gstreamer/gst/audiorate/gstaudiorate.c

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/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
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/**
* SECTION:element-audiorate
* @title: audiorate
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* @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).
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*
* 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.
*
* ## Example pipelines
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* |[
* gst-launch-1.0 -v autoaudiosrc ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav
* ]|
* Capture audio from the sound card and turn it into a perfect stream
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* for saving in a raw audio file.
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.file ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav
* ]|
* Decodes an audio file and transforms it into a perfect stream for saving
* in a raw audio WAV file. Without the audio rate, the timing might not be
* preserved correctly in the WAV file in case the decoded stream is jittery
* or there are samples missing.
*
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*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
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#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 (40 * GST_MSECOND)
#define DEFAULT_SKIP_TO_FIRST FALSE
enum
{
PROP_0,
PROP_IN,
PROP_OUT,
PROP_ADD,
PROP_DROP,
PROP_SILENT,
PROP_TOLERANCE,
PROP_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_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
", layout = (string) { interleaved, non-interleaved }")
);
static GstStaticPadTemplate gst_audio_rate_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, non-interleaved }")
);
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static gboolean gst_audio_rate_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_audio_rate_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstObject * parent,
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 guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */
static GParamSpec *pspec_drop = NULL;
static GParamSpec *pspec_add = NULL;
#define gst_audio_rate_parent_class parent_class
G_DEFINE_TYPE (GstAudioRate, gst_audio_rate, GST_TYPE_ELEMENT);
static void
gst_audio_rate_class_init (GstAudioRateClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (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, PROP_IN,
g_param_spec_uint64 ("in", "In",
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
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"Number of input samples", 0, G_MAXUINT64, 0,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (object_class, PROP_OUT,
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
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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, PROP_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, PROP_DROP, pspec_drop);
g_object_class_install_property (object_class, PROP_SILENT,
g_param_spec_boolean ("silent", "silent",
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
"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.
*/
g_object_class_install_property (object_class, PROP_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.
*/
g_object_class_install_property (object_class, PROP_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));
gst_element_class_set_static_metadata (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_static_pad_template (element_class,
&gst_audio_rate_sink_template);
gst_element_class_add_static_pad_template (element_class,
&gst_audio_rate_src_template);
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 (GstAudioRate * audiorate, GstCaps * caps)
{
GstAudioInfo info;
gint prev_rate = 0;
if (!gst_audio_info_from_caps (&info, caps))
goto wrong_caps;
prev_rate = audiorate->info.rate;
audiorate->info = info;
if (audiorate->next_offset >= 0 && prev_rate > 0 && prev_rate != info.rate) {
GST_DEBUG_OBJECT (audiorate,
"rate changed from %d to %d", prev_rate, info.rate);
/* calculate next_offset based on new rate value */
audiorate->next_offset =
gst_util_uint64_scale_int_round (audiorate->next_ts,
info.rate, GST_SECOND);
}
return TRUE;
/* ERRORS */
wrong_caps:
{
GST_DEBUG_OBJECT (audiorate, "could not parse caps");
return FALSE;
}
}
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_PROXY_CAPS (audiorate->sinkpad);
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_PROXY_CAPS (audiorate->srcpad);
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;
2011-11-17 11:48:25 +00:00
gst_audio_rate_chain (audiorate->sinkpad, GST_OBJECT_CAST (audiorate), buf);
}
static gboolean
2011-11-17 11:48:25 +00:00
gst_audio_rate_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
gboolean res;
GstAudioRate *audiorate;
2011-11-17 11:48:25 +00:00
audiorate = GST_AUDIO_RATE (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
if ((res = gst_audio_rate_setcaps (audiorate, caps))) {
res = gst_pad_push_event (audiorate->srcpad, event);
} else {
gst_event_unref (event);
}
break;
}
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;
2011-05-16 11:48:11 +00:00
case GST_EVENT_SEGMENT:
{
gst_event_copy_segment (event, &audiorate->sink_segment);
GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT");
2011-05-16 11:48:11 +00:00
#if 0
/* 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);
2011-05-16 11:48:11 +00:00
#endif
audiorate->next_offset = -1;
audiorate->next_ts = -1;
2011-05-16 11:48:11 +00:00
#if 0
} else {
gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.start);
}
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#endif
GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT,
&audiorate->sink_segment);
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if (audiorate->sink_segment.format == GST_FORMAT_TIME) {
/* TIME formats can be copied to src and forwarded */
res = gst_pad_push_event (audiorate->srcpad, event);
2011-05-16 11:48:11 +00:00
gst_segment_copy_into (&audiorate->sink_segment,
&audiorate->src_segment);
} 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;
case GST_EVENT_GAP:
{
/* Fill until end of gap */
GstClockTime timestamp, duration;
gst_event_parse_gap (event, &timestamp, &duration);
gst_event_unref (event);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
if (GST_CLOCK_TIME_IS_VALID (duration))
timestamp += duration;
gst_audio_rate_fill_to_time (audiorate, timestamp);
}
res = TRUE;
break;
}
default:
res = gst_pad_event_default (pad, parent, event);
break;
}
return res;
}
static gboolean
2011-11-17 11:48:25 +00:00
gst_audio_rate_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
gboolean res;
GstAudioRate *audiorate;
2011-11-17 11:48:25 +00:00
audiorate = GST_AUDIO_RATE (parent);
switch (GST_EVENT_TYPE (event)) {
default:
res = gst_pad_push_event (audiorate->sinkpad, event);
break;
}
return res;
}
static gboolean
gst_audio_rate_convert (GstAudioRate * audiorate,
2011-05-16 11:48:11 +00:00
GstFormat src_fmt, guint64 src_val, GstFormat dest_fmt, guint64 * dest_val)
{
return gst_audio_info_convert (&audiorate->info, src_fmt, src_val, dest_fmt,
(gint64 *) dest_val);
}
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.flags = audiorate->src_segment.flags;
audiorate->sink_segment.applied_rate = audiorate->src_segment.applied_rate;
CONVERT_VAL (start);
CONVERT_VAL (stop);
CONVERT_VAL (time);
2011-05-16 11:48:11 +00:00
CONVERT_VAL (base);
CONVERT_VAL (position);
#undef CONVERT_VAL
return TRUE;
}
static void
gst_audio_rate_notify_drop (GstAudioRate * audiorate)
{
g_object_notify_by_pspec ((GObject *) audiorate, pspec_drop);
}
static void
gst_audio_rate_notify_add (GstAudioRate * audiorate)
{
g_object_notify_by_pspec ((GObject *) audiorate, pspec_add);
}
static GstFlowReturn
2011-11-17 11:48:25 +00:00
gst_audio_rate_chain (GstPad * pad, GstObject * parent, 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;
gint rate, bpf;
GstAudioMeta *meta;
2011-11-17 11:48:25 +00:00
audiorate = GST_AUDIO_RATE (parent);
rate = GST_AUDIO_INFO_RATE (&audiorate->info);
bpf = GST_AUDIO_INFO_BPF (&audiorate->info);
/* need to be negotiated now */
if (bpf == 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_round (audiorate->src_segment.start,
GST_AUDIO_INFO_RATE (&audiorate->info), 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_round (audiorate->next_offset, GST_SECOND,
GST_AUDIO_INFO_RATE (&audiorate->info));
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_round (GST_BUFFER_TIMESTAMP (buf),
GST_AUDIO_INFO_RATE (&audiorate->info), 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);
}
}
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;
}
meta = gst_buffer_get_audio_meta (buf);
in_size = gst_buffer_get_size (buf);
in_samples = meta ? meta->samples : in_size / bpf;
audiorate->in += in_samples;
/* calculate the buffer offset */
in_offset = gst_util_uint64_scale_int_round (in_time, rate, GST_SECOND);
in_offset_end = in_offset + in_samples;
GST_LOG_OBJECT (audiorate,
"in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT
2009-10-09 12:20:47 +00:00
", 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, 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;
audiorate->out += 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, rate);
GstMapInfo fillmap;
fillsamples -= cursamples;
fillsize = cursamples * bpf;
fill = gst_buffer_new_and_alloc (fillsize);
gst_buffer_map (fill, &fillmap, GST_MAP_WRITE);
gst_audio_format_fill_silence (audiorate->info.finfo, fillmap.data,
fillmap.size);
gst_buffer_unmap (fill, &fillmap);
if (audiorate->info.layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (fill, &audiorate->info, cursamples, NULL);
}
2009-10-09 12:20:47 +00:00
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. Necessary complexity to get 'perfect'
* streams */
GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
audiorate->next_ts =
gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND,
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;
}
fill = gst_audio_buffer_clip (fill, &audiorate->src_segment, rate, bpf);
if (fill)
ret = gst_pad_push (audiorate->srcpad, fill);
if (ret != GST_FLOW_OK)
goto beach;
audiorate->out += cursamples;
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_samples;
audiorate->drop += drop;
2009-10-09 12:20:47 +00:00
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, leftsamples;
/* truncate buffer */
truncsamples = audiorate->next_offset - in_offset;
leftsamples = in_samples - truncsamples;
buf = gst_audio_buffer_truncate (buf, bpf, truncsamples, leftsamples);
audiorate->drop += truncsamples;
audiorate->out += leftsamples;
GST_DEBUG_OBJECT (audiorate, "truncating %" G_GUINT64_FORMAT " samples",
truncsamples);
if (!audiorate->silent)
gst_audio_rate_notify_drop (audiorate);
}
}
send:
if (gst_buffer_get_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_round (in_offset_end,
GST_SECOND, 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_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_writable (buf);
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
}
buf = gst_audio_buffer_clip (buf, &audiorate->src_segment, rate, bpf);
if (buf) {
/* set last_stop on segment */
audiorate->src_segment.position =
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
ret = gst_pad_push (audiorate->srcpad, buf);
}
buf = NULL;
audiorate->next_offset = in_offset_end;
beach:
if (buf)
gst_buffer_unref (buf);
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 PROP_SILENT:
audiorate->silent = g_value_get_boolean (value);
break;
case PROP_TOLERANCE:
audiorate->tolerance = g_value_get_uint64 (value);
break;
case PROP_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 PROP_IN:
g_value_set_uint64 (value, audiorate->in);
break;
case PROP_OUT:
g_value_set_uint64 (value, audiorate->out);
break;
case PROP_ADD:
g_value_set_uint64 (value, audiorate->add);
break;
case PROP_DROP:
g_value_set_uint64 (value, audiorate->drop);
break;
case PROP_SILENT:
g_value_set_boolean (value, audiorate->silent);
break;
case PROP_TOLERANCE:
g_value_set_uint64 (value, audiorate->tolerance);
break;
case PROP_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->add = 0;
gst_audio_info_init (&audiorate->info);
gst_audio_rate_reset (audiorate);
break;
default:
break;
}
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
}
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)