--- /dev/null
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-audiorate
+ * @see_also: #GstVideoRate
+ *
+ * This element takes an incoming stream of timestamped raw audio frames and
+ * produces a perfect stream by inserting or dropping samples as needed.
+ *
+ * This operation may be of use to link to elements that require or otherwise
+ * implicitly assume a perfect stream as they do not store timestamps,
+ * but derive this by some means (e.g. bitrate for some AVI cases).
+ *
+ * The properties #GstAudioRate:in, #GstAudioRate:out, #GstAudioRate:add
+ * and #GstAudioRate:drop can be read to obtain information about number of
+ * input samples, output samples, dropped samples (i.e. the number of unused
+ * input samples) and inserted samples (i.e. the number of samples added to
+ * stream).
+ *
+ * When the #GstAudioRate:silent property is set to FALSE, a GObject property
+ * notification will be emitted whenever one of the #GstAudioRate:add or
+ * #GstAudioRate:drop values changes.
+ * This can potentially cause performance degradation.
+ * Note that property notification will happen from the streaming thread, so
+ * applications should be prepared for this.
+ *
+ * If the #GstAudioRate:tolerance property is non-zero, and an incoming buffer's
+ * timestamp deviates less than the property indicates from what would make a
+ * 'perfect time', then no samples will be added or dropped.
+ * Note that the output is still guaranteed to be a perfect stream, which means
+ * that the incoming data is then simply shifted (by less than the indicated
+ * tolerance) to a perfect time.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v alsasrc ! audiorate ! wavenc ! filesink location=alsa.wav
+ * ]| Capture audio from an ALSA device, and turn it into a perfect stream
+ * for saving in a raw audio file.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include <stdlib.h>
+
+#include "gstaudiorate.h"
+
+#define GST_CAT_DEFAULT audio_rate_debug
+GST_DEBUG_CATEGORY_STATIC (audio_rate_debug);
+
+/* GstAudioRate signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+#define DEFAULT_SILENT TRUE
+#define DEFAULT_TOLERANCE 0
+#define DEFAULT_SKIP_TO_FIRST FALSE
+
+enum
+{
+ ARG_0,
+ ARG_IN,
+ ARG_OUT,
+ ARG_ADD,
+ ARG_DROP,
+ ARG_SILENT,
+ ARG_TOLERANCE,
+ ARG_SKIP_TO_FIRST
+};
+
+static GstStaticPadTemplate gst_audio_rate_src_template =
+ GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS ";"
+ GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
+ );
+
+static GstStaticPadTemplate gst_audio_rate_sink_template =
+ GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS ";"
+ GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
+ );
+
+static void gst_audio_rate_base_init (gpointer g_class);
+static void gst_audio_rate_class_init (GstAudioRateClass * klass);
+static void gst_audio_rate_init (GstAudioRate * audiorate);
+static gboolean gst_audio_rate_sink_event (GstPad * pad, GstEvent * event);
+static gboolean gst_audio_rate_src_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstBuffer * buf);
+
+static void gst_audio_rate_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_rate_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element,
+ GstStateChange transition);
+
+static GstElementClass *parent_class = NULL;
+
+/*static guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */
+
+static GParamSpec *pspec_drop = NULL;
+static GParamSpec *pspec_add = NULL;
+
+static GType
+gst_audio_rate_get_type (void)
+{
+ static GType audio_rate_type = 0;
+
+ if (!audio_rate_type) {
+ static const GTypeInfo audio_rate_info = {
+ sizeof (GstAudioRateClass),
+ gst_audio_rate_base_init,
+ NULL,
+ (GClassInitFunc) gst_audio_rate_class_init,
+ NULL,
+ NULL,
+ sizeof (GstAudioRate),
+ 0,
+ (GInstanceInitFunc) gst_audio_rate_init,
+ };
+
+ audio_rate_type = g_type_register_static (GST_TYPE_ELEMENT,
+ "GstAudioRate", &audio_rate_info, 0);
+ }
+
+ return audio_rate_type;
+}
+
+static void
+gst_audio_rate_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+ gst_element_class_set_details_simple (element_class,
+ "Audio rate adjuster", "Filter/Effect/Audio",
+ "Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream",
+ "Wim Taymans <wim@fluendo.com>");
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_audio_rate_sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_audio_rate_src_template));
+}
+
+static void
+gst_audio_rate_class_init (GstAudioRateClass * klass)
+{
+ GObjectClass *object_class = G_OBJECT_CLASS (klass);
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ object_class->set_property = gst_audio_rate_set_property;
+ object_class->get_property = gst_audio_rate_get_property;
+
+ g_object_class_install_property (object_class, ARG_IN,
+ g_param_spec_uint64 ("in", "In",
+ "Number of input samples", 0, G_MAXUINT64, 0,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (object_class, ARG_OUT,
+ g_param_spec_uint64 ("out", "Out", "Number of output samples", 0,
+ G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+ pspec_add = g_param_spec_uint64 ("add", "Add", "Number of added samples",
+ 0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS);
+ g_object_class_install_property (object_class, ARG_ADD, pspec_add);
+ pspec_drop = g_param_spec_uint64 ("drop", "Drop", "Number of dropped samples",
+ 0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS);
+ g_object_class_install_property (object_class, ARG_DROP, pspec_drop);
+ g_object_class_install_property (object_class, ARG_SILENT,
+ g_param_spec_boolean ("silent", "silent",
+ "Don't emit notify for dropped and duplicated frames", DEFAULT_SILENT,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstAudioRate:tolerance
+ *
+ * The difference between incoming timestamp and next timestamp must exceed
+ * the given value for audiorate to add or drop samples.
+ *
+ * Since: 0.10.26
+ **/
+ g_object_class_install_property (object_class, ARG_TOLERANCE,
+ g_param_spec_uint64 ("tolerance", "tolerance",
+ "Only act if timestamp jitter/imperfection exceeds indicated tolerance (ns)",
+ 0, G_MAXUINT64, DEFAULT_TOLERANCE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstAudioRate:skip-to-first:
+ *
+ * Don't produce buffers before the first one we receive.
+ *
+ * Since: 0.10.33
+ **/
+ g_object_class_install_property (object_class, ARG_SKIP_TO_FIRST,
+ g_param_spec_boolean ("skip-to-first", "Skip to first buffer",
+ "Don't produce buffers before the first one we receive",
+ DEFAULT_SKIP_TO_FIRST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ element_class->change_state = gst_audio_rate_change_state;
+}
+
+static void
+gst_audio_rate_reset (GstAudioRate * audiorate)
+{
+ audiorate->next_offset = -1;
+ audiorate->next_ts = -1;
+ audiorate->discont = TRUE;
+ gst_segment_init (&audiorate->sink_segment, GST_FORMAT_UNDEFINED);
+ gst_segment_init (&audiorate->src_segment, GST_FORMAT_TIME);
+
+ GST_DEBUG_OBJECT (audiorate, "handle reset");
+}
+
+static gboolean
+gst_audio_rate_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstAudioRate *audiorate;
+ GstStructure *structure;
+ GstPad *otherpad;
+ gboolean ret = FALSE;
+ gint channels, width, rate;
+
+ audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ if (!gst_structure_get_int (structure, "channels", &channels))
+ goto wrong_caps;
+ if (!gst_structure_get_int (structure, "width", &width))
+ goto wrong_caps;
+ if (!gst_structure_get_int (structure, "rate", &rate))
+ goto wrong_caps;
+
+ audiorate->bytes_per_sample = channels * (width / 8);
+ if (audiorate->bytes_per_sample == 0)
+ goto wrong_format;
+
+ audiorate->rate = rate;
+
+ /* the format is correct, configure caps on other pad */
+ otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
+ audiorate->srcpad;
+
+ ret = gst_pad_set_caps (otherpad, caps);
+
+done:
+ gst_object_unref (audiorate);
+ return ret;
+
+ /* ERRORS */
+wrong_caps:
+ {
+ GST_DEBUG_OBJECT (audiorate, "could not get channels/width from caps");
+ goto done;
+ }
+wrong_format:
+ {
+ GST_DEBUG_OBJECT (audiorate, "bytes_per_samples gave 0");
+ goto done;
+ }
+}
+
+static void
+gst_audio_rate_init (GstAudioRate * audiorate)
+{
+ audiorate->sinkpad =
+ gst_pad_new_from_static_template (&gst_audio_rate_sink_template, "sink");
+ gst_pad_set_event_function (audiorate->sinkpad, gst_audio_rate_sink_event);
+ gst_pad_set_chain_function (audiorate->sinkpad, gst_audio_rate_chain);
+ gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audio_rate_setcaps);
+ gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);
+ gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
+
+ audiorate->srcpad =
+ gst_pad_new_from_static_template (&gst_audio_rate_src_template, "src");
+ gst_pad_set_event_function (audiorate->srcpad, gst_audio_rate_src_event);
+ gst_pad_set_setcaps_function (audiorate->srcpad, gst_audio_rate_setcaps);
+ gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);
+ gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
+
+ audiorate->in = 0;
+ audiorate->out = 0;
+ audiorate->drop = 0;
+ audiorate->add = 0;
+ audiorate->silent = DEFAULT_SILENT;
+ audiorate->tolerance = DEFAULT_TOLERANCE;
+}
+
+static void
+gst_audio_rate_fill_to_time (GstAudioRate * audiorate, GstClockTime time)
+{
+ GstBuffer *buf;
+
+ GST_DEBUG_OBJECT (audiorate, "next_ts: %" GST_TIME_FORMAT
+ ", filling to %" GST_TIME_FORMAT, GST_TIME_ARGS (audiorate->next_ts),
+ GST_TIME_ARGS (time));
+
+ if (!GST_CLOCK_TIME_IS_VALID (time) ||
+ !GST_CLOCK_TIME_IS_VALID (audiorate->next_ts))
+ return;
+
+ /* feed an empty buffer to chain with the given timestamp,
+ * it will take care of filling */
+ buf = gst_buffer_new ();
+ GST_BUFFER_TIMESTAMP (buf) = time;
+ gst_audio_rate_chain (audiorate->sinkpad, buf);
+}
+
+static gboolean
+gst_audio_rate_sink_event (GstPad * pad, GstEvent * event)
+{
+ gboolean res;
+ GstAudioRate *audiorate;
+
+ audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_FLUSH_STOP:
+ GST_DEBUG_OBJECT (audiorate, "handling FLUSH_STOP");
+ gst_audio_rate_reset (audiorate);
+ res = gst_pad_push_event (audiorate->srcpad, event);
+ break;
+ case GST_EVENT_NEWSEGMENT:
+ {
+ GstFormat format;
+ gdouble rate, arate;
+ gint64 start, stop, time;
+ gboolean update;
+
+ gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+ &start, &stop, &time);
+
+ GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT");
+ /* FIXME: bad things will likely happen if rate < 0 ... */
+ if (!update) {
+ /* a new segment starts. We need to figure out what will be the next
+ * sample offset. We mark the offsets as invalid so that the _chain
+ * function will perform this calculation. */
+ gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop);
+ audiorate->next_offset = -1;
+ audiorate->next_ts = -1;
+ } else {
+ gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.start);
+ }
+
+ /* we accept all formats */
+ gst_segment_set_newsegment_full (&audiorate->sink_segment, update, rate,
+ arate, format, start, stop, time);
+
+ GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT,
+ &audiorate->sink_segment);
+
+ if (format == GST_FORMAT_TIME) {
+ /* TIME formats can be copied to src and forwarded */
+ res = gst_pad_push_event (audiorate->srcpad, event);
+ memcpy (&audiorate->src_segment, &audiorate->sink_segment,
+ sizeof (GstSegment));
+ } else {
+ /* other formats will be handled in the _chain function */
+ gst_event_unref (event);
+ res = TRUE;
+ }
+ break;
+ }
+ case GST_EVENT_EOS:
+ /* Fill segment until the end */
+ if (GST_CLOCK_TIME_IS_VALID (audiorate->src_segment.stop))
+ gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop);
+ res = gst_pad_push_event (audiorate->srcpad, event);
+ break;
+ default:
+ res = gst_pad_push_event (audiorate->srcpad, event);
+ break;
+ }
+
+ gst_object_unref (audiorate);
+
+ return res;
+}
+
+static gboolean
+gst_audio_rate_src_event (GstPad * pad, GstEvent * event)
+{
+ gboolean res;
+ GstAudioRate *audiorate;
+
+ audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
+
+ switch (GST_EVENT_TYPE (event)) {
+ default:
+ res = gst_pad_push_event (audiorate->sinkpad, event);
+ break;
+ }
+
+ gst_object_unref (audiorate);
+
+ return res;
+}
+
+static gboolean
+gst_audio_rate_convert (GstAudioRate * audiorate,
+ GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
+{
+ if (src_fmt == dest_fmt) {
+ *dest_val = src_val;
+ return TRUE;
+ }
+
+ switch (src_fmt) {
+ case GST_FORMAT_DEFAULT:
+ switch (dest_fmt) {
+ case GST_FORMAT_BYTES:
+ *dest_val = src_val * audiorate->bytes_per_sample;
+ break;
+ case GST_FORMAT_TIME:
+ *dest_val =
+ gst_util_uint64_scale_int (src_val, GST_SECOND, audiorate->rate);
+ break;
+ default:
+ return FALSE;;
+ }
+ break;
+ case GST_FORMAT_BYTES:
+ switch (dest_fmt) {
+ case GST_FORMAT_DEFAULT:
+ *dest_val = src_val / audiorate->bytes_per_sample;
+ break;
+ case GST_FORMAT_TIME:
+ *dest_val = gst_util_uint64_scale_int (src_val, GST_SECOND,
+ audiorate->rate * audiorate->bytes_per_sample);
+ break;
+ default:
+ return FALSE;;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (dest_fmt) {
+ case GST_FORMAT_BYTES:
+ *dest_val = gst_util_uint64_scale_int (src_val,
+ audiorate->rate * audiorate->bytes_per_sample, GST_SECOND);
+ break;
+ case GST_FORMAT_DEFAULT:
+ *dest_val =
+ gst_util_uint64_scale_int (src_val, audiorate->rate, GST_SECOND);
+ break;
+ default:
+ return FALSE;;
+ }
+ break;
+ default:
+ return FALSE;
+ }
+ return TRUE;
+}
+
+
+static gboolean
+gst_audio_rate_convert_segments (GstAudioRate * audiorate)
+{
+ GstFormat src_fmt, dst_fmt;
+
+ src_fmt = audiorate->sink_segment.format;
+ dst_fmt = audiorate->src_segment.format;
+
+#define CONVERT_VAL(field) gst_audio_rate_convert (audiorate, \
+ src_fmt, audiorate->sink_segment.field, \
+ dst_fmt, &audiorate->src_segment.field);
+
+ audiorate->sink_segment.rate = audiorate->src_segment.rate;
+ audiorate->sink_segment.abs_rate = audiorate->src_segment.abs_rate;
+ audiorate->sink_segment.flags = audiorate->src_segment.flags;
+ audiorate->sink_segment.applied_rate = audiorate->src_segment.applied_rate;
+ CONVERT_VAL (start);
+ CONVERT_VAL (stop);
+ CONVERT_VAL (time);
+ CONVERT_VAL (accum);
+ CONVERT_VAL (last_stop);
+#undef CONVERT_VAL
+
+ return TRUE;
+}
+
+static void
+gst_audio_rate_notify_drop (GstAudioRate * audiorate)
+{
+#if !GLIB_CHECK_VERSION(2,26,0)
+ g_object_notify ((GObject *) audiorate, "drop");
+#else
+ g_object_notify_by_pspec ((GObject *) audiorate, pspec_drop);
+#endif
+}
+
+static void
+gst_audio_rate_notify_add (GstAudioRate * audiorate)
+{
+#if !GLIB_CHECK_VERSION(2,26,0)
+ g_object_notify ((GObject *) audiorate, "add");
+#else
+ g_object_notify_by_pspec ((GObject *) audiorate, pspec_add);
+#endif
+}
+
+static GstFlowReturn
+gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstAudioRate *audiorate;
+ GstClockTime in_time;
+ guint64 in_offset, in_offset_end, in_samples;
+ guint in_size;
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstClockTimeDiff diff;
+
+ audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
+
+ /* need to be negotiated now */
+ if (audiorate->bytes_per_sample == 0)
+ goto not_negotiated;
+
+ /* we have a new pending segment */
+ if (audiorate->next_offset == -1) {
+ gint64 pos;
+
+ /* update the TIME segment */
+ gst_audio_rate_convert_segments (audiorate);
+
+ /* first buffer, we are negotiated and we have a segment, calculate the
+ * current expected offsets based on the segment.start, which is the first
+ * media time of the segment and should match the media time of the first
+ * buffer in that segment, which is the offset expressed in DEFAULT units.
+ */
+ /* convert first timestamp of segment to sample position */
+ pos = gst_util_uint64_scale_int (audiorate->src_segment.start,
+ audiorate->rate, GST_SECOND);
+
+ GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);
+
+ /* resyncing is a discont */
+ audiorate->discont = TRUE;
+
+ audiorate->next_offset = pos;
+ audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,
+ GST_SECOND, audiorate->rate);
+
+ if (audiorate->skip_to_first && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+ GST_DEBUG_OBJECT (audiorate, "but skipping to first buffer instead");
+ pos = gst_util_uint64_scale_int (GST_BUFFER_TIMESTAMP (buf),
+ audiorate->rate, GST_SECOND);
+ GST_DEBUG_OBJECT (audiorate, "so resync to offset %" G_GINT64_FORMAT,
+ pos);
+ audiorate->next_offset = pos;
+ audiorate->next_ts = GST_BUFFER_TIMESTAMP (buf);
+ }
+ }
+
+ audiorate->in++;
+
+ in_time = GST_BUFFER_TIMESTAMP (buf);
+ if (in_time == GST_CLOCK_TIME_NONE) {
+ GST_DEBUG_OBJECT (audiorate, "no timestamp, using expected next time");
+ in_time = audiorate->next_ts;
+ }
+
+ in_size = GST_BUFFER_SIZE (buf);
+ in_samples = in_size / audiorate->bytes_per_sample;
+
+ /* calculate the buffer offset */
+ in_offset = gst_util_uint64_scale_int_round (in_time, audiorate->rate,
+ GST_SECOND);
+ in_offset_end = in_offset + in_samples;
+
+ GST_LOG_OBJECT (audiorate,
+ "in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT
+ ", in_size:%u, in_offset:%" G_GUINT64_FORMAT ", in_offset_end:%"
+ G_GUINT64_FORMAT ", ->next_offset:%" G_GUINT64_FORMAT ", ->next_ts:%"
+ GST_TIME_FORMAT, GST_TIME_ARGS (in_time),
+ GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (in_samples, audiorate->rate)),
+ in_size, in_offset, in_offset_end, audiorate->next_offset,
+ GST_TIME_ARGS (audiorate->next_ts));
+
+ diff = in_time - audiorate->next_ts;
+ if (diff <= (GstClockTimeDiff) audiorate->tolerance &&
+ diff >= (GstClockTimeDiff) - audiorate->tolerance) {
+ /* buffer time close enough to expected time,
+ * so produce a perfect stream by simply 'shifting'
+ * it to next ts and offset and sending */
+ GST_LOG_OBJECT (audiorate, "within tolerance %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (audiorate->tolerance));
+ /* The outgoing buffer's offset will be set to ->next_offset, we also
+ * need to adjust the offset_end value accordingly */
+ in_offset_end = audiorate->next_offset + in_samples;
+ goto send;
+ }
+
+ /* do we need to insert samples */
+ if (in_offset > audiorate->next_offset) {
+ GstBuffer *fill;
+ gint fillsize;
+ guint64 fillsamples;
+
+ /* We don't want to allocate a single unreasonably huge buffer - it might
+ be hundreds of megabytes. So, limit each output buffer to one second of
+ audio */
+ fillsamples = in_offset - audiorate->next_offset;
+
+ while (fillsamples > 0) {
+ guint64 cursamples = MIN (fillsamples, audiorate->rate);
+
+ fillsamples -= cursamples;
+ fillsize = cursamples * audiorate->bytes_per_sample;
+
+ fill = gst_buffer_new_and_alloc (fillsize);
+ /* FIXME, 0 might not be the silence byte for the negotiated format. */
+ memset (GST_BUFFER_DATA (fill), 0, fillsize);
+
+ GST_DEBUG_OBJECT (audiorate, "inserting %" G_GUINT64_FORMAT " samples",
+ cursamples);
+
+ GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
+ audiorate->next_offset += cursamples;
+ GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset;
+
+ /* Use next timestamp, then calculate following timestamp based on
+ * offset to get duration. Neccesary complexity to get 'perfect'
+ * streams */
+ GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
+ audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,
+ GST_SECOND, audiorate->rate);
+ GST_BUFFER_DURATION (fill) = audiorate->next_ts -
+ GST_BUFFER_TIMESTAMP (fill);
+
+ /* we created this buffer to fill a gap */
+ GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);
+ /* set discont if it's pending, this is mostly done for the first buffer
+ * and after a flushing seek */
+ if (audiorate->discont) {
+ GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT);
+ audiorate->discont = FALSE;
+ }
+ gst_buffer_set_caps (fill, GST_PAD_CAPS (audiorate->srcpad));
+
+ ret = gst_pad_push (audiorate->srcpad, fill);
+ if (ret != GST_FLOW_OK)
+ goto beach;
+ audiorate->out++;
+ audiorate->add += cursamples;
+
+ if (!audiorate->silent)
+ gst_audio_rate_notify_add (audiorate);
+ }
+
+ } else if (in_offset < audiorate->next_offset) {
+ /* need to remove samples */
+ if (in_offset_end <= audiorate->next_offset) {
+ guint64 drop = in_size / audiorate->bytes_per_sample;
+
+ audiorate->drop += drop;
+
+ GST_DEBUG_OBJECT (audiorate, "dropping %" G_GUINT64_FORMAT " samples",
+ drop);
+
+ /* we can drop the buffer completely */
+ gst_buffer_unref (buf);
+ buf = NULL;
+
+ if (!audiorate->silent)
+ gst_audio_rate_notify_drop (audiorate);
+
+ goto beach;
+ } else {
+ guint64 truncsamples;
+ guint truncsize, leftsize;
+ GstBuffer *trunc;
+
+ /* truncate buffer */
+ truncsamples = audiorate->next_offset - in_offset;
+ truncsize = truncsamples * audiorate->bytes_per_sample;
+ leftsize = in_size - truncsize;
+
+ trunc = gst_buffer_create_sub (buf, truncsize, leftsize);
+
+ gst_buffer_unref (buf);
+ buf = trunc;
+
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (audiorate->srcpad));
+
+ audiorate->drop += truncsamples;
+ GST_DEBUG_OBJECT (audiorate, "truncating %" G_GUINT64_FORMAT " samples",
+ truncsamples);
+
+ if (!audiorate->silent)
+ gst_audio_rate_notify_drop (audiorate);
+ }
+ }
+
+send:
+ if (GST_BUFFER_SIZE (buf) == 0)
+ goto beach;
+
+ /* Now calculate parameters for whichever buffer (either the original
+ * or truncated one) we're pushing. */
+ GST_BUFFER_OFFSET (buf) = audiorate->next_offset;
+ GST_BUFFER_OFFSET_END (buf) = in_offset_end;
+
+ GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts;
+ audiorate->next_ts = gst_util_uint64_scale_int (in_offset_end,
+ GST_SECOND, audiorate->rate);
+ GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf);
+
+ if (audiorate->discont) {
+ /* we need to output a discont buffer, do so now */
+ GST_DEBUG_OBJECT (audiorate, "marking DISCONT on output buffer");
+ buf = gst_buffer_make_metadata_writable (buf);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
+ audiorate->discont = FALSE;
+ } else if (GST_BUFFER_IS_DISCONT (buf)) {
+ /* else we make everything continuous so we can safely remove the DISCONT
+ * flag from the buffer if there was one */
+ GST_DEBUG_OBJECT (audiorate, "removing DISCONT from buffer");
+ buf = gst_buffer_make_metadata_writable (buf);
+ GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
+ }
+
+ /* set last_stop on segment */
+ gst_segment_set_last_stop (&audiorate->src_segment, GST_FORMAT_TIME,
+ GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf));
+
+ ret = gst_pad_push (audiorate->srcpad, buf);
+ buf = NULL;
+ audiorate->out++;
+
+ audiorate->next_offset = in_offset_end;
+beach:
+
+ if (buf)
+ gst_buffer_unref (buf);
+
+ gst_object_unref (audiorate);
+
+ return ret;
+
+ /* ERRORS */
+not_negotiated:
+ {
+ gst_buffer_unref (buf);
+
+ GST_ELEMENT_ERROR (audiorate, STREAM, FORMAT,
+ (NULL), ("pipeline error, format was not negotiated"));
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+}
+
+static void
+gst_audio_rate_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+ GstAudioRate *audiorate = GST_AUDIO_RATE (object);
+
+ switch (prop_id) {
+ case ARG_SILENT:
+ audiorate->silent = g_value_get_boolean (value);
+ break;
+ case ARG_TOLERANCE:
+ audiorate->tolerance = g_value_get_uint64 (value);
+ break;
+ case ARG_SKIP_TO_FIRST:
+ audiorate->skip_to_first = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_rate_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec)
+{
+ GstAudioRate *audiorate = GST_AUDIO_RATE (object);
+
+ switch (prop_id) {
+ case ARG_IN:
+ g_value_set_uint64 (value, audiorate->in);
+ break;
+ case ARG_OUT:
+ g_value_set_uint64 (value, audiorate->out);
+ break;
+ case ARG_ADD:
+ g_value_set_uint64 (value, audiorate->add);
+ break;
+ case ARG_DROP:
+ g_value_set_uint64 (value, audiorate->drop);
+ break;
+ case ARG_SILENT:
+ g_value_set_boolean (value, audiorate->silent);
+ break;
+ case ARG_TOLERANCE:
+ g_value_set_uint64 (value, audiorate->tolerance);
+ break;
+ case ARG_SKIP_TO_FIRST:
+ g_value_set_boolean (value, audiorate->skip_to_first);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstStateChangeReturn
+gst_audio_rate_change_state (GstElement * element, GstStateChange transition)
+{
+ GstAudioRate *audiorate = GST_AUDIO_RATE (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ audiorate->in = 0;
+ audiorate->out = 0;
+ audiorate->drop = 0;
+ audiorate->bytes_per_sample = 0;
+ audiorate->add = 0;
+ gst_audio_rate_reset (audiorate);
+ break;
+ default:
+ break;
+ }
+
+ if (parent_class->change_state)
+ return parent_class->change_state (element, transition);
+
+ return GST_STATE_CHANGE_SUCCESS;
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+ GST_DEBUG_CATEGORY_INIT (audio_rate_debug, "audiorate", 0,
+ "AudioRate stream fixer");
+
+ return gst_element_register (plugin, "audiorate", GST_RANK_NONE,
+ GST_TYPE_AUDIO_RATE);
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "audiorate",
+ "Adjusts audio frames",
+ plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
projects / mafwsubrenderer / blobdiff