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GST-Tensordecoder-ov_ep/subprojects/gst-plugins-bad/gst/audiobuffersplit/gstaudiobuffersplit.c
Guillaume Desmottes dd0896f05a audiobuffersplit: disable max-silence-time if set to 0 
According to the property documentation max-silence-time is supposed to be
disabled when set to 0 but it was not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5442>
2023-10-06 14:48:46 +02:00

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/*
* GStreamer
* Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.com>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstaudiobuffersplit.h"
#define GST_CAT_DEFAULT gst_audio_buffer_split_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
", layout = (string) interleaved")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
", layout = (string) interleaved")
);
enum
{
PROP_0,
PROP_OUTPUT_BUFFER_DURATION,
PROP_OUTPUT_BUFFER_SIZE,
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT,
PROP_STRICT_BUFFER_SIZE,
PROP_GAPLESS,
PROP_MAX_SILENCE_TIME,
LAST_PROP
};
#define DEFAULT_OUTPUT_BUFFER_DURATION_N (1)
#define DEFAULT_OUTPUT_BUFFER_DURATION_D (50)
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
#define DEFAULT_STRICT_BUFFER_SIZE (FALSE)
#define DEFAULT_GAPLESS (FALSE)
#define DEFAULT_MAX_SILENCE_TIME (0)
#define parent_class gst_audio_buffer_split_parent_class
G_DEFINE_TYPE_WITH_CODE (GstAudioBufferSplit, gst_audio_buffer_split,
GST_TYPE_ELEMENT, GST_DEBUG_CATEGORY_INIT (gst_audio_buffer_split_debug,
"audiobuffersplit", 0, "Audio buffer splitter"););
GST_ELEMENT_REGISTER_DEFINE (audiobuffersplit, "audiobuffersplit",
GST_RANK_NONE, GST_TYPE_AUDIO_BUFFER_SPLIT);
static GstFlowReturn gst_audio_buffer_split_sink_chain (GstPad * pad,
GstObject * parent, GstBuffer * buffer);
static gboolean gst_audio_buffer_split_sink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_audio_buffer_split_src_query (GstPad * pad,
GstObject * parent, GstQuery * query);
static void gst_audio_buffer_split_finalize (GObject * object);
static void gst_audio_buffer_split_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_audio_buffer_split_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_audio_buffer_split_change_state (GstElement *
element, GstStateChange transition);
static void
gst_audio_buffer_split_class_init (GstAudioBufferSplitClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audio_buffer_split_set_property;
gobject_class->get_property = gst_audio_buffer_split_get_property;
gobject_class->finalize = gst_audio_buffer_split_finalize;
g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION,
gst_param_spec_fraction ("output-buffer-duration",
"Output Buffer Duration", "Output block size in seconds", 1, G_MAXINT,
G_MAXINT, 1, DEFAULT_OUTPUT_BUFFER_DURATION_N,
DEFAULT_OUTPUT_BUFFER_DURATION_D,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
/**
* GstAudioBufferSplit:output-buffer-size
*
* Allow specifying a buffer size for splitting. Zero by default.
* Takes precedence over output-buffer-duration when set to a
* non zero value else will not be in effect.
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_SIZE,
g_param_spec_uint ("output-buffer-size", "Output buffer size",
"Output block size in bytes, takes precedence over "
"buffer duration when set to non zero", 0, G_MAXINT, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
"Timestamp alignment threshold in nanoseconds", 0,
G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
g_param_spec_uint64 ("discont-wait", "Discont Wait",
"Window of time in nanoseconds to wait before "
"creating a discontinuity", 0,
G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class, PROP_STRICT_BUFFER_SIZE,
g_param_spec_boolean ("strict-buffer-size", "Strict buffer size",
"Discard the last samples at EOS or discont if they are too "
"small to fill a buffer", DEFAULT_STRICT_BUFFER_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class, PROP_GAPLESS,
g_param_spec_boolean ("gapless", "Gapless",
"Insert silence/drop samples instead of creating a discontinuity",
DEFAULT_GAPLESS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class, PROP_MAX_SILENCE_TIME,
g_param_spec_uint64 ("max-silence-time",
"Maximum time of silence to insert",
"Do not insert silence in gapless mode if the gap exceeds this "
"period (in ns) (0 = disabled)",
0, G_MAXUINT64, DEFAULT_MAX_SILENCE_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
gst_element_class_set_static_metadata (gstelement_class,
"Audio Buffer Split", "Audio/Filter",
"Splits raw audio buffers into equal sized chunks",
"Sebastian Dröge <sebastian@centricular.com>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template));
gstelement_class->change_state = gst_audio_buffer_split_change_state;
}
static void
gst_audio_buffer_split_init (GstAudioBufferSplit * self)
{
self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_buffer_split_sink_chain));
gst_pad_set_event_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_buffer_split_sink_event));
GST_PAD_SET_PROXY_CAPS (self->sinkpad);
gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
self->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_set_query_function (self->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_buffer_split_src_query));
GST_PAD_SET_PROXY_CAPS (self->srcpad);
gst_pad_use_fixed_caps (self->srcpad);
gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
self->output_buffer_duration_n = DEFAULT_OUTPUT_BUFFER_DURATION_N;
self->output_buffer_duration_d = DEFAULT_OUTPUT_BUFFER_DURATION_D;
self->strict_buffer_size = DEFAULT_STRICT_BUFFER_SIZE;
self->gapless = DEFAULT_GAPLESS;
self->output_buffer_size = 0;
self->adapter = gst_adapter_new ();
self->stream_align =
gst_audio_stream_align_new (48000, DEFAULT_ALIGNMENT_THRESHOLD,
DEFAULT_DISCONT_WAIT);
}
static void
gst_audio_buffer_split_finalize (GObject * object)
{
GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);
if (self->adapter) {
gst_object_unref (self->adapter);
self->adapter = NULL;
}
if (self->stream_align) {
gst_audio_stream_align_free (self->stream_align);
self->stream_align = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_audio_buffer_split_update_samples_per_buffer (GstAudioBufferSplit * self)
{
gboolean ret = TRUE;
GST_OBJECT_LOCK (self);
/* For a later time */
if (!self->info.finfo
|| GST_AUDIO_INFO_FORMAT (&self->info) == GST_AUDIO_FORMAT_UNKNOWN) {
self->samples_per_buffer = 0;
goto out;
}
if (self->output_buffer_size) {
self->output_buffer_duration_n =
self->output_buffer_size / GST_AUDIO_INFO_BPF (&self->info);
self->output_buffer_duration_d = GST_AUDIO_INFO_RATE (&self->info);
}
self->samples_per_buffer =
(((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
self->output_buffer_duration_n) / self->output_buffer_duration_d;
if (self->samples_per_buffer == 0) {
ret = FALSE;
goto out;
}
self->error_per_buffer =
(((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
self->output_buffer_duration_n) % self->output_buffer_duration_d;
self->accumulated_error = 0;
GST_DEBUG_OBJECT (self, "Buffer duration: %u/%u",
self->output_buffer_duration_n, self->output_buffer_duration_d);
GST_DEBUG_OBJECT (self, "Samples per buffer: %u (error: %u/%u)",
self->samples_per_buffer, self->error_per_buffer,
self->output_buffer_duration_d);
out:
GST_OBJECT_UNLOCK (self);
return ret;
}
static void
gst_audio_buffer_split_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);
switch (property_id) {
case PROP_OUTPUT_BUFFER_DURATION:
self->output_buffer_duration_n = gst_value_get_fraction_numerator (value);
self->output_buffer_duration_d =
gst_value_get_fraction_denominator (value);
gst_audio_buffer_split_update_samples_per_buffer (self);
break;
case PROP_OUTPUT_BUFFER_SIZE:
self->output_buffer_size = g_value_get_uint (value);
gst_audio_buffer_split_update_samples_per_buffer (self);
break;
case PROP_ALIGNMENT_THRESHOLD:
GST_OBJECT_LOCK (self);
gst_audio_stream_align_set_alignment_threshold (self->stream_align,
g_value_get_uint64 (value));
GST_OBJECT_UNLOCK (self);
break;
case PROP_DISCONT_WAIT:
GST_OBJECT_LOCK (self);
gst_audio_stream_align_set_discont_wait (self->stream_align,
g_value_get_uint64 (value));
GST_OBJECT_UNLOCK (self);
break;
case PROP_STRICT_BUFFER_SIZE:
self->strict_buffer_size = g_value_get_boolean (value);
break;
case PROP_GAPLESS:
self->gapless = g_value_get_boolean (value);
break;
case PROP_MAX_SILENCE_TIME:
self->max_silence_time = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
gst_audio_buffer_split_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);
switch (property_id) {
case PROP_OUTPUT_BUFFER_DURATION:
gst_value_set_fraction (value, self->output_buffer_duration_n,
self->output_buffer_duration_d);
break;
case PROP_OUTPUT_BUFFER_SIZE:
g_value_set_uint (value, self->output_buffer_size);
break;
case PROP_ALIGNMENT_THRESHOLD:
GST_OBJECT_LOCK (self);
g_value_set_uint64 (value,
gst_audio_stream_align_get_alignment_threshold (self->stream_align));
GST_OBJECT_UNLOCK (self);
break;
case PROP_DISCONT_WAIT:
GST_OBJECT_LOCK (self);
g_value_set_uint64 (value,
gst_audio_stream_align_get_discont_wait (self->stream_align));
GST_OBJECT_UNLOCK (self);
break;
case PROP_STRICT_BUFFER_SIZE:
g_value_set_boolean (value, self->strict_buffer_size);
break;
case PROP_GAPLESS:
g_value_set_boolean (value, self->gapless);
break;
case PROP_MAX_SILENCE_TIME:
g_value_set_uint64 (value, self->max_silence_time);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_audio_buffer_split_change_state (GstElement * element,
GstStateChange transition)
{
GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (element);
GstStateChangeReturn state_ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_audio_info_init (&self->info);
gst_segment_init (&self->in_segment, GST_FORMAT_TIME);
gst_segment_init (&self->out_segment, GST_FORMAT_UNDEFINED);
self->segment_pending = FALSE;
GST_OBJECT_LOCK (self);
gst_audio_stream_align_mark_discont (self->stream_align);
GST_OBJECT_UNLOCK (self);
self->current_offset = -1;
self->accumulated_error = 0;
self->samples_per_buffer = 0;
break;
default:
break;
}
state_ret =
GST_ELEMENT_CLASS (gst_audio_buffer_split_parent_class)->change_state
(element, transition);
if (state_ret == GST_STATE_CHANGE_FAILURE)
return state_ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_adapter_clear (self->adapter);
GST_OBJECT_LOCK (self);
gst_audio_stream_align_mark_discont (self->stream_align);
GST_OBJECT_UNLOCK (self);
break;
default:
break;
}
return state_ret;
}
static GstFlowReturn
gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force,
gint rate, gint bpf, guint samples_per_buffer)
{
gint size, avail;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime resync_pts;
resync_pts = self->resync_pts;
size = samples_per_buffer * bpf;
/* If we accumulated enough error for one sample, include one
* more sample in this buffer. Accumulated error is updated below */
if (self->error_per_buffer + self->accumulated_error >=
self->output_buffer_duration_d)
size += bpf;
while ((avail = gst_adapter_available (self->adapter)) >= size || (force
&& avail > 0)) {
GstBuffer *buffer;
GstClockTime resync_time_diff;
size = MIN (size, avail);
buffer = gst_adapter_take_buffer (self->adapter, size);
buffer = gst_buffer_make_writable (buffer);
/* After a reset we have to set the discont flag */
if (self->current_offset == 0)
GST_BUFFER_FLAG_SET (buffer,
GST_BUFFER_FLAG_DISCONT | GST_BUFFER_FLAG_RESYNC);
else
GST_BUFFER_FLAG_UNSET (buffer,
GST_BUFFER_FLAG_DISCONT | GST_BUFFER_FLAG_RESYNC);
resync_time_diff =
gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
if (self->out_segment.rate < 0.0) {
if (resync_pts > resync_time_diff)
GST_BUFFER_PTS (buffer) = resync_pts - resync_time_diff;
else
GST_BUFFER_PTS (buffer) = 0;
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale (size / bpf, GST_SECOND, rate);
self->current_offset += size / bpf;
} else {
GST_BUFFER_PTS (buffer) = resync_pts + resync_time_diff;
self->current_offset += size / bpf;
resync_time_diff =
gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
GST_BUFFER_DURATION (buffer) =
resync_time_diff - (GST_BUFFER_PTS (buffer) - resync_pts);
}
GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
self->accumulated_error =
(self->accumulated_error +
self->error_per_buffer) % self->output_buffer_duration_d;
GST_LOG_OBJECT (self,
"Outputting buffer at running time %" GST_TIME_FORMAT
" with timestamp %" GST_TIME_FORMAT " with duration %" GST_TIME_FORMAT
" (%u samples)",
GST_TIME_ARGS (gst_segment_to_running_time (&self->out_segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (buffer))),
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)), size / bpf);
ret = gst_pad_push (self->srcpad, buffer);
if (ret != GST_FLOW_OK)
break;
/* Update the size based on the accumulated error we have now after
* taking out a buffer. Same code as above */
size = samples_per_buffer * bpf;
if (self->error_per_buffer + self->accumulated_error >=
self->output_buffer_duration_d)
size += bpf;
}
return ret;
}
static GstFlowReturn
gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
GstBuffer * buffer, GstAudioFormat format, gint rate, gint bpf,
guint samples_per_buffer)
{
gboolean discont;
GstFlowReturn ret = GST_FLOW_OK;
guint avail = gst_adapter_available (self->adapter);
guint avail_samples = avail / bpf;
guint64 new_offset;
GstClockTime input_rt, current_rt;
GstClockTime input_duration;
GstClockTime current_rt_end;
input_rt =
gst_segment_to_running_time (&self->in_segment, GST_FORMAT_TIME,
GST_BUFFER_PTS (buffer));
input_duration =
(gst_buffer_get_size (buffer) / bpf) / ABS (self->in_segment.rate);
GST_OBJECT_LOCK (self);
if (self->in_segment.rate < 0) {
discont = FALSE;
} else {
discont = GST_BUFFER_IS_DISCONT (buffer)
|| GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_RESYNC);
}
/* If the segment rate is changing this is a discontinuity */
discont = discont || (self->out_segment.format != GST_FORMAT_UNDEFINED
&& self->in_segment.rate != self->out_segment.rate);
/* If this is the very first buffer we also have a discontinuity */
discont = discont || self->current_offset == -1;
discont =
gst_audio_stream_align_process (self->stream_align,
discont, input_rt, input_duration, NULL, NULL, NULL);
GST_OBJECT_UNLOCK (self);
if (!discont)
return ret;
/* Reset */
self->drop_samples = 0;
if (self->in_segment.rate < 0.0) {
current_rt =
self->resync_rt - gst_util_uint64_scale (self->current_offset +
avail_samples, GST_SECOND, rate * ABS (self->in_segment.rate));
current_rt_end =
self->resync_rt - gst_util_uint64_scale (self->current_offset,
GST_SECOND, rate * ABS (self->in_segment.rate));
} else {
current_rt =
self->resync_rt + gst_util_uint64_scale (self->current_offset,
GST_SECOND, rate * self->in_segment.rate);
current_rt_end =
self->resync_rt + gst_util_uint64_scale (self->current_offset +
avail_samples, GST_SECOND, rate * self->in_segment.rate);
}
if (self->gapless && self->current_offset != -1) {
GST_DEBUG_OBJECT (self,
"Got discont in gapless mode: Current running time %" GST_TIME_FORMAT
", current end running time %" GST_TIME_FORMAT
", running time after discont %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_rt),
GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt));
new_offset =
gst_util_uint64_scale (input_rt - self->resync_rt,
rate * ABS (self->in_segment.rate), GST_SECOND);
if (input_rt < self->resync_rt) {
guint64 drop_samples;
new_offset =
gst_util_uint64_scale (self->resync_rt -
input_rt, rate * ABS (self->in_segment.rate), GST_SECOND);
drop_samples = self->current_offset + avail_samples + new_offset;
GST_DEBUG_OBJECT (self,
"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
GST_SECOND, rate)));
self->drop_samples = drop_samples;
discont = FALSE;
} else if (new_offset > self->current_offset + avail_samples) {
guint64 silence_samples =
new_offset - (self->current_offset + avail_samples);
const GstAudioFormatInfo *info = gst_audio_format_get_info (format);
GstClockTime silence_time =
gst_util_uint64_scale (silence_samples, GST_SECOND, rate);
if (silence_time > self->max_silence_time && self->max_silence_time != 0) {
GST_DEBUG_OBJECT (self,
"Not inserting %" G_GUINT64_FORMAT " samples of silence (%"
GST_TIME_FORMAT " exceeds maximum %" GST_TIME_FORMAT ")",
silence_samples, GST_TIME_ARGS (silence_time),
GST_TIME_ARGS (self->max_silence_time));
} else {
GST_DEBUG_OBJECT (self,
"Inserting %" G_GUINT64_FORMAT " samples of silence (%"
GST_TIME_FORMAT ")", silence_samples, GST_TIME_ARGS (silence_time));
/* Insert silence buffers to fill the gap in 1s chunks */
while (silence_samples > 0) {
guint n_samples = MIN (silence_samples, rate);
GstBuffer *silence;
GstMapInfo map;
silence = gst_buffer_new_and_alloc (n_samples * bpf);
GST_BUFFER_FLAG_SET (silence, GST_BUFFER_FLAG_GAP);
gst_buffer_map (silence, &map, GST_MAP_WRITE);
gst_audio_format_info_fill_silence (info, map.data, map.size);
gst_buffer_unmap (silence, &map);
gst_adapter_push (self->adapter, silence);
ret =
gst_audio_buffer_split_output (self, FALSE, rate, bpf,
samples_per_buffer);
if (ret != GST_FLOW_OK)
return ret;
silence_samples -= n_samples;
}
discont = FALSE;
}
} else if (new_offset < self->current_offset + avail_samples) {
guint64 drop_samples = self->current_offset + avail_samples - new_offset;
GST_DEBUG_OBJECT (self,
"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
GST_SECOND, rate)));
self->drop_samples = drop_samples;
discont = FALSE;
}
}
if (discont) {
/* We might end up in here also in gapless mode, if the above code decided
* that no silence is to be inserted, because e.g. the gap is too big */
GST_DEBUG_OBJECT (self,
"Got %s: Current running time %" GST_TIME_FORMAT
", current end running time %" GST_TIME_FORMAT
", running time after discont %" GST_TIME_FORMAT,
self->current_offset == -1 ? "first buffer" : "discont",
GST_TIME_ARGS (current_rt),
GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt));
if (self->strict_buffer_size) {
gst_adapter_clear (self->adapter);
ret = GST_FLOW_OK;
} else {
ret =
gst_audio_buffer_split_output (self, TRUE, rate, bpf,
samples_per_buffer);
}
self->current_offset = 0;
self->accumulated_error = 0;
self->resync_pts = GST_BUFFER_PTS (buffer);
self->resync_rt = input_rt;
if (self->segment_pending) {
GstEvent *event;
self->out_segment = self->in_segment;
GST_DEBUG_OBJECT (self, "Updating output segment %" GST_SEGMENT_FORMAT,
&self->out_segment);
event = gst_event_new_segment (&self->out_segment);
gst_event_set_seqnum (event, self->segment_seqnum);
gst_pad_push_event (self->srcpad, event);
self->segment_pending = FALSE;
}
}
return ret;
}
static GstBuffer *
gst_audio_buffer_split_clip_buffer_start_for_gapless (GstAudioBufferSplit *
self, GstBuffer * buffer, gint rate, gint bpf)
{
guint nsamples;
if (!self->gapless || self->drop_samples == 0)
return buffer;
nsamples = gst_buffer_get_size (buffer) / bpf;
GST_DEBUG_OBJECT (self, "Have to drop %" G_GUINT64_FORMAT
" samples, got %u samples", self->drop_samples, nsamples);
if (nsamples <= self->drop_samples) {
gst_buffer_unref (buffer);
self->drop_samples -= nsamples;
return NULL;
}
if (self->out_segment.rate < 0.0) {
buffer =
gst_audio_buffer_truncate (buffer, bpf, 0,
nsamples - self->drop_samples);
self->drop_samples = 0;
return buffer;
} else {
buffer = gst_audio_buffer_truncate (buffer, bpf, self->drop_samples, -1);
self->drop_samples = 0;
return buffer;
}
return buffer;
}
static GstFlowReturn
gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
GstFlowReturn ret;
GstAudioFormat format;
gint rate, bpf, samples_per_buffer;
GST_OBJECT_LOCK (self);
format =
self->info.
finfo ? GST_AUDIO_INFO_FORMAT (&self->info) : GST_AUDIO_FORMAT_UNKNOWN;
rate = GST_AUDIO_INFO_RATE (&self->info);
bpf = GST_AUDIO_INFO_BPF (&self->info);
samples_per_buffer = self->samples_per_buffer;
GST_OBJECT_UNLOCK (self);
GST_LOG_OBJECT (self,
"Processing buffer at running time %" GST_TIME_FORMAT
" with timestamp %" GST_TIME_FORMAT " with duration %" GST_TIME_FORMAT
" (%u samples)",
GST_TIME_ARGS (gst_segment_to_running_time (&self->in_segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (buffer))),
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
(guint) (gst_buffer_get_size (buffer) / bpf));
if (format == GST_AUDIO_FORMAT_UNKNOWN || samples_per_buffer == 0) {
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
buffer = gst_audio_buffer_clip (buffer, &self->in_segment, rate, bpf);
if (!buffer)
return GST_FLOW_OK;
ret =
gst_audio_buffer_split_handle_discont (self, buffer, format, rate, bpf,
samples_per_buffer);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buffer);
return ret;
}
buffer =
gst_audio_buffer_split_clip_buffer_start_for_gapless (self, buffer, rate,
bpf);
if (!buffer)
return GST_FLOW_OK;
gst_adapter_push (self->adapter, buffer);
return gst_audio_buffer_split_output (self, FALSE, rate, bpf,
samples_per_buffer);
}
static gboolean
gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
gboolean ret = FALSE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:{
GstCaps *caps;
GstAudioInfo info;
gst_event_parse_caps (event, &caps);
ret = gst_audio_info_from_caps (&info, caps);
if (ret) {
GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps);
if (!gst_audio_info_is_equal (&info, &self->info)) {
if (self->strict_buffer_size) {
gst_adapter_clear (self->adapter);
} else {
GstAudioFormat format;
gint rate, bpf, samples_per_buffer;
GST_OBJECT_LOCK (self);
format =
self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
GST_AUDIO_FORMAT_UNKNOWN;
rate = GST_AUDIO_INFO_RATE (&self->info);
bpf = GST_AUDIO_INFO_BPF (&self->info);
samples_per_buffer = self->samples_per_buffer;
GST_OBJECT_UNLOCK (self);
if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
gst_audio_buffer_split_output (self, TRUE, rate, bpf,
samples_per_buffer);
}
}
self->info = info;
GST_OBJECT_LOCK (self);
gst_audio_stream_align_set_rate (self->stream_align, self->info.rate);
GST_OBJECT_UNLOCK (self);
ret = gst_audio_buffer_split_update_samples_per_buffer (self);
} else {
ret = FALSE;
}
if (ret)
ret = gst_pad_event_default (pad, parent, event);
else
gst_event_unref (event);
break;
}
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&self->in_segment, GST_FORMAT_TIME);
gst_segment_init (&self->out_segment, GST_FORMAT_UNDEFINED);
self->segment_pending = FALSE;
GST_OBJECT_LOCK (self);
gst_audio_stream_align_mark_discont (self->stream_align);
GST_OBJECT_UNLOCK (self);
self->current_offset = -1;
self->accumulated_error = 0;
gst_adapter_clear (self->adapter);
ret = gst_pad_event_default (pad, parent, event);
break;
case GST_EVENT_SEGMENT:
gst_event_copy_segment (event, &self->in_segment);
if (self->in_segment.format != GST_FORMAT_TIME) {
gst_event_unref (event);
ret = FALSE;
} else {
GST_DEBUG_OBJECT (self,
"Received new input segment %" GST_SEGMENT_FORMAT,
&self->in_segment);
self->segment_pending = TRUE;
self->segment_seqnum = gst_event_get_seqnum (event);
gst_event_unref (event);
ret = TRUE;
}
break;
case GST_EVENT_EOS:
if (self->strict_buffer_size) {
gst_adapter_clear (self->adapter);
} else {
GstAudioFormat format;
gint rate, bpf, samples_per_buffer;
GST_OBJECT_LOCK (self);
format =
self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
GST_AUDIO_FORMAT_UNKNOWN;
rate = GST_AUDIO_INFO_RATE (&self->info);
bpf = GST_AUDIO_INFO_BPF (&self->info);
samples_per_buffer = self->samples_per_buffer;
GST_OBJECT_UNLOCK (self);
if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
gst_audio_buffer_split_output (self, TRUE, rate, bpf,
samples_per_buffer);
}
ret = gst_pad_event_default (pad, parent, event);
break;
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static gboolean
gst_audio_buffer_split_src_query (GstPad * pad,
GstObject * parent, GstQuery * query)
{
GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
gboolean ret = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
if ((ret = gst_pad_peer_query (self->sinkpad, query))) {
GstClockTime latency;
GstClockTime min, max;
gboolean live;
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG_OBJECT (self, "Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
latency =
gst_util_uint64_scale (GST_SECOND, self->output_buffer_duration_n,
self->output_buffer_duration_d);
GST_DEBUG_OBJECT (self, "Our latency: min %" GST_TIME_FORMAT
", max %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency), GST_TIME_ARGS (latency));
min += latency;
if (max != GST_CLOCK_TIME_NONE)
max += latency;
GST_DEBUG_OBJECT (self, "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:
ret = gst_pad_query_default (pad, parent, query);
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return GST_ELEMENT_REGISTER (audiobuffersplit, plugin);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audiobuffersplit,
"Audio buffer splitter",
plugin_init, VERSION, "LGPL", PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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