diff options
Diffstat (limited to 'gst/audiorate/gstaudiorate.c')
| -rw-r--r-- | gst/audiorate/gstaudiorate.c | 816 |
1 files changed, 0 insertions, 816 deletions
diff --git a/gst/audiorate/gstaudiorate.c b/gst/audiorate/gstaudiorate.c deleted file mode 100644 index 0adf8e18..00000000 --- a/gst/audiorate/gstaudiorate.c +++ /dev/null @@ -1,816 +0,0 @@ -/* 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); - -/* elementfactory information */ -static const GstElementDetails audio_rate_details = -GST_ELEMENT_DETAILS ("Audio rate adjuster", - "Filter/Effect/Audio", - "Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream", - "Wim Taymans <wim@fluendo.com>"); - -/* GstAudioRate signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - -#define DEFAULT_SILENT TRUE -#define DEFAULT_TOLERANCE 0 - -enum -{ - ARG_0, - ARG_IN, - ARG_OUT, - ARG_ADD, - ARG_DROP, - ARG_SILENT, - ARG_TOLERANCE, - /* FILL ME */ -}; - -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 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 (element_class, &audio_rate_details); - - 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)); - g_object_class_install_property (object_class, ARG_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_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_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)); - - 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: - /* FIXME, fill last segment */ - 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 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); - } - - 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, - 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); - - 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)); - 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) - g_object_notify (G_OBJECT (audiorate), "add"); - } - - } 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); - - if (!audiorate->silent) - g_object_notify (G_OBJECT (audiorate), "drop"); - - 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) - g_object_notify (G_OBJECT (audiorate), "drop"); - } - } - -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); - audiorate->out++; - - audiorate->next_offset = in_offset_end; -beach: - - gst_object_unref (audiorate); - - return ret; - - /* ERRORS */ -not_negotiated: - { - 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; - 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; - 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) |