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gstreamer/gst/audioresample/gstaudioresample.c
Sebastian Dröge 6d423cbba2 audioresample: Drain the resampler and reset timestamp tracking on caps changes 
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 as the resampler gets reset if anything
but the sample rate changes.

This is usually not a problem as the first buffer after a caps event
usually has the discont flag set, but can cause problems if
 - the caps event is followed by a segment event, which then causes
   draining according to the new sample rate
 - the caps were changed because of rengotiation due to a reconfigure
   event and there is not discontinuity from upstream

In both cases we would output buffers with completely wrong timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/670>
2020-05-27 17:06:08 +00:00

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/* 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));
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);
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;
}
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 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;
gpointer out[1];
out_len =
gst_audio_converter_get_out_frames (resample->converter, history_len);
if (out_len == 0)
return;
outsize = out_len * resample->out.bpf;
out[0] = g_malloc (outsize);
gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
out, out_len);
g_free (out[0]);
}
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_SEGMENT:
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;
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);
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))) {
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,
g_quark_from_string (GST_META_TAG_AUDIO_STR))))
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)
{
GST_DEBUG_CATEGORY_INIT (audio_resample_debug, "audioresample", 0,
"audio resampling element");
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
GST_TYPE_AUDIO_RESAMPLE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audioresample,
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN);
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