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gstreamer/gst/rawparse/gstrawaudioparse.c
Sebastian Dröge 0c79a06dd1 rawparse: Properly align raw audio/video output buffers 
That is, aligned to the basic type for audio and to 32 bytes for video.
Fixes crashes if the raw buffers are passed to SIMD processing functions.

https://bugzilla.gnome.org/show_bug.cgi?id=774428
2016-11-27 11:45:33 +02:00

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/* GStreamer
* Copyright (C) <2016> Carlos Rafael Giani <dv at pseudoterminal dot 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 St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rawaudioparse
*
* This element parses incoming data as raw audio samples and timestamps it.
* It also handles seek queries in said raw audio data, and ensures that output
* buffers contain an integer number of samples, even if the input buffers don't.
* For example, with sample format S16LE and 2 channels, an input buffer of 411
* bytes contains 102.75 samples. rawaudioparse will then output 102 samples
* (= 408 bytes) and keep the remaining 3 bytes. These will then be prepended to
* the next input data.
*
* The element implements the properties and sink caps configuration as specified
* in the #GstRawBaseParse documentation. The properties configuration can be
* modified by using the sample-rate, num-channels, channel-positions, format,
* and pcm-format properties.
*
* Currently, this parser supports raw data in a-law, mu-law, or linear PCM format.
*
* To facilitate operation with the unalignedaudioparse element, rawaudioparse
* supports the "audio/x-unaligned-raw" media type. This is treated identically to
* "audio/x-raw", except that it is used by source elements which do not guarantee
* that the buffers they push out are timestamped and contain an integer amount of
* samples (see the 411 bytes example above). By using a different media type, it
* is guaranteed that unalignedaudioparse is autoplugged, making sure that the
* autoplugged chain does not push unparsed content downstream. The source caps'
* media type with linear PCM data is always "audio/x-raw", even if the sink caps
* use "audio/x-unaligned-raw".
*
* The channel-positions property can be used to set explicit position information
* for each channel. If the array that is passed to this property does not match
* the number of channels indicated by num-channels, then said number of channels
* is updated to the array length. If channel-positions is NULL, then the default
* GStreamer positioning is used. This property is also useful for swapping left
* and right in a stereo signal for example.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 souphttpsrc http://my-dlna-server/track.l16 \
* rawaudioparse ! audioconvert ! audioresample ! autoaudiosink
* ]| Receive L16 data from a DLNA server, parse and timestamp it with
* rawaudioparse, and play it. use-sink-caps is set to true since souphttpsrc
* will set its source pad's caps to audio/x-unaligned-raw for the L16 stream.
* |[
* gst-launch-1.0 filesrc location=audio.raw ! rawaudioparse use-sink-caps=false \
* format=pcm pcm-format=s16le sample-rate=48000 num-channels=2 \
* audioconvert ! audioresample ! autoaudiosink
* ]| Read raw data from a local file and parse it as PCM data with 48000 Hz sample
* rate, signed 16 bit integer samples, and 2 channels. use-sink-caps is set to
* false to ensure the property information is used and the parser does not expect
* audio/x-raw or audio/x-unaligned-raw caps.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
/* FIXME: GValueArray is deprecated, but there is currently no viabla alternative
* See https://bugzilla.gnome.org/show_bug.cgi?id=667228 */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include "gstrawaudioparse.h"
#include "unalignedaudio.h"
GST_DEBUG_CATEGORY_STATIC (raw_audio_parse_debug);
#define GST_CAT_DEFAULT raw_audio_parse_debug
enum
{
PROP_0,
PROP_FORMAT,
PROP_PCM_FORMAT,
PROP_SAMPLE_RATE,
PROP_NUM_CHANNELS,
PROP_INTERLEAVED,
PROP_CHANNEL_POSITIONS
};
#define DEFAULT_FORMAT GST_RAW_AUDIO_PARSE_FORMAT_PCM
#define DEFAULT_PCM_FORMAT GST_AUDIO_FORMAT_S16
#define DEFAULT_SAMPLE_RATE 44100
#define DEFAULT_NUM_CHANNELS 2
#define DEFAULT_INTERLEAVED TRUE
#define GST_RAW_AUDIO_PARSE_CAPS \
GST_AUDIO_CAPS_MAKE(GST_AUDIO_FORMATS_ALL) \
", layout = (string) { interleaved, non-interleaved }; " \
"audio/x-alaw, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; " \
"audio/x-mulaw, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; "
static GstStaticPadTemplate static_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_UNALIGNED_RAW_AUDIO_CAPS "; " GST_RAW_AUDIO_PARSE_CAPS)
);
static GstStaticPadTemplate static_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_RAW_AUDIO_PARSE_CAPS)
);
#define gst_raw_audio_parse_parent_class parent_class
G_DEFINE_TYPE (GstRawAudioParse, gst_raw_audio_parse, GST_TYPE_RAW_BASE_PARSE);
static void gst_raw_audio_parse_set_property (GObject * object, guint prop_id,
GValue const *value, GParamSpec * pspec);
static void gst_raw_audio_parse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_raw_audio_parse_stop (GstBaseParse * parse);
static gboolean gst_raw_audio_parse_set_current_config (GstRawBaseParse *
raw_base_parse, GstRawBaseParseConfig config);
static GstRawBaseParseConfig
gst_raw_audio_parse_get_current_config (GstRawBaseParse * raw_base_parse);
static gboolean gst_raw_audio_parse_set_config_from_caps (GstRawBaseParse *
raw_base_parse, GstRawBaseParseConfig config, GstCaps * caps);
static gboolean gst_raw_audio_parse_get_caps_from_config (GstRawBaseParse *
raw_base_parse, GstRawBaseParseConfig config, GstCaps ** caps);
static gsize gst_raw_audio_parse_get_config_frame_size (GstRawBaseParse *
raw_base_parse, GstRawBaseParseConfig config);
static gboolean gst_raw_audio_parse_is_config_ready (GstRawBaseParse *
raw_base_parse, GstRawBaseParseConfig config);
static gboolean gst_raw_audio_parse_process (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config, GstBuffer * in_data, gsize total_num_in_bytes,
gsize num_valid_in_bytes, GstBuffer ** processed_data);
static gboolean gst_raw_audio_parse_is_unit_format_supported (GstRawBaseParse *
raw_base_parse, GstFormat format);
static void gst_raw_audio_parse_get_units_per_second (GstRawBaseParse *
raw_base_parse, GstFormat format, GstRawBaseParseConfig config,
gsize * units_per_sec_n, gsize * units_per_sec_d);
static gint gst_raw_audio_parse_get_alignment (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config);
static gboolean gst_raw_audio_parse_is_using_sink_caps (GstRawAudioParse *
raw_audio_parse);
static GstRawAudioParseConfig
* gst_raw_audio_parse_get_config_ptr (GstRawAudioParse * raw_audio_parse,
GstRawBaseParseConfig config);
static void gst_raw_audio_parse_init_config (GstRawAudioParseConfig * config);
static gboolean gst_raw_audio_parse_set_config_channels (GstRawAudioParseConfig
* config, guint num_channels, guint64 channel_mask, gboolean set_positions);
static gboolean
gst_raw_audio_parse_update_channel_reordering_flag (GstRawAudioParseConfig *
config);
static void gst_raw_audio_parse_update_config_bpf (GstRawAudioParseConfig *
config);
static gboolean gst_raw_audio_parse_caps_to_config (GstRawAudioParse *
raw_audio_parse, GstCaps * caps, GstRawAudioParseConfig * config);
static gboolean gst_raw_audio_parse_config_to_caps (GstRawAudioParse *
raw_audio_parse, GstCaps ** caps, GstRawAudioParseConfig * config);
static void
gst_raw_audio_parse_class_init (GstRawAudioParseClass * klass)
{
GObjectClass *object_class;
GstElementClass *element_class;
GstBaseParseClass *baseparse_class;
GstRawBaseParseClass *rawbaseparse_class;
GST_DEBUG_CATEGORY_INIT (raw_audio_parse_debug, "rawaudioparse", 0,
"rawaudioparse element");
object_class = G_OBJECT_CLASS (klass);
element_class = GST_ELEMENT_CLASS (klass);
baseparse_class = GST_BASE_PARSE_CLASS (klass);
rawbaseparse_class = GST_RAW_BASE_PARSE_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&static_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&static_src_template));
object_class->set_property =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_set_property);
object_class->get_property =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_get_property);
baseparse_class->stop = GST_DEBUG_FUNCPTR (gst_raw_audio_parse_stop);
rawbaseparse_class->set_current_config =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_set_current_config);
rawbaseparse_class->get_current_config =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_get_current_config);
rawbaseparse_class->set_config_from_caps =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_set_config_from_caps);
rawbaseparse_class->get_caps_from_config =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_get_caps_from_config);
rawbaseparse_class->get_config_frame_size =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_get_config_frame_size);
rawbaseparse_class->is_config_ready =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_is_config_ready);
rawbaseparse_class->process = GST_DEBUG_FUNCPTR (gst_raw_audio_parse_process);
rawbaseparse_class->is_unit_format_supported =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_is_unit_format_supported);
rawbaseparse_class->get_units_per_second =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_get_units_per_second);
rawbaseparse_class->get_alignment =
GST_DEBUG_FUNCPTR (gst_raw_audio_parse_get_alignment);
g_object_class_install_property (object_class,
PROP_FORMAT,
g_param_spec_enum ("format",
"Format",
"Format of the raw audio stream",
gst_raw_audio_parse_format_get_type (),
GST_RAW_AUDIO_PARSE_FORMAT_PCM,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
g_object_class_install_property (object_class,
PROP_PCM_FORMAT,
g_param_spec_enum ("pcm-format",
"PCM format",
"Format of audio samples in PCM stream (ignored if format property is not set to pcm)",
GST_TYPE_AUDIO_FORMAT,
GST_RAW_AUDIO_PARSE_FORMAT_PCM,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
g_object_class_install_property (object_class,
PROP_SAMPLE_RATE,
g_param_spec_int ("sample-rate",
"Sample rate",
"Rate of audio samples in raw stream",
1, INT_MAX,
DEFAULT_SAMPLE_RATE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
g_object_class_install_property (object_class,
PROP_NUM_CHANNELS,
g_param_spec_int ("num-channels",
"Number of channels",
"Number of channels in raw stream",
1, INT_MAX,
DEFAULT_NUM_CHANNELS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
g_object_class_install_property (object_class,
PROP_INTERLEAVED,
g_param_spec_boolean ("interleaved",
"Interleaved layout",
"True if audio has interleaved layout",
DEFAULT_INTERLEAVED, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
g_object_class_install_property (object_class,
PROP_CHANNEL_POSITIONS,
g_param_spec_value_array ("channel-positions",
"Channel positions",
"Channel positions used on the output",
g_param_spec_enum ("channel-position",
"Channel position",
"Channel position of the n-th input",
GST_TYPE_AUDIO_CHANNEL_POSITION,
GST_AUDIO_CHANNEL_POSITION_NONE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
gst_element_class_set_static_metadata (element_class,
"rawaudioparse",
"Codec/Parser/Audio",
"Converts unformatted data streams into timestamped raw audio frames",
"Carlos Rafael Giani <dv@pseudoterminal.org>");
}
static void
gst_raw_audio_parse_init (GstRawAudioParse * raw_audio_parse)
{
/* Setup configs and select which one shall be the current one from the start. */
gst_raw_audio_parse_init_config (&(raw_audio_parse->properties_config));
gst_raw_audio_parse_init_config (&(raw_audio_parse->sink_caps_config));
/* As required by GstRawBaseParse, ensure that the current configuration
* is initially set to be the properties config */
raw_audio_parse->current_config = &(raw_audio_parse->properties_config);
/* Properties config must be valid from the start, so set its ready value
* to TRUE, and make sure its bpf value is valid. */
raw_audio_parse->properties_config.ready = TRUE;
gst_raw_audio_parse_update_config_bpf (&(raw_audio_parse->properties_config));
}
static void
gst_raw_audio_parse_set_property (GObject * object, guint prop_id,
GValue const *value, GParamSpec * pspec)
{
GstBaseParse *base_parse = GST_BASE_PARSE (object);
GstRawBaseParse *raw_base_parse = GST_RAW_BASE_PARSE (object);
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (object);
/* All properties are handled similarly:
* - if the new value is the same as the current value, nothing is done
* - the parser lock is held while the new value is set
* - if the properties config is the current config, the source caps are
* invalidated to ensure that the code in handle_frame pushes a new CAPS
* event out
* - properties that affect the bpf value call the function to update
* the bpf and also call gst_base_parse_set_min_frame_size() to ensure
* that the minimum frame size can hold 1 frame (= one sample for each
* channel)
*/
switch (prop_id) {
case PROP_FORMAT:
{
GstRawAudioParseFormat new_format = g_value_get_enum (value);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
if (new_format != raw_audio_parse->properties_config.format) {
raw_audio_parse->properties_config.format = new_format;
gst_raw_audio_parse_update_config_bpf (&
(raw_audio_parse->properties_config));
if (!gst_raw_audio_parse_is_using_sink_caps (raw_audio_parse)) {
gst_raw_base_parse_invalidate_src_caps (raw_base_parse);
gst_base_parse_set_min_frame_size (base_parse,
raw_audio_parse->properties_config.bpf);
}
}
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
}
case PROP_PCM_FORMAT:
{
GstAudioFormat new_pcm_format = g_value_get_enum (value);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
if (new_pcm_format != raw_audio_parse->properties_config.pcm_format) {
raw_audio_parse->properties_config.pcm_format = new_pcm_format;
gst_raw_audio_parse_update_config_bpf (&
(raw_audio_parse->properties_config));
if (!gst_raw_audio_parse_is_using_sink_caps (raw_audio_parse)) {
gst_raw_base_parse_invalidate_src_caps (raw_base_parse);
gst_base_parse_set_min_frame_size (base_parse,
raw_audio_parse->properties_config.bpf);
}
}
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
}
case PROP_SAMPLE_RATE:
{
guint new_sample_rate = g_value_get_int (value);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
if (new_sample_rate != raw_audio_parse->properties_config.sample_rate) {
raw_audio_parse->properties_config.sample_rate = new_sample_rate;
if (!gst_raw_audio_parse_is_using_sink_caps (raw_audio_parse))
gst_raw_base_parse_invalidate_src_caps (raw_base_parse);
}
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
}
case PROP_NUM_CHANNELS:
{
guint new_num_channels = g_value_get_int (value);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
if (new_num_channels != raw_audio_parse->properties_config.num_channels) {
gst_raw_audio_parse_set_config_channels (&
(raw_audio_parse->properties_config), new_num_channels, 0, TRUE);
raw_audio_parse->properties_config.num_channels = new_num_channels;
gst_raw_audio_parse_update_config_bpf (&
(raw_audio_parse->properties_config));
if (!gst_raw_audio_parse_is_using_sink_caps (raw_audio_parse)) {
gst_raw_base_parse_invalidate_src_caps (raw_base_parse);
gst_base_parse_set_min_frame_size (base_parse,
raw_audio_parse->properties_config.bpf);
}
}
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
}
case PROP_INTERLEAVED:
{
gboolean new_interleaved = g_value_get_boolean (value);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
if (new_interleaved != raw_audio_parse->properties_config.interleaved) {
raw_audio_parse->properties_config.interleaved = new_interleaved;
if (!gst_raw_audio_parse_is_using_sink_caps (raw_audio_parse))
gst_raw_base_parse_invalidate_src_caps (raw_base_parse);
}
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
}
case PROP_CHANNEL_POSITIONS:
{
GValueArray *valarray = g_value_get_boxed (value);
GstRawAudioParseConfig *config = &(raw_audio_parse->properties_config);
/* Sanity check - reject empty arrays */
if ((valarray != NULL) && (valarray->n_values == 0)) {
GST_ELEMENT_ERROR (raw_audio_parse, LIBRARY, SETTINGS,
("channel position property holds an empty array"), (NULL));
break;
}
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
if ((valarray == NULL) && (config->num_channels > 0)) {
/* NULL value given, and number of channels is nonzero.
* Use the default GStreamer positioning. Call
* set_config_channels with the set_positions parameter
* set to TRUE to ensure the position values are filled. */
gst_raw_audio_parse_set_config_channels (&
(raw_audio_parse->properties_config), config->num_channels, 0,
TRUE);
} else {
/* Non-NULL value given. Make sure the channel_positions
* array in the properties config has enough room, and that
* the num_channels value equals the array length. Then copy
* the values from the valarray to channel_positions, and
* produce a copy of that array in case its channel positions
* are not in a valid GStreamer order (to be able to apply
* channel reordering later).
*/
guint i;
if (valarray->n_values != config->num_channels) {
/* Call with set_positions == FALSE to ensure that
* the array is properly allocated but not filled
* (it is filled below) */
gst_raw_audio_parse_set_config_channels (config, valarray->n_values,
0, FALSE);
}
for (i = 0; i < config->num_channels; ++i) {
GValue *val = g_value_array_get_nth (valarray, i);
config->channel_positions[i] = g_value_get_enum (val);
}
gst_raw_audio_parse_update_channel_reordering_flag (config);
}
gst_raw_audio_parse_update_config_bpf (&
(raw_audio_parse->properties_config));
if (!gst_raw_audio_parse_is_using_sink_caps (raw_audio_parse)) {
gst_raw_base_parse_invalidate_src_caps (raw_base_parse);
gst_base_parse_set_min_frame_size (base_parse,
raw_audio_parse->properties_config.bpf);
}
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_raw_audio_parse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (object);
switch (prop_id) {
case PROP_FORMAT:
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
g_value_set_enum (value, raw_audio_parse->properties_config.format);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
case PROP_PCM_FORMAT:
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
g_value_set_enum (value, raw_audio_parse->properties_config.pcm_format);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
case PROP_SAMPLE_RATE:
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
g_value_set_int (value, raw_audio_parse->properties_config.sample_rate);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
case PROP_NUM_CHANNELS:
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
g_value_set_int (value, raw_audio_parse->properties_config.num_channels);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
case PROP_INTERLEAVED:
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
g_value_set_boolean (value,
raw_audio_parse->properties_config.interleaved);
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
break;
case PROP_CHANNEL_POSITIONS:
{
GstRawAudioParseConfig *config;
GValueArray *valarray;
GST_RAW_BASE_PARSE_CONFIG_MUTEX_LOCK (object);
valarray = NULL;
config = &(raw_audio_parse->properties_config);
/* Copy channel positions into the valuearray */
if (config->num_channels > 0) {
guint i;
GValue val = G_VALUE_INIT;
g_assert (config->channel_positions);
g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
valarray = g_value_array_new (config->num_channels);
for (i = 0; i < config->num_channels; ++i) {
g_value_set_enum (&val, config->channel_positions[i]);
g_value_array_insert (valarray, i, &val);
}
g_value_unset (&val);
}
GST_RAW_BASE_PARSE_CONFIG_MUTEX_UNLOCK (object);
/* Pass on ownership to the value array,
* since we don't need it anymore */
g_value_take_boxed (value, valarray);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_raw_audio_parse_stop (GstBaseParse * parse)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (parse);
/* Sink caps config is not ready until caps come in.
* We are stopping processing, the element is being reset,
* so the config has to be un-readied.
* (Since the properties config is not depending on caps,
* its ready status is always TRUE.) */
raw_audio_parse->sink_caps_config.ready = FALSE;
return GST_BASE_PARSE_CLASS (parent_class)->stop (parse);
}
static gboolean
gst_raw_audio_parse_set_current_config (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
switch (config) {
case GST_RAW_BASE_PARSE_CONFIG_PROPERTIES:
raw_audio_parse->current_config = &(raw_audio_parse->properties_config);
break;
case GST_RAW_BASE_PARSE_CONFIG_SINKCAPS:
raw_audio_parse->current_config = &(raw_audio_parse->sink_caps_config);
break;
default:
g_assert_not_reached ();
}
return TRUE;
}
static GstRawBaseParseConfig
gst_raw_audio_parse_get_current_config (GstRawBaseParse * raw_base_parse)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
return gst_raw_audio_parse_is_using_sink_caps (raw_audio_parse) ?
GST_RAW_BASE_PARSE_CONFIG_SINKCAPS : GST_RAW_BASE_PARSE_CONFIG_PROPERTIES;
}
static gboolean
gst_raw_audio_parse_set_config_from_caps (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config, GstCaps * caps)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
return gst_raw_audio_parse_caps_to_config (raw_audio_parse, caps,
gst_raw_audio_parse_get_config_ptr (raw_audio_parse, config));
}
static gboolean
gst_raw_audio_parse_get_caps_from_config (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config, GstCaps ** caps)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
return gst_raw_audio_parse_config_to_caps (raw_audio_parse, caps,
gst_raw_audio_parse_get_config_ptr (raw_audio_parse, config));
}
static gsize
gst_raw_audio_parse_get_config_frame_size (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
return gst_raw_audio_parse_get_config_ptr (raw_audio_parse, config)->bpf;
}
static gboolean
gst_raw_audio_parse_is_config_ready (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
return gst_raw_audio_parse_get_config_ptr (raw_audio_parse, config)->ready;
}
static guint
round_up_pow2 (guint n)
{
n = n - 1;
n = n | (n >> 1);
n = n | (n >> 2);
n = n | (n >> 4);
n = n | (n >> 8);
n = n | (n >> 16);
return n + 1;
}
static gint
gst_raw_audio_parse_get_alignment (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
GstRawAudioParseConfig *config_ptr =
gst_raw_audio_parse_get_config_ptr (raw_audio_parse, config);
gint width;
if (config_ptr->format != GST_RAW_AUDIO_PARSE_FORMAT_PCM)
return 1;
width =
GST_AUDIO_FORMAT_INFO_WIDTH (gst_audio_format_get_info
(config_ptr->pcm_format)) / 8;
width = GST_ROUND_UP_8 (width);
width = round_up_pow2 (width);
return width;
}
static gboolean
gst_raw_audio_parse_process (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config, GstBuffer * in_data, gsize total_num_in_bytes,
gsize num_valid_in_bytes, GstBuffer ** processed_data)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
GstRawAudioParseConfig *config_ptr =
gst_raw_audio_parse_get_config_ptr (raw_audio_parse, config);
if ((config_ptr->format == GST_RAW_AUDIO_PARSE_FORMAT_PCM)
&& config_ptr->needs_channel_reordering) {
/* Need to reorder samples, since they are in an invalid
* channel order. */
GstBuffer *outbuf;
GST_LOG_OBJECT (raw_audio_parse,
"using %" G_GSIZE_FORMAT " bytes out of the %" G_GSIZE_FORMAT
" bytes from the input buffer with reordering", num_valid_in_bytes,
total_num_in_bytes);
outbuf =
gst_buffer_copy_region (in_data,
GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
GST_BUFFER_COPY_META | GST_BUFFER_COPY_MEMORY, 0, num_valid_in_bytes);
gst_audio_buffer_reorder_channels (outbuf,
config_ptr->pcm_format,
config_ptr->num_channels,
config_ptr->channel_positions, config_ptr->reordered_channel_positions);
*processed_data = outbuf;
} else {
/* Nothing needs to be done with the sample data.
* Instruct the baseparse class to just take out_size bytes
* from the input buffer */
GST_LOG_OBJECT (raw_audio_parse,
"using %" G_GSIZE_FORMAT " bytes out of the %" G_GSIZE_FORMAT
" bytes from the input buffer without reordering", num_valid_in_bytes,
total_num_in_bytes);
*processed_data = NULL;
}
return TRUE;
}
static gboolean
gst_raw_audio_parse_is_unit_format_supported (G_GNUC_UNUSED GstRawBaseParse *
raw_base_parse, GstFormat format)
{
switch (format) {
case GST_FORMAT_BYTES:
case GST_FORMAT_DEFAULT:
return TRUE;
default:
return FALSE;
}
}
static void
gst_raw_audio_parse_get_units_per_second (GstRawBaseParse * raw_base_parse,
GstFormat format, GstRawBaseParseConfig config, gsize * units_per_sec_n,
gsize * units_per_sec_d)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
GstRawAudioParseConfig *config_ptr =
gst_raw_audio_parse_get_config_ptr (raw_audio_parse, config);
switch (format) {
case GST_FORMAT_BYTES:
*units_per_sec_n = config_ptr->sample_rate * config_ptr->bpf;
*units_per_sec_d = 1;
break;
case GST_FORMAT_DEFAULT:
*units_per_sec_n = config_ptr->sample_rate;
*units_per_sec_d = 1;
break;
default:
g_assert_not_reached ();
}
}
static gboolean
gst_raw_audio_parse_is_using_sink_caps (GstRawAudioParse * raw_audio_parse)
{
return raw_audio_parse->current_config ==
&(raw_audio_parse->sink_caps_config);
}
static GstRawAudioParseConfig *
gst_raw_audio_parse_get_config_ptr (GstRawAudioParse * raw_audio_parse,
GstRawBaseParseConfig config)
{
g_assert (raw_audio_parse->current_config != NULL);
switch (config) {
case GST_RAW_BASE_PARSE_CONFIG_PROPERTIES:
return &(raw_audio_parse->properties_config);
case GST_RAW_BASE_PARSE_CONFIG_SINKCAPS:
return &(raw_audio_parse->sink_caps_config);
default:
g_assert (raw_audio_parse->current_config != NULL);
return raw_audio_parse->current_config;
}
}
static void
gst_raw_audio_parse_init_config (GstRawAudioParseConfig * config)
{
config->ready = FALSE;
config->format = DEFAULT_FORMAT;
config->pcm_format = DEFAULT_PCM_FORMAT;
config->bpf = 0;
config->sample_rate = DEFAULT_SAMPLE_RATE;
config->num_channels = DEFAULT_NUM_CHANNELS;
config->interleaved = DEFAULT_INTERLEAVED;
config->needs_channel_reordering = FALSE;
gst_raw_audio_parse_set_config_channels (config, config->num_channels, 0,
TRUE);
}
static gboolean
gst_raw_audio_parse_set_config_channels (GstRawAudioParseConfig * config,
guint num_channels, guint64 channel_mask, gboolean set_positions)
{
g_assert (num_channels > 0);
config->num_channels = num_channels;
/* Setting this to FALSE, since initially, after setting the channels,
* the default GStreamer channel ordering is used. */
config->needs_channel_reordering = FALSE;
/* Set the channel positions based on the given channel mask if set_positions
* is set to TRUE. A channel mask of 0 signifies that a fallback mask should be
* used for the given number of channels. */
if (set_positions) {
if (channel_mask == 0)
channel_mask = gst_audio_channel_get_fallback_mask (config->num_channels);
return gst_audio_channel_positions_from_mask (config->num_channels,
channel_mask, config->channel_positions);
} else {
return TRUE;
}
}
static gboolean
gst_raw_audio_parse_update_channel_reordering_flag (GstRawAudioParseConfig *
config)
{
g_assert (config->num_channels > 0);
/* If the channel_positions array contains channel positions which are in an
* order that conforms to the valid GStreamer order, ensure that channel
* reordering is disabled.
* Otherwise, if the order of the positions in the channel_positions array
* does not conform to the GStreamer order, ensure it is enabled.
*/
if (gst_audio_check_valid_channel_positions (config->channel_positions,
config->num_channels, TRUE)) {
config->needs_channel_reordering = FALSE;
return TRUE;
} else {
config->needs_channel_reordering = TRUE;
memcpy (config->reordered_channel_positions, config->channel_positions,
sizeof (GstAudioChannelPosition) * config->num_channels);
return
gst_audio_channel_positions_to_valid_order
(config->reordered_channel_positions, config->num_channels);
}
}
static void
gst_raw_audio_parse_update_config_bpf (GstRawAudioParseConfig * config)
{
switch (config->format) {
case GST_RAW_AUDIO_PARSE_FORMAT_PCM:
{
GstAudioFormatInfo const *fmt_info =
gst_audio_format_get_info (config->pcm_format);
g_assert (fmt_info != NULL);
config->bpf =
GST_AUDIO_FORMAT_INFO_WIDTH (fmt_info) * config->num_channels / 8;
break;
}
case GST_RAW_AUDIO_PARSE_FORMAT_ALAW:
case GST_RAW_AUDIO_PARSE_FORMAT_MULAW:
/* A-law and mu-law both use 1 byte per sample */
config->bpf = 1 * config->num_channels;
break;
default:
g_assert_not_reached ();
}
}
static gboolean
gst_raw_audio_parse_caps_to_config (GstRawAudioParse * raw_audio_parse,
GstCaps * caps, GstRawAudioParseConfig * config)
{
gboolean ret = FALSE;
GstStructure *structure;
/* Caps might get copied, and the copy needs to be unref'd.
* Also, the caller retains ownership over the original caps.
* So, to make this mechanism also work with cases where the
* caps are *not* copied, ref the original caps here first. */
gst_caps_ref (caps);
structure = gst_caps_get_structure (caps, 0);
/* For unaligned raw data, the output caps stay the same,
* except that audio/x-unaligned-raw becomes audio/x-raw,
* since the parser aligns the sample data */
if (gst_structure_has_name (structure, "audio/x-unaligned-raw")) {
/* Copy the caps to be able to modify them */
GstCaps *new_caps = gst_caps_copy (caps);
gst_caps_unref (caps);
caps = new_caps;
/* Change the media type to audio/x-raw , otherwise
* gst_audio_info_from_caps() won't work */
structure = gst_caps_get_structure (caps, 0);
gst_structure_set_name (structure, "audio/x-raw");
}
if (gst_structure_has_name (structure, "audio/x-raw")) {
guint num_channels;
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (raw_audio_parse,
"failed to parse caps %" GST_PTR_FORMAT, (gpointer) caps);
goto done;
}
num_channels = GST_AUDIO_INFO_CHANNELS (&info);
config->format = GST_RAW_AUDIO_PARSE_FORMAT_PCM;
config->pcm_format = GST_AUDIO_INFO_FORMAT (&info);
config->bpf = GST_AUDIO_INFO_BPF (&info);
config->sample_rate = GST_AUDIO_INFO_RATE (&info);
config->interleaved =
(GST_AUDIO_INFO_LAYOUT (&info) == GST_AUDIO_LAYOUT_INTERLEAVED);
gst_raw_audio_parse_set_config_channels (config, num_channels, 0, FALSE);
memcpy (config->channel_positions, &(GST_AUDIO_INFO_POSITION (&info, 0)),
sizeof (GstAudioChannelPosition) * num_channels);
} else if (gst_structure_has_name (structure, "audio/x-alaw")
|| gst_structure_has_name (structure, "audio/x-mulaw")) {
gint i;
guint64 channel_mask;
guint num_channels;
config->format =
gst_structure_has_name (structure,
"audio/x-alaw") ? GST_RAW_AUDIO_PARSE_FORMAT_ALAW :
GST_RAW_AUDIO_PARSE_FORMAT_MULAW;
if (!gst_structure_get_int (structure, "rate", &i)) {
GST_ERROR_OBJECT (raw_audio_parse,
"missing rate value in caps %" GST_PTR_FORMAT, (gpointer) caps);
goto done;
}
config->sample_rate = i;
if (!gst_structure_get_int (structure, "channels", &i)) {
GST_ERROR_OBJECT (raw_audio_parse,
"missing channels value in caps %" GST_PTR_FORMAT, (gpointer) caps);
goto done;
}
num_channels = i;
if (!gst_structure_get (structure, "channel-mask", GST_TYPE_BITMASK,
&channel_mask, NULL)) {
channel_mask = gst_audio_channel_get_fallback_mask (num_channels);
GST_DEBUG_OBJECT (raw_audio_parse,
"input caps have no channel mask - using fallback mask %#"
G_GINT64_MODIFIER "x for %u channels", channel_mask, num_channels);
}
if (!gst_raw_audio_parse_set_config_channels (config, num_channels,
channel_mask, TRUE)) {
GST_ERROR_OBJECT (raw_audio_parse,
"could not use channel mask %#" G_GINT64_MODIFIER
"x for channel positions", channel_mask);
goto done;
}
/* A-law and mu-law both use 1 byte per sample */
config->bpf = 1 * num_channels;
} else {
GST_ERROR_OBJECT (raw_audio_parse,
"caps %" GST_PTR_FORMAT " have an unsupported media type",
(gpointer) caps);
goto done;
}
ret = TRUE;
done:
gst_caps_unref (caps);
if (ret)
config->ready = TRUE;
return ret;
}
static gboolean
gst_raw_audio_parse_config_to_caps (GstRawAudioParse * raw_audio_parse,
GstCaps ** caps, GstRawAudioParseConfig * config)
{
gboolean ret = TRUE;
GstAudioChannelPosition *channel_positions;
g_assert (caps != NULL);
if (config->bpf == 0) {
GST_ERROR_OBJECT (raw_audio_parse,
"cannot convert config to caps - config not filled with valid values");
*caps = NULL;
return FALSE;
}
channel_positions =
config->needs_channel_reordering ? &(config->
reordered_channel_positions[0]) : &(config->channel_positions[0]);
switch (config->format) {
case GST_RAW_AUDIO_PARSE_FORMAT_PCM:
{
GstAudioInfo info;
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
config->pcm_format,
config->sample_rate, config->num_channels, channel_positions);
*caps = gst_audio_info_to_caps (&info);
break;
}
case GST_RAW_AUDIO_PARSE_FORMAT_ALAW:
case GST_RAW_AUDIO_PARSE_FORMAT_MULAW:
{
guint64 channel_mask;
if (!gst_audio_channel_positions_to_mask (channel_positions,
config->num_channels, TRUE, &channel_mask)) {
GST_ERROR_OBJECT (raw_audio_parse, "invalid channel positions");
ret = FALSE;
break;
}
*caps = gst_caps_new_simple (
(config->format ==
GST_RAW_AUDIO_PARSE_FORMAT_ALAW) ? "audio/x-alaw" :
"audio/x-mulaw", "rate", G_TYPE_INT, config->sample_rate, "channels",
G_TYPE_INT, config->num_channels, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
break;
}
default:
g_assert_not_reached ();
ret = FALSE;
}
if (!ret)
*caps = NULL;
return ret;
}
GType
gst_raw_audio_parse_format_get_type (void)
{
static GType audio_parse_format_gtype = 0;
static const GEnumValue types[] = {
{GST_RAW_AUDIO_PARSE_FORMAT_PCM, "PCM", "pcm"},
{GST_RAW_AUDIO_PARSE_FORMAT_ALAW, "A-Law", "alaw"},
{GST_RAW_AUDIO_PARSE_FORMAT_MULAW, "\302\265-Law", "mulaw"},
{0, NULL, NULL}
};
if (!audio_parse_format_gtype)
audio_parse_format_gtype =
g_enum_register_static ("GstRawAudioParseFormat", types);
return audio_parse_format_gtype;
}
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