mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-23 16:50:47 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-base/gst/audioresample/gstaudioresample.c

1134 lines
38 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
* Copyright (C) 2007-2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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-audioresample
* @title: audioresample
*
* audioresample resamples raw audio buffers to different sample rates using
* a configurable windowing function to enhance quality.
*
* By default, the resampler uses a reduced sinc table, with cubic interpolation filling in
* the gaps. This ensures that the table does not become too big. However, the interpolation
* increases the CPU usage considerably. As an alternative, a full sinc table can be used.
* Doing so can drastically reduce CPU usage (4x faster with 44.1 -> 48 kHz conversions for
* example), at the cost of increased memory consumption, plus the sinc table takes longer
* to initialize when the element is created. A third mode exists, which uses the full table
* unless said table would become too large, in which case the interpolated one is used instead.
*
* ## Example launch line
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! audio/x-raw, rate=8000 ! autoaudiosink
* ]|
* Decode an audio file and downsample it to 8Khz and play sound.
* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
* This assumes there is an audio sink that will accept/handle 8kHz audio.
*
*/
/* TODO:
* - Enable SSE/ARM optimizations and select at runtime
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include "gstaudioresample.h"
#include <gst/gstutils.h>
#include <gst/audio/audio.h>
#include <gst/base/gstbasetransform.h>
GST_DEBUG_CATEGORY (audio_resample_debug);
#define GST_CAT_DEFAULT audio_resample_debug
#undef USE_SPEEX
#define DEFAULT_QUALITY GST_AUDIO_RESAMPLER_QUALITY_DEFAULT
#define DEFAULT_RESAMPLE_METHOD GST_AUDIO_RESAMPLER_METHOD_KAISER
#define DEFAULT_SINC_FILTER_MODE GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO
#define DEFAULT_SINC_FILTER_AUTO_THRESHOLD (1*1048576)
#define DEFAULT_SINC_FILTER_INTERPOLATION GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC
enum
{
PROP_0,
PROP_QUALITY,
PROP_RESAMPLE_METHOD,
PROP_SINC_FILTER_MODE,
PROP_SINC_FILTER_AUTO_THRESHOLD,
PROP_SINC_FILTER_INTERPOLATION
};
#define SUPPORTED_CAPS \
GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
", layout = (string) { interleaved, non-interleaved }"
static GstStaticPadTemplate gst_audio_resample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SUPPORTED_CAPS));
static GstStaticPadTemplate gst_audio_resample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SUPPORTED_CAPS));
/* cached quark to avoid contention on the global quark table lock */
#define META_TAG_AUDIO meta_tag_audio_quark
static GQuark meta_tag_audio_quark;
static void gst_audio_resample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_resample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base,
GstCaps * caps, gsize * size);
static GstCaps *gst_audio_resample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * filter);
static GstCaps *gst_audio_resample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
static gboolean gst_audio_resample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * incaps, gsize insize,
GstCaps * outcaps, gsize * outsize);
static gboolean gst_audio_resample_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean gst_audio_resample_transform_meta (GstBaseTransform * trans,
GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf);
static GstFlowReturn gst_audio_resample_submit_input_buffer (GstBaseTransform *
base, gboolean is_discont, GstBuffer * input);
static gboolean gst_audio_resample_sink_event (GstBaseTransform * base,
GstEvent * event);
static gboolean gst_audio_resample_start (GstBaseTransform * base);
static gboolean gst_audio_resample_stop (GstBaseTransform * base);
static gboolean gst_audio_resample_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static void gst_audio_resample_push_drain (GstAudioResample * resample,
guint history_len);
#define gst_audio_resample_parent_class parent_class
G_DEFINE_TYPE (GstAudioResample, gst_audio_resample, GST_TYPE_BASE_TRANSFORM);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (audioresample, "audioresample",
GST_RANK_PRIMARY, GST_TYPE_AUDIO_RESAMPLE,
GST_DEBUG_CATEGORY_INIT (audio_resample_debug, "audioresample", 0,
"audio resampling element"));
static void
gst_audio_resample_class_init (GstAudioResampleClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audio_resample_set_property;
gobject_class->get_property = gst_audio_resample_get_property;
g_object_class_install_property (gobject_class, PROP_QUALITY,
g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
"the lowest and 10 being the best",
GST_AUDIO_RESAMPLER_QUALITY_MIN, GST_AUDIO_RESAMPLER_QUALITY_MAX,
DEFAULT_QUALITY,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RESAMPLE_METHOD,
g_param_spec_enum ("resample-method", "Resample method to use",
"What resample method to use",
GST_TYPE_AUDIO_RESAMPLER_METHOD,
DEFAULT_RESAMPLE_METHOD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SINC_FILTER_MODE,
g_param_spec_enum ("sinc-filter-mode", "Sinc filter table mode",
"What sinc filter table mode to use",
GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE,
DEFAULT_SINC_FILTER_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_SINC_FILTER_AUTO_THRESHOLD,
g_param_spec_uint ("sinc-filter-auto-threshold",
"Sinc filter auto mode threshold",
"Memory usage threshold to use if sinc filter mode is AUTO, given in bytes",
0, G_MAXUINT, DEFAULT_SINC_FILTER_AUTO_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_SINC_FILTER_INTERPOLATION,
g_param_spec_enum ("sinc-filter-interpolation",
"Sinc filter interpolation",
"How to interpolate the sinc filter table",
GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION,
DEFAULT_SINC_FILTER_INTERPOLATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class,
&gst_audio_resample_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_audio_resample_sink_template);
gst_element_class_set_static_metadata (gstelement_class, "Audio resampler",
"Filter/Converter/Audio", "Resamples audio",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
GST_BASE_TRANSFORM_CLASS (klass)->start =
GST_DEBUG_FUNCPTR (gst_audio_resample_start);
GST_BASE_TRANSFORM_CLASS (klass)->stop =
GST_DEBUG_FUNCPTR (gst_audio_resample_stop);
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
GST_DEBUG_FUNCPTR (gst_audio_resample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
GST_DEBUG_FUNCPTR (gst_audio_resample_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
GST_DEBUG_FUNCPTR (gst_audio_resample_transform_caps);
GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
GST_DEBUG_FUNCPTR (gst_audio_resample_fixate_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
GST_DEBUG_FUNCPTR (gst_audio_resample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
GST_DEBUG_FUNCPTR (gst_audio_resample_transform);
GST_BASE_TRANSFORM_CLASS (klass)->sink_event =
GST_DEBUG_FUNCPTR (gst_audio_resample_sink_event);
GST_BASE_TRANSFORM_CLASS (klass)->transform_meta =
GST_DEBUG_FUNCPTR (gst_audio_resample_transform_meta);
GST_BASE_TRANSFORM_CLASS (klass)->submit_input_buffer =
GST_DEBUG_FUNCPTR (gst_audio_resample_submit_input_buffer);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_RESAMPLER_METHOD, 0);
gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION,
0);
gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE, 0);
meta_tag_audio_quark = g_quark_from_static_string (GST_META_TAG_AUDIO_STR);
}
static void
gst_audio_resample_init (GstAudioResample * resample)
{
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
resample->method = DEFAULT_RESAMPLE_METHOD;
resample->quality = DEFAULT_QUALITY;
resample->sinc_filter_mode = DEFAULT_SINC_FILTER_MODE;
resample->sinc_filter_auto_threshold = DEFAULT_SINC_FILTER_AUTO_THRESHOLD;
resample->sinc_filter_interpolation = DEFAULT_SINC_FILTER_INTERPOLATION;
gst_base_transform_set_gap_aware (trans, TRUE);
gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query);
}
/* vmethods */
static gboolean
gst_audio_resample_start (GstBaseTransform * base)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
resample->need_discont = TRUE;
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
resample->samples_in = 0;
resample->samples_out = 0;
return TRUE;
}
static gboolean
gst_audio_resample_stop (GstBaseTransform * base)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
if (resample->converter) {
gst_audio_converter_free (resample->converter);
resample->converter = NULL;
}
return TRUE;
}
static gboolean
gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
gsize * size)
{
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps))
goto invalid_caps;
*size = GST_AUDIO_INFO_BPF (&info);
return TRUE;
/* ERRORS */
invalid_caps:
{
GST_ERROR_OBJECT (base, "invalid caps");
return FALSE;
}
}
static GstCaps *
gst_audio_resample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * filter)
{
const GValue *val;
GstStructure *s;
GstCaps *res;
gint i, n;
/* transform single caps into input_caps + input_caps with the rate
* field set to our supported range. This ensures that upstream knows
* about downstream's preferred rate(s) and can negotiate accordingly. */
res = gst_caps_new_empty ();
n = gst_caps_get_size (caps);
for (i = 0; i < n; i++) {
s = gst_caps_get_structure (caps, i);
/* If this is already expressed by the existing caps
* skip this structure */
if (i > 0 && gst_caps_is_subset_structure (res, s))
continue;
/* first, however, check if the caps contain a range for the rate field, in
* which case that side isn't going to care much about the exact sample rate
* chosen and we should just assume things will get fixated to something sane
* and we may just as well offer our full range instead of the range in the
* caps. If the rate is not an int range value, it's likely to express a
* real preference or limitation and we should maintain that structure as
* preference by putting it first into the transformed caps, and only add
* our full rate range as second option */
s = gst_structure_copy (s);
val = gst_structure_get_value (s, "rate");
if (val == NULL || GST_VALUE_HOLDS_INT_RANGE (val)) {
/* overwrite existing range, or add field if it doesn't exist yet */
gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
} else {
/* append caps with full range to existing caps with non-range rate field */
gst_caps_append_structure (res, gst_structure_copy (s));
gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
}
gst_caps_append_structure (res, s);
}
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (res);
res = intersection;
}
return res;
}
/* Fixate rate to the allowed rate that has the smallest difference */
static GstCaps *
gst_audio_resample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
{
GstStructure *s;
gint rate;
s = gst_caps_get_structure (caps, 0);
if (G_UNLIKELY (!gst_structure_get_int (s, "rate", &rate)))
return othercaps;
othercaps = gst_caps_truncate (othercaps);
othercaps = gst_caps_make_writable (othercaps);
s = gst_caps_get_structure (othercaps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", rate);
return gst_caps_fixate (othercaps);
}
static GstStructure *
make_options (GstAudioResample * resample, GstAudioInfo * in,
GstAudioInfo * out)
{
GstStructure *options;
options = gst_structure_new_empty ("resampler-options");
if (in != NULL && out != NULL)
gst_audio_resampler_options_set_quality (resample->method,
resample->quality, in->rate, out->rate, options);
gst_structure_set (options,
GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD, GST_TYPE_AUDIO_RESAMPLER_METHOD,
resample->method,
GST_AUDIO_RESAMPLER_OPT_FILTER_MODE, GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE,
resample->sinc_filter_mode, GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD,
G_TYPE_UINT, resample->sinc_filter_auto_threshold,
GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION,
GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION,
resample->sinc_filter_interpolation, NULL);
return options;
}
static gboolean
gst_audio_resample_update_state (GstAudioResample * resample, GstAudioInfo * in,
GstAudioInfo * out)
{
gboolean updated_latency = FALSE;
gsize old_latency = -1;
GstStructure *options;
if (resample->converter == NULL && in == NULL && out == NULL)
return TRUE;
options = make_options (resample, in, out);
if (resample->converter)
old_latency = gst_audio_converter_get_max_latency (resample->converter);
/* if channels and layout changed, destroy existing resampler */
if (in != NULL && (in->finfo != resample->in.finfo ||
in->channels != resample->in.channels ||
in->layout != resample->in.layout) && resample->converter) {
gst_audio_converter_free (resample->converter);
resample->converter = NULL;
}
if (resample->converter == NULL) {
resample->converter =
gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE, in,
out, options);
if (resample->converter == NULL)
goto resampler_failed;
} else if (in && out) {
gboolean ret;
ret =
gst_audio_converter_update_config (resample->converter, in->rate,
out->rate, options);
if (!ret)
goto update_failed;
} else {
gst_structure_free (options);
}
if (old_latency != -1)
updated_latency =
old_latency !=
gst_audio_converter_get_max_latency (resample->converter);
if (updated_latency)
gst_element_post_message (GST_ELEMENT (resample),
gst_message_new_latency (GST_OBJECT (resample)));
return TRUE;
/* ERRORS */
resampler_failed:
{
GST_ERROR_OBJECT (resample, "failed to create resampler");
return FALSE;
}
update_failed:
{
GST_ERROR_OBJECT (resample, "failed to update resampler");
return FALSE;
}
}
static void
gst_audio_resample_reset_state (GstAudioResample * resample)
{
if (resample->converter)
gst_audio_converter_reset (resample->converter);
}
static gboolean
gst_audio_resample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps,
gsize * othersize)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
gboolean ret = TRUE;
gint bpf;
GST_LOG_OBJECT (base, "asked to transform size %" G_GSIZE_FORMAT
" in direction %s", size, direction == GST_PAD_SINK ? "SINK" : "SRC");
/* Number of samples in either buffer is size / (width*channels) ->
* calculate the factor */
bpf = GST_AUDIO_INFO_BPF (&resample->in);
/* Convert source buffer size to samples */
size /= bpf;
if (direction == GST_PAD_SINK) {
/* asked to convert size of an incoming buffer */
*othersize = gst_audio_converter_get_out_frames (resample->converter, size);
*othersize *= bpf;
} else {
/* asked to convert size of an outgoing buffer */
*othersize = gst_audio_converter_get_in_frames (resample->converter, size);
*othersize *= bpf;
}
GST_LOG_OBJECT (base,
"transformed size %" G_GSIZE_FORMAT " to %" G_GSIZE_FORMAT,
size * bpf, *othersize);
return ret;
}
static gboolean
gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
GstAudioInfo in, out;
GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
if (!gst_audio_info_from_caps (&in, incaps))
goto invalid_incaps;
if (!gst_audio_info_from_caps (&out, outcaps))
goto invalid_outcaps;
/* Reset timestamp tracking and drain the resampler if the audio format is
* changing. Especially when changing the sample rate our timestamp tracking
* will be completely off, but even otherwise we would usually lose the last
* few samples if we don't drain here */
if (!gst_audio_info_is_equal (&in, &resample->in) ||
!gst_audio_info_is_equal (&out, &resample->out)) {
if (resample->converter) {
gsize latency = gst_audio_converter_get_max_latency (resample->converter);
gst_audio_resample_push_drain (resample, latency);
}
gst_audio_resample_reset_state (resample);
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
resample->samples_in = 0;
resample->samples_out = 0;
resample->need_discont = TRUE;
}
gst_audio_resample_update_state (resample, &in, &out);
resample->in = in;
resample->out = out;
return TRUE;
/* ERROR */
invalid_incaps:
{
GST_ERROR_OBJECT (base, "invalid incaps");
return FALSE;
}
invalid_outcaps:
{
GST_ERROR_OBJECT (base, "invalid outcaps");
return FALSE;
}
}
/* Push history_len zeros into the filter, but discard the output. */
static void
gst_audio_resample_dump_drain (GstAudioResample * resample, guint history_len)
{
gsize out_len, outsize;
GstBuffer *outbuf;
GstAudioBuffer abuf;
out_len =
gst_audio_converter_get_out_frames (resample->converter, history_len);
if (out_len == 0)
return;
outsize = out_len * resample->out.bpf;
outbuf = gst_buffer_new_and_alloc (outsize);
if (GST_AUDIO_INFO_LAYOUT (&resample->out) ==
GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL);
}
gst_audio_buffer_map (&abuf, &resample->out, outbuf, GST_MAP_WRITE);
gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
abuf.planes, out_len);
gst_audio_buffer_unmap (&abuf);
gst_buffer_unref (outbuf);
}
static void
gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
{
GstBuffer *outbuf;
GstFlowReturn res;
gint outsize;
gsize out_len;
GstAudioBuffer abuf;
g_assert (resample->converter != NULL);
/* Don't drain samples if we were reset. */
if (!GST_CLOCK_TIME_IS_VALID (resample->t0))
return;
out_len =
gst_audio_converter_get_out_frames (resample->converter, history_len);
if (out_len == 0)
return;
outsize = out_len * resample->in.bpf;
outbuf = gst_buffer_new_and_alloc (outsize);
if (GST_AUDIO_INFO_LAYOUT (&resample->out) ==
GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL);
}
gst_audio_buffer_map (&abuf, &resample->out, outbuf, GST_MAP_WRITE);
gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
abuf.planes, out_len);
gst_audio_buffer_unmap (&abuf);
/* time */
if (GST_CLOCK_TIME_IS_VALID (resample->t0)) {
GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 +
gst_util_uint64_scale_int_round (resample->samples_out, GST_SECOND,
resample->out.rate);
GST_BUFFER_DURATION (outbuf) = resample->t0 +
gst_util_uint64_scale_int_round (resample->samples_out + out_len,
GST_SECOND, resample->out.rate) - GST_BUFFER_TIMESTAMP (outbuf);
} else {
GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
}
/* offset */
if (resample->out_offset0 != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = resample->out_offset0 + resample->samples_out;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_len;
} else {
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
}
/* move along */
resample->samples_out += out_len;
resample->samples_in += history_len;
GST_LOG_OBJECT (resample,
"Pushing drain buffer of %u bytes with timestamp %" GST_TIME_FORMAT
" duration %" GST_TIME_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
G_GUINT64_FORMAT, outsize,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf));
res = gst_pad_push (GST_BASE_TRANSFORM_SRC_PAD (resample), outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK))
GST_WARNING_OBJECT (resample, "Failed to push drain: %s",
gst_flow_get_name (res));
return;
}
static gboolean
gst_audio_resample_sink_event (GstBaseTransform * base, GstEvent * event)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_audio_resample_reset_state (resample);
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
resample->samples_in = 0;
resample->samples_out = 0;
resample->need_discont = TRUE;
break;
case GST_EVENT_STREAM_START:
case GST_EVENT_SEGMENT:
case GST_EVENT_EOS:
if (resample->converter) {
gsize latency =
gst_audio_converter_get_max_latency (resample->converter);
gst_audio_resample_push_drain (resample, latency);
}
gst_audio_resample_reset_state (resample);
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
resample->samples_in = 0;
resample->samples_out = 0;
resample->need_discont = TRUE;
break;
default:
break;
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (base, event);
}
static gboolean
gst_audio_resample_check_discont (GstAudioResample * resample, GstBuffer * buf)
{
guint64 offset;
guint64 delta;
/* is the incoming buffer a discontinuity? */
if (G_UNLIKELY (GST_BUFFER_IS_DISCONT (buf)))
return TRUE;
/* no valid timestamps or offsets to compare --> no discontinuity */
if (G_UNLIKELY (!(GST_BUFFER_TIMESTAMP_IS_VALID (buf) &&
GST_CLOCK_TIME_IS_VALID (resample->t0))))
return FALSE;
/* convert the inbound timestamp to an offset. */
offset =
gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf) -
resample->t0, resample->in.rate, GST_SECOND);
/* many elements generate imperfect streams due to rounding errors, so we
* permit a small error (up to one sample) without triggering a filter
* flush/restart (if triggered incorrectly, this will be audible) */
/* allow even up to more samples, since sink is not so strict anyway,
* so give that one a chance to handle this as configured */
delta = ABS ((gint64) (offset - resample->samples_in));
if (delta <= (resample->in.rate >> 5))
return FALSE;
GST_WARNING_OBJECT (resample,
"encountered timestamp discontinuity of %" G_GUINT64_FORMAT " samples = %"
GST_TIME_FORMAT, delta,
GST_TIME_ARGS (gst_util_uint64_scale_int_round (delta, GST_SECOND,
resample->in.rate)));
return TRUE;
}
static GstFlowReturn
gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstAudioBuffer srcabuf, dstabuf;
gsize outsize;
gsize in_len;
gsize out_len;
guint filt_len =
gst_audio_converter_get_max_latency (resample->converter) * 2;
gboolean inbuf_writable;
inbuf_writable = gst_buffer_is_writable (inbuf)
&& gst_buffer_n_memory (inbuf) == 1
&& gst_memory_is_writable (gst_buffer_peek_memory (inbuf, 0));
gst_audio_buffer_map (&srcabuf, &resample->in, inbuf,
inbuf_writable ? GST_MAP_READWRITE : GST_MAP_READ);
in_len = srcabuf.n_samples;
out_len = gst_audio_converter_get_out_frames (resample->converter, in_len);
/* ensure that the output buffer is not bigger than what we need */
gst_buffer_set_size (outbuf, out_len * resample->in.bpf);
if (GST_AUDIO_INFO_LAYOUT (&resample->out) ==
GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL);
}
gst_audio_buffer_map (&dstabuf, &resample->out, outbuf, GST_MAP_WRITE);
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
resample->num_nongap_samples = 0;
if (resample->num_gap_samples < filt_len) {
guint zeros_to_push;
if (in_len >= filt_len - resample->num_gap_samples)
zeros_to_push = filt_len - resample->num_gap_samples;
else
zeros_to_push = in_len;
gst_audio_resample_push_drain (resample, zeros_to_push);
in_len -= zeros_to_push;
resample->num_gap_samples += zeros_to_push;
}
{
guint num, den;
gint i;
num = resample->in.rate;
den = resample->out.rate;
if (resample->samples_in + in_len >= filt_len / 2)
out_len =
gst_util_uint64_scale_int_ceil (resample->samples_in + in_len -
filt_len / 2, den, num) - resample->samples_out;
else
out_len = 0;
for (i = 0; i < dstabuf.n_planes; i++)
memset (dstabuf.planes[i], 0, GST_AUDIO_BUFFER_PLANE_SIZE (&dstabuf));
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
resample->num_gap_samples += in_len;
}
} else { /* not a gap */
if (resample->num_gap_samples > filt_len) {
/* push in enough zeros to restore the filter to the right offset */
guint num;
num = resample->in.rate;
gst_audio_resample_dump_drain (resample,
(resample->num_gap_samples - filt_len) % num);
}
resample->num_gap_samples = 0;
if (resample->num_nongap_samples < filt_len) {
resample->num_nongap_samples += in_len;
if (resample->num_nongap_samples > filt_len)
resample->num_nongap_samples = filt_len;
}
{
/* process */
GstAudioConverterFlags flags;
flags = 0;
if (inbuf_writable)
flags |= GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
gst_audio_converter_samples (resample->converter, flags, srcabuf.planes,
in_len, dstabuf.planes, out_len);
}
}
/* time */
if (GST_CLOCK_TIME_IS_VALID (resample->t0)) {
GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 +
gst_util_uint64_scale_int_round (resample->samples_out, GST_SECOND,
resample->out.rate);
GST_BUFFER_DURATION (outbuf) = resample->t0 +
gst_util_uint64_scale_int_round (resample->samples_out + out_len,
GST_SECOND, resample->out.rate) - GST_BUFFER_TIMESTAMP (outbuf);
} else {
GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
}
/* offset */
if (resample->out_offset0 != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = resample->out_offset0 + resample->samples_out;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_len;
} else {
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
}
/* move along */
resample->samples_out += out_len;
resample->samples_in += in_len;
gst_audio_buffer_unmap (&srcabuf);
gst_audio_buffer_unmap (&dstabuf);
outsize = out_len * resample->in.bpf;
GST_LOG_OBJECT (resample,
"Converted to buffer of %" G_GSIZE_FORMAT
" samples (%" G_GSIZE_FORMAT " bytes) with timestamp %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
", offset_end %" G_GUINT64_FORMAT, out_len, outsize,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
if (outsize == 0)
return GST_BASE_TRANSFORM_FLOW_DROPPED;
else
return GST_FLOW_OK;
}
static GstFlowReturn
gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
GstFlowReturn ret;
GST_LOG_OBJECT (resample, "transforming buffer of %" G_GSIZE_FORMAT " bytes,"
" ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
gst_buffer_get_size (inbuf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
/* check for timestamp discontinuities; flush/reset if needed, and set
* flag to resync timestamp and offset counters and send event
* downstream */
if (G_UNLIKELY (gst_audio_resample_check_discont (resample, inbuf))) {
if (resample->converter) {
gsize latency = gst_audio_converter_get_max_latency (resample->converter);
gst_audio_resample_push_drain (resample, latency);
}
gst_audio_resample_reset_state (resample);
resample->need_discont = TRUE;
}
/* handle discontinuity */
if (G_UNLIKELY (resample->need_discont)) {
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
/* reset */
resample->samples_in = 0;
resample->samples_out = 0;
GST_DEBUG_OBJECT (resample, "found discontinuity; resyncing");
/* resync the timestamp and offset counters if possible */
if (GST_BUFFER_TIMESTAMP_IS_VALID (inbuf)) {
resample->t0 = GST_BUFFER_TIMESTAMP (inbuf);
} else {
GST_DEBUG_OBJECT (resample, "... but new timestamp is invalid");
resample->t0 = GST_CLOCK_TIME_NONE;
}
if (GST_BUFFER_OFFSET_IS_VALID (inbuf)) {
resample->in_offset0 = GST_BUFFER_OFFSET (inbuf);
resample->out_offset0 =
gst_util_uint64_scale_int_round (resample->in_offset0,
resample->out.rate, resample->in.rate);
} else {
GST_DEBUG_OBJECT (resample, "... but new offset is invalid");
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
}
/* set DISCONT flag on output buffer */
GST_DEBUG_OBJECT (resample, "marking this buffer with the DISCONT flag");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
resample->need_discont = FALSE;
}
ret = gst_audio_resample_process (resample, inbuf, outbuf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
return ret;
GST_DEBUG_OBJECT (resample, "input = samples [%" G_GUINT64_FORMAT ", %"
G_GUINT64_FORMAT ") = [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT
") ns; output = samples [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT
") = [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ") ns",
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf),
GST_BUFFER_TIMESTAMP (inbuf), GST_BUFFER_TIMESTAMP (inbuf) +
GST_BUFFER_DURATION (inbuf), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), GST_BUFFER_TIMESTAMP (outbuf),
GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf));
return GST_FLOW_OK;
}
static gboolean
gst_audio_resample_transform_meta (GstBaseTransform * trans, GstBuffer * outbuf,
GstMeta * meta, GstBuffer * inbuf)
{
const GstMetaInfo *info = meta->info;
const gchar *const *tags;
tags = gst_meta_api_type_get_tags (info->api);
if (!tags || (g_strv_length ((gchar **) tags) == 1
&& gst_meta_api_type_has_tag (info->api, META_TAG_AUDIO)))
return TRUE;
return FALSE;
}
static GstFlowReturn
gst_audio_resample_submit_input_buffer (GstBaseTransform * base,
gboolean is_discont, GstBuffer * input)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
if (base->segment.format == GST_FORMAT_TIME) {
input =
gst_audio_buffer_clip (input, &base->segment, resample->in.rate,
resample->in.bpf);
if (!input)
return GST_FLOW_OK;
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->submit_input_buffer (base,
is_discont, input);
}
static gboolean
gst_audio_resample_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (parent);
GstBaseTransform *trans;
gboolean res = TRUE;
trans = GST_BASE_TRANSFORM (resample);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
guint64 latency;
gint rate = resample->in.rate;
gint resampler_latency;
if (resample->converter)
resampler_latency =
gst_audio_converter_get_max_latency (resample->converter);
else
resampler_latency = 0;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
if ((res =
gst_pad_peer_query (GST_BASE_TRANSFORM_SINK_PAD (trans),
query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG_OBJECT (resample, "Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
if (rate != 0 && resampler_latency != 0)
latency = gst_util_uint64_scale_round (resampler_latency,
GST_SECOND, rate);
else
latency = 0;
GST_DEBUG_OBJECT (resample, "Our latency: %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency));
min += latency;
if (GST_CLOCK_TIME_IS_VALID (max))
max += latency;
GST_DEBUG_OBJECT (resample, "Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
static void
gst_audio_resample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioResample *resample;
resample = GST_AUDIO_RESAMPLE (object);
switch (prop_id) {
case PROP_QUALITY:
/* FIXME locking! */
resample->quality = g_value_get_int (value);
GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
gst_audio_resample_update_state (resample, NULL, NULL);
break;
case PROP_RESAMPLE_METHOD:
resample->method = g_value_get_enum (value);
gst_audio_resample_update_state (resample, NULL, NULL);
break;
case PROP_SINC_FILTER_MODE:
/* FIXME locking! */
resample->sinc_filter_mode = g_value_get_enum (value);
gst_audio_resample_update_state (resample, NULL, NULL);
break;
case PROP_SINC_FILTER_AUTO_THRESHOLD:
/* FIXME locking! */
resample->sinc_filter_auto_threshold = g_value_get_uint (value);
gst_audio_resample_update_state (resample, NULL, NULL);
break;
case PROP_SINC_FILTER_INTERPOLATION:
/* FIXME locking! */
resample->sinc_filter_interpolation = g_value_get_enum (value);
gst_audio_resample_update_state (resample, NULL, NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_resample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioResample *resample;
resample = GST_AUDIO_RESAMPLE (object);
switch (prop_id) {
case PROP_QUALITY:
g_value_set_int (value, resample->quality);
break;
case PROP_RESAMPLE_METHOD:
g_value_set_enum (value, resample->method);
break;
case PROP_SINC_FILTER_MODE:
g_value_set_enum (value, resample->sinc_filter_mode);
break;
case PROP_SINC_FILTER_AUTO_THRESHOLD:
g_value_set_uint (value, resample->sinc_filter_auto_threshold);
break;
case PROP_SINC_FILTER_INTERPOLATION:
g_value_set_enum (value, resample->sinc_filter_interpolation);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return GST_ELEMENT_REGISTER (audioresample, plugin);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audioresample,
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN);
Reference in a new issue View git blame Copy permalink