_CHANNEL:
g_value_set_string (value, interaudiosrc->channel);
break;
case PROP_BUFFER_TIME:
g_value_set_uint64 (value, interaudiosrc->buffer_time);
break;
case PROP_LATENCY_TIME:
g_value_set_uint64 (value, interaudiosrc->latency_time);
break;
case PROP_PERIOD_TIME:
g_value_set_uint64 (value, interaudiosrc->period_time);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_src_finalize (GObject * object)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
/* clean up object here */
g_free (interaudiosrc->channel);
G_OBJECT_CLASS (gst_inter_audio_src_parent_class)->finalize (object);
}
static GstCaps *
gst_inter_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GstCaps *caps;
GST_DEBUG_OBJECT (interaudiosrc, "get_caps");
if (!interaudiosrc->surface)
return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter);
g_mutex_lock (&interaudiosrc->surface->mutex);
if (interaudiosrc->surface->audio_info.finfo) {
caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
if (filter) {
GstCaps *tmp;
tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
} else {
caps = NULL;
}
g_mutex_unlock (&interaudiosrc->surface->mutex);
if (caps)
return caps;
else
return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter);
}
static gboolean
gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "set_caps");
if (!gst_audio_info_from_caps (&interaudiosrc->info, caps)) {
GST_ERROR_OBJECT (src, "Failed to parse caps %" GST_PTR_FORMAT, caps);
return FALSE;
}
return TRUE;
}
static gboolean
gst_inter_audio_src_start (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "start");
interaudiosrc->surface = gst_inter_surface_get (interaudiosrc->channel);
interaudiosrc->timestamp_offset = 0;
interaudiosrc->n_samples = 0;
g_mutex_lock (&interaudiosrc->surface->mutex);
interaudiosrc->surface->audio_buffer_time = interaudiosrc->buffer_time;
interaudiosrc->surface->audio_latency_time = interaudiosrc->latency_time;
interaudiosrc->surface->audio_period_time = interaudiosrc->period_time;
g_mutex_unlock (&interaudiosrc->surface->mutex);
return TRUE;
}
static gboolean
gst_inter_audio_src_stop (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "stop");
gst_inter_surface_unref (interaudiosrc->surface);
interaudiosrc->surface = NULL;
return TRUE;
}
static void
gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (src, "get_times");
/* for live sources, sync on the timestamp of the buffer */
if (gst_base_src_is_live (src)) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
*start = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
*end = *start + GST_BUFFER_DURATION (buffer);
} else {
if (interaudiosrc->info.rate > 0) {
*end = *start +
gst_util_uint64_scale_int (gst_buffer_get_size (buffer),
GST_SECOND, interaudiosrc->info.rate * interaudiosrc->info.bpf);
}
}
}
}
}
static GstFlowReturn
gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GstCaps *caps;
GstBuffer *buffer;
guint n, bpf;
guint64 period_time;
guint64 period_samples;
GST_DEBUG_OBJECT (interaudiosrc, "create");
buffer = NULL;
caps = NULL;
g_mutex_lock (&interaudiosrc->surface->mutex);
if (interaudiosrc->surface->audio_info.finfo) {
if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info,
&interaudiosrc->info)) {
caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
interaudiosrc->timestamp_offset +=
gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->info.rate);
interaudiosrc->n_samples = 0;
}
}
bpf = interaudiosrc->surface->audio_info.bpf;
period_time = interaudiosrc->surface->audio_period_time;
period_samples =
gst_util_uint64_scale (period_time, interaudiosrc->info.rate, GST_SECOND);
if (bpf > 0)
n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf;
else
n = 0;
if (n > period_samples)
n = period_samples;
if (n > 0) {
buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
n * bpf);
} else {
buffer = gst_buffer_new ();
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
}
g_mutex_unlock (&interaudiosrc->surface->mutex);
if (caps) {
gboolean ret = gst_base_src_set_caps (src, caps);
gst_caps_unref (caps);
if (!ret) {
GST_ERROR_OBJECT (src, "Failed to set caps %" GST_PTR_FORMAT, caps);
if (buffer)
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
}
buffer = gst_buffer_make_writable (buffer);
bpf = interaudiosrc->info.bpf;
if (n < period_samples) {
GstMapInfo map;
GstMemory *mem;
GST_DEBUG_OBJECT (interaudiosrc,
"creating %" G_GUINT64_FORMAT " samples of silence",
period_samples - n);
mem = gst_allocator_alloc (NULL, (period_samples - n) * bpf, NULL);
if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
gst_audio_format_info_fill_silence (interaudiosrc->info.finfo, map.data,
map.size);
gst_memory_unmap (mem, &map);
}
gst_buffer_prepend_memory (buffer, mem);
}
n = period_samples;
/* audioconvert expects an audio meta for planar layout audio inputs. */
if (GST_AUDIO_INFO_LAYOUT (&interaudiosrc->info) ==
GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (buffer, &interaudiosrc->info, n, NULL);
}
GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
GST_BUFFER_DTS (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_PTS (buffer) = interaudiosrc->timestamp_offset +
gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->info.rate);
GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
GST_BUFFER_DURATION (buffer) = interaudiosrc->timestamp_offset +
gst_util_uint64_scale (interaudiosrc->n_samples + n, GST_SECOND,
interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer);
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
if (interaudiosrc->n_samples == 0) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
}
interaudiosrc->n_samples += n;
*buf = buffer;
return GST_FLOW_OK;
}
static gboolean
gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
gboolean ret;
GST_DEBUG_OBJECT (src, "query");
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
GstClockTime min_latency, max_latency;
min_latency = interaudiosrc->latency_time;
max_latency = interaudiosrc->buffer_time;
GST_DEBUG_OBJECT (src,
"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
gst_query_set_latency (query,
gst_base_src_is_live (src), min_latency, max_latency);
ret = TRUE;
break;
}
default:
ret = GST_BASE_SRC_CLASS (gst_inter_audio_src_parent_class)->query (src,
query);
break;
}
return ret;
}
static GstCaps *
gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps)
{
GstStructure *structure;
GST_DEBUG_OBJECT (src, "fixate");
caps = gst_caps_make_writable (caps);
caps = gst_caps_truncate (caps);
structure = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_string (structure, "format", GST_AUDIO_NE (S16));
gst_structure_fixate_field_nearest_int (structure, "channels", 2);
gst_structure_fixate_field_nearest_int (structure, "rate", 48000);
gst_structure_fixate_field_string (structure, "layout", "interleaved");
return caps;
}