mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-27 02:30:35 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/audiomixer/gstaudiomixer.c

1998 lines
64 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2001 Thomas <thomas@apestaart.org>
* 2005,2006 Wim Taymans <wim@fluendo.com>
* 2013 Sebastian Dröge <sebastian@centricular.com>
*
* audiomixer.c: AudioMixer element, N in, one out, samples are added
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-audiomixer
*
* The audiomixer allows to mix several streams into one by adding the data.
* Mixed data is clamped to the min/max values of the data format.
*
* The audiomixer currently mixes all data received on the sinkpads as soon as
* possible without trying to synchronize the streams.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
* ]| This pipeline produces two sine waves mixed together.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstaudiomixer.h"
#include <gst/audio/audio.h>
#include <string.h> /* strcmp */
#include "gstaudiomixerorc.h"
#define GST_CAT_DEFAULT gst_audiomixer_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
typedef struct _GstAudioMixerCollect GstAudioMixerCollect;
struct _GstAudioMixerCollect
{
GstCollectData collect; /* we extend the CollectData */
GstBuffer *buffer; /* current buffer we're mixing,
for comparison with collect.buffer
to see if we need to update our
cached values. */
guint position, size;
guint64 output_offset; /* Offset in output segment that
collect.pos refers to in the
current buffer. */
guint64 next_offset; /* Next expected offset in the input segment */
};
#define DEFAULT_PAD_VOLUME (1.0)
#define DEFAULT_PAD_MUTE (FALSE)
/* some defines for audio processing */
/* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0
* we map 1.0 to VOLUME_UNITY_INT*
*/
#define VOLUME_UNITY_INT8 8 /* internal int for unity 2^(8-5) */
#define VOLUME_UNITY_INT8_BIT_SHIFT 3 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT16 2048 /* internal int for unity 2^(16-5) */
#define VOLUME_UNITY_INT16_BIT_SHIFT 11 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT24 524288 /* internal int for unity 2^(24-5) */
#define VOLUME_UNITY_INT24_BIT_SHIFT 19 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */
#define VOLUME_UNITY_INT32_BIT_SHIFT 27
enum
{
PROP_PAD_0,
PROP_PAD_VOLUME,
PROP_PAD_MUTE
};
G_DEFINE_TYPE (GstAudioMixerPad, gst_audiomixer_pad, GST_TYPE_PAD);
static void
gst_audiomixer_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object);
switch (prop_id) {
case PROP_PAD_VOLUME:
g_value_set_double (value, pad->volume);
break;
case PROP_PAD_MUTE:
g_value_set_boolean (value, pad->mute);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audiomixer_pad_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object);
switch (prop_id) {
case PROP_PAD_VOLUME:
GST_OBJECT_LOCK (pad);
pad->volume = g_value_get_double (value);
pad->volume_i8 = pad->volume * VOLUME_UNITY_INT8;
pad->volume_i16 = pad->volume * VOLUME_UNITY_INT16;
pad->volume_i32 = pad->volume * VOLUME_UNITY_INT32;
GST_OBJECT_UNLOCK (pad);
break;
case PROP_PAD_MUTE:
GST_OBJECT_LOCK (pad);
pad->mute = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audiomixer_pad_class_init (GstAudioMixerPadClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_audiomixer_pad_set_property;
gobject_class->get_property = gst_audiomixer_pad_get_property;
g_object_class_install_property (gobject_class, PROP_PAD_VOLUME,
g_param_spec_double ("volume", "Volume", "Volume of this pad",
0.0, 10.0, DEFAULT_PAD_VOLUME,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PAD_MUTE,
g_param_spec_boolean ("mute", "Mute", "Mute this pad",
DEFAULT_PAD_MUTE,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
}
static void
gst_audiomixer_pad_init (GstAudioMixerPad * pad)
{
pad->volume = DEFAULT_PAD_VOLUME;
pad->mute = DEFAULT_PAD_MUTE;
}
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
#define DEFAULT_BLOCKSIZE (1024)
enum
{
PROP_0,
PROP_FILTER_CAPS,
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT,
PROP_BLOCKSIZE
};
/* elementfactory information */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define CAPS \
GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \
", layout = (string) { interleaved, non-interleaved }"
#else
#define CAPS \
GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \
", layout = (string) { interleaved, non-interleaved }"
#endif
static GstStaticPadTemplate gst_audiomixer_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CAPS)
);
static GstStaticPadTemplate gst_audiomixer_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS (CAPS)
);
static void gst_audiomixer_child_proxy_init (gpointer g_iface,
gpointer iface_data);
#define gst_audiomixer_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstAudioMixer, gst_audiomixer, GST_TYPE_ELEMENT,
G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
gst_audiomixer_child_proxy_init));
static void gst_audiomixer_dispose (GObject * object);
static void gst_audiomixer_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audiomixer_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audiomixer_setcaps (GstAudioMixer * audiomixer,
GstPad * pad, GstCaps * caps);
static gboolean gst_audiomixer_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_audiomixer_sink_query (GstCollectPads * pads,
GstCollectData * pad, GstQuery * query, gpointer user_data);
static gboolean gst_audiomixer_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_audiomixer_sink_event (GstCollectPads * pads,
GstCollectData * pad, GstEvent * event, gpointer user_data);
static GstPad *gst_audiomixer_request_new_pad (GstElement * element,
GstPadTemplate * temp, const gchar * unused, const GstCaps * caps);
static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad);
static GstStateChangeReturn gst_audiomixer_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_audiomixer_do_clip (GstCollectPads * pads,
GstCollectData * data, GstBuffer * buffer, GstBuffer ** out,
gpointer user_data);
static GstFlowReturn gst_audiomixer_collected (GstCollectPads * pads,
gpointer user_data);
/* we can only accept caps that we and downstream can handle.
* if we have filtercaps set, use those to constrain the target caps.
*/
static GstCaps *
gst_audiomixer_sink_getcaps (GstPad * pad, GstCaps * filter)
{
GstAudioMixer *audiomixer;
GstCaps *result, *peercaps, *current_caps, *filter_caps;
GstStructure *s;
gint i, n;
audiomixer = GST_AUDIO_MIXER (GST_PAD_PARENT (pad));
GST_OBJECT_LOCK (audiomixer);
/* take filter */
if ((filter_caps = audiomixer->filter_caps)) {
if (filter)
filter_caps =
gst_caps_intersect_full (filter, filter_caps,
GST_CAPS_INTERSECT_FIRST);
else
gst_caps_ref (filter_caps);
} else {
filter_caps = filter ? gst_caps_ref (filter) : NULL;
}
GST_OBJECT_UNLOCK (audiomixer);
if (filter_caps && gst_caps_is_empty (filter_caps)) {
GST_WARNING_OBJECT (pad, "Empty filter caps");
return filter_caps;
}
/* get the downstream possible caps */
peercaps = gst_pad_peer_query_caps (audiomixer->srcpad, filter_caps);
/* get the allowed caps on this sinkpad */
GST_OBJECT_LOCK (audiomixer);
current_caps =
audiomixer->current_caps ? gst_caps_ref (audiomixer->current_caps) : NULL;
if (current_caps == NULL) {
current_caps = gst_pad_get_pad_template_caps (pad);
if (!current_caps)
current_caps = gst_caps_new_any ();
}
GST_OBJECT_UNLOCK (audiomixer);
if (peercaps) {
/* if the peer has caps, intersect */
GST_DEBUG_OBJECT (audiomixer, "intersecting peer and our caps");
result =
gst_caps_intersect_full (peercaps, current_caps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
gst_caps_unref (current_caps);
} else {
/* the peer has no caps (or there is no peer), just use the allowed caps
* of this sinkpad. */
/* restrict with filter-caps if any */
if (filter_caps) {
GST_DEBUG_OBJECT (audiomixer, "no peer caps, using filtered caps");
result =
gst_caps_intersect_full (filter_caps, current_caps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (current_caps);
} else {
GST_DEBUG_OBJECT (audiomixer, "no peer caps, using our caps");
result = current_caps;
}
}
result = gst_caps_make_writable (result);
n = gst_caps_get_size (result);
for (i = 0; i < n; i++) {
GstStructure *sref;
s = gst_caps_get_structure (result, i);
sref = gst_structure_copy (s);
gst_structure_set (sref, "channels", GST_TYPE_INT_RANGE, 0, 2, NULL);
if (gst_structure_is_subset (s, sref)) {
/* This field is irrelevant when in mono or stereo */
gst_structure_remove_field (s, "channel-mask");
}
gst_structure_free (sref);
}
if (filter_caps)
gst_caps_unref (filter_caps);
GST_LOG_OBJECT (audiomixer, "getting caps on pad %p,%s to %" GST_PTR_FORMAT,
pad, GST_PAD_NAME (pad), result);
return result;
}
static gboolean
gst_audiomixer_sink_query (GstCollectPads * pads, GstCollectData * pad,
GstQuery * query, gpointer user_data)
{
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
caps = gst_audiomixer_sink_getcaps (pad->pad, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
break;
}
default:
res = gst_collect_pads_query_default (pads, pad, query, FALSE);
break;
}
return res;
}
/* the first caps we receive on any of the sinkpads will define the caps for all
* the other sinkpads because we can only mix streams with the same caps.
*/
static gboolean
gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad,
GstCaps * orig_caps)
{
GstCaps *caps;
GstAudioInfo info;
GstStructure *s;
gint channels;
caps = gst_caps_copy (orig_caps);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "channels", &channels))
if (channels <= 2)
gst_structure_remove_field (s, "channel-mask");
if (!gst_audio_info_from_caps (&info, caps))
goto invalid_format;
GST_OBJECT_LOCK (audiomixer);
/* don't allow reconfiguration for now; there's still a race between the
* different upstream threads doing query_caps + accept_caps + sending
* (possibly different) CAPS events, but there's not much we can do about
* that, upstream needs to deal with it. */
if (audiomixer->current_caps != NULL) {
if (gst_audio_info_is_equal (&info, &audiomixer->info)) {
GST_OBJECT_UNLOCK (audiomixer);
gst_caps_unref (caps);
return TRUE;
} else {
GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but "
"current caps are %" GST_PTR_FORMAT, caps, audiomixer->current_caps);
GST_OBJECT_UNLOCK (audiomixer);
gst_pad_push_event (pad, gst_event_new_reconfigure ());
gst_caps_unref (caps);
return FALSE;
}
}
GST_INFO_OBJECT (pad, "setting caps to %" GST_PTR_FORMAT, caps);
gst_caps_replace (&audiomixer->current_caps, caps);
memcpy (&audiomixer->info, &info, sizeof (info));
audiomixer->send_caps = TRUE;
GST_OBJECT_UNLOCK (audiomixer);
/* send caps event later, after stream-start event */
GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps);
gst_caps_unref (caps);
return TRUE;
/* ERRORS */
invalid_format:
{
gst_caps_unref (caps);
GST_WARNING_OBJECT (audiomixer, "invalid format set as caps");
return FALSE;
}
}
/* FIXME, the duration query should reflect how long you will produce
* data, that is the amount of stream time until you will emit EOS.
*
* For synchronized mixing this is always the max of all the durations
* of upstream since we emit EOS when all of them finished.
*
* We don't do synchronized mixing so this really depends on where the
* streams where punched in and what their relative offsets are against
* eachother which we can get from the first timestamps we see.
*
* When we add a new stream (or remove a stream) the duration might
* also become invalid again and we need to post a new DURATION
* message to notify this fact to the parent.
* For now we take the max of all the upstream elements so the simple
* cases work at least somewhat.
*/
static gboolean
gst_audiomixer_query_duration (GstAudioMixer * audiomixer, GstQuery * query)
{
gint64 max;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
GValue item = { 0, };
/* parse format */
gst_query_parse_duration (query, &format, NULL);
max = -1;
res = TRUE;
done = FALSE;
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = g_value_get_object (&item);
gint64 duration;
/* ask sink peer for duration */
res &= gst_pad_peer_query_duration (pad, format, &duration);
/* take max from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
max = duration;
done = TRUE;
}
/* else see if bigger than current max */
else if (duration > max)
max = duration;
}
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
max = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* and store the max */
GST_DEBUG_OBJECT (audiomixer, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
gst_query_set_duration (query, format, max);
}
return res;
}
static gboolean
gst_audiomixer_query_latency (GstAudioMixer * audiomixer, GstQuery * query)
{
GstClockTime min, max;
gboolean live;
gboolean res;
GstIterator *it;
gboolean done;
GValue item = { 0, };
res = TRUE;
done = FALSE;
live = FALSE;
min = 0;
max = GST_CLOCK_TIME_NONE;
/* Take maximum of all latency values */
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = g_value_get_object (&item);
GstQuery *peerquery;
GstClockTime min_cur, max_cur;
gboolean live_cur;
peerquery = gst_query_new_latency ();
/* Ask peer for latency */
res &= gst_pad_peer_query (pad, peerquery);
/* take max from all valid return values */
if (res) {
gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur);
if (min_cur > min)
min = min_cur;
if (max_cur != GST_CLOCK_TIME_NONE &&
((max != GST_CLOCK_TIME_NONE && max_cur > max) ||
(max == GST_CLOCK_TIME_NONE)))
max = max_cur;
live = live || live_cur;
}
gst_query_unref (peerquery);
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
live = FALSE;
min = 0;
max = GST_CLOCK_TIME_NONE;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* store the results */
GST_DEBUG_OBJECT (audiomixer, "Calculated total latency: live %s, min %"
GST_TIME_FORMAT ", max %" GST_TIME_FORMAT,
(live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
return res;
}
static gboolean
gst_audiomixer_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (parent);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
/* FIXME, bring to stream time, might be tricky */
gst_query_set_position (query, format, audiomixer->segment.position);
res = TRUE;
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, format, audiomixer->offset);
res = TRUE;
break;
default:
break;
}
break;
}
case GST_QUERY_DURATION:
res = gst_audiomixer_query_duration (audiomixer, query);
break;
case GST_QUERY_LATENCY:
res = gst_audiomixer_query_latency (audiomixer, query);
break;
default:
/* FIXME, needs a custom query handler because we have multiple
* sinkpads */
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
/* event handling */
typedef struct
{
GstEvent *event;
gboolean flush;
} EventData;
/* FIXME: What is this supposed to solve? */
static gboolean
forward_event_func (const GValue * val, GValue * ret, EventData * data)
{
GstPad *pad = g_value_get_object (val);
GstEvent *event = data->event;
GstPad *peer;
gst_event_ref (event);
GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event));
peer = gst_pad_get_peer (pad);
/* collect pad might have been set flushing,
* so bypass core checking that and send directly to peer */
if (!peer || !gst_pad_send_event (peer, event)) {
if (!peer)
gst_event_unref (event);
GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.",
event, GST_EVENT_TYPE_NAME (event));
/* quick hack to unflush the pads, ideally we need a way to just unflush
* this single collect pad */
if (data->flush)
gst_pad_send_event (pad, gst_event_new_flush_stop (TRUE));
} else {
g_value_set_boolean (ret, TRUE);
GST_LOG_OBJECT (pad, "Sent event %p (%s).",
event, GST_EVENT_TYPE_NAME (event));
}
if (peer)
gst_object_unref (peer);
/* continue on other pads, even if one failed */
return TRUE;
}
/* forwards the event to all sinkpads, takes ownership of the
* event
*
* Returns: TRUE if the event could be forwarded on all
* sinkpads.
*/
static gboolean
forward_event (GstAudioMixer * audiomixer, GstEvent * event, gboolean flush)
{
gboolean ret;
GstIterator *it;
GstIteratorResult ires;
GValue vret = { 0 };
EventData data;
GST_LOG_OBJECT (audiomixer, "Forwarding event %p (%s)", event,
GST_EVENT_TYPE_NAME (event));
data.event = event;
data.flush = flush;
g_value_init (&vret, G_TYPE_BOOLEAN);
g_value_set_boolean (&vret, FALSE);
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer));
while (TRUE) {
ires =
gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func,
&vret, &data);
switch (ires) {
case GST_ITERATOR_RESYNC:
GST_WARNING ("resync");
gst_iterator_resync (it);
g_value_set_boolean (&vret, TRUE);
break;
case GST_ITERATOR_OK:
case GST_ITERATOR_DONE:
ret = g_value_get_boolean (&vret);
goto done;
default:
ret = FALSE;
goto done;
}
}
done:
gst_iterator_free (it);
GST_LOG_OBJECT (audiomixer, "Forwarded event %p (%s), ret=%d", event,
GST_EVENT_TYPE_NAME (event), ret);
gst_event_unref (event);
return ret;
}
static gboolean
gst_audiomixer_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstAudioMixer *audiomixer;
gboolean result;
audiomixer = GST_AUDIO_MIXER (parent);
GST_DEBUG_OBJECT (pad, "Got %s event on src pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
/* TODO: Update from videomixer */
case GST_EVENT_SEEK:
{
GstSeekFlags flags;
gdouble rate;
GstSeekType start_type, stop_type;
gint64 start, stop;
GstFormat seek_format, dest_format;
gboolean flush;
/* parse the seek parameters */
gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type,
&start, &stop_type, &stop);
if ((start_type != GST_SEEK_TYPE_NONE)
&& (start_type != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (audiomixer,
"seeking failed, unhandled seek type for start: %d", start_type);
goto done;
}
if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (audiomixer,
"seeking failed, unhandled seek type for end: %d", stop_type);
goto done;
}
dest_format = audiomixer->segment.format;
if (seek_format != dest_format) {
result = FALSE;
GST_DEBUG_OBJECT (audiomixer,
"seeking failed, unhandled seek format: %d", seek_format);
goto done;
}
flush = (flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH;
/* check if we are flushing */
if (flush) {
/* flushing seek, start flush downstream, the flush will be done
* when all pads received a FLUSH_STOP.
* Make sure we accept nothing anymore and return WRONG_STATE.
* We send a flush-start before, to ensure no streaming is done
* as we need to take the stream lock.
*/
gst_pad_push_event (audiomixer->srcpad, gst_event_new_flush_start ());
gst_collect_pads_set_flushing (audiomixer->collect, TRUE);
/* We can't send FLUSH_STOP here since upstream could start pushing data
* after we unlock audiomixer->collect.
* We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
* forwarding the seek upstream or from gst_audiomixer_collected,
* whichever happens first.
*/
GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect);
audiomixer->flush_stop_pending = TRUE;
GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect);
GST_DEBUG_OBJECT (audiomixer, "mark pending flush stop event");
}
GST_DEBUG_OBJECT (audiomixer, "handling seek event: %" GST_PTR_FORMAT,
event);
/* now wait for the collected to be finished and mark a new
* segment. After we have the lock, no collect function is running and no
* new collect function will be called for as long as we're flushing. */
GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect);
/* clip position and update our segment */
if (audiomixer->segment.stop != -1) {
audiomixer->segment.position = audiomixer->segment.stop;
}
gst_segment_do_seek (&audiomixer->segment, rate, seek_format, flags,
start_type, start, stop_type, stop, NULL);
if (flush) {
/* Yes, we need to call _set_flushing again *WHEN* the streaming threads
* have stopped so that the cookie gets properly updated. */
gst_collect_pads_set_flushing (audiomixer->collect, TRUE);
}
GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect);
GST_DEBUG_OBJECT (audiomixer, "forwarding seek event: %" GST_PTR_FORMAT,
event);
GST_DEBUG_OBJECT (audiomixer, "updated segment: %" GST_SEGMENT_FORMAT,
&audiomixer->segment);
/* we're forwarding seek to all upstream peers and wait for one to reply
* with a newsegment-event before we send a newsegment-event downstream */
g_atomic_int_set (&audiomixer->segment_pending, TRUE);
result = forward_event (audiomixer, event, flush);
/* FIXME: We should use the seek segment and forward that downstream next time
* not any upstream segment event */
if (!result) {
/* seek failed. maybe source is a live source. */
GST_DEBUG_OBJECT (audiomixer, "seeking failed");
}
if (g_atomic_int_compare_and_exchange (&audiomixer->flush_stop_pending,
TRUE, FALSE)) {
GST_DEBUG_OBJECT (audiomixer, "pending flush stop");
if (!gst_pad_push_event (audiomixer->srcpad,
gst_event_new_flush_stop (TRUE))) {
GST_WARNING_OBJECT (audiomixer, "Sending flush stop event failed");
}
}
break;
}
case GST_EVENT_QOS:
/* QoS might be tricky */
result = FALSE;
gst_event_unref (event);
break;
case GST_EVENT_NAVIGATION:
/* navigation is rather pointless. */
result = FALSE;
gst_event_unref (event);
break;
default:
/* just forward the rest for now */
GST_DEBUG_OBJECT (audiomixer, "forward unhandled event: %s",
GST_EVENT_TYPE_NAME (event));
result = forward_event (audiomixer, event, FALSE);
break;
}
done:
return result;
}
static gboolean
gst_audiomixer_sink_event (GstCollectPads * pads, GstCollectData * pad,
GstEvent * event, gpointer user_data)
{
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (user_data);
GstAudioMixerCollect *adata = (GstAudioMixerCollect *) pad;
gboolean res = TRUE, discard = FALSE;
GST_DEBUG_OBJECT (pad->pad, "Got %s event on sink pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
res = gst_audiomixer_setcaps (audiomixer, pad->pad, caps);
gst_event_unref (event);
event = NULL;
break;
}
/* FIXME: Who cares about flushes from upstream? We should
* not forward them at all */
case GST_EVENT_FLUSH_START:
/* ensure that we will send a flush stop */
GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect);
audiomixer->flush_stop_pending = TRUE;
res = gst_collect_pads_event_default (pads, pad, event, discard);
event = NULL;
GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect);
break;
case GST_EVENT_FLUSH_STOP:
/* we received a flush-stop. We will only forward it when
* flush_stop_pending is set, and we will unset it then.
*/
g_atomic_int_set (&audiomixer->segment_pending, TRUE);
GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect);
if (audiomixer->flush_stop_pending) {
GST_DEBUG_OBJECT (pad->pad, "forwarding flush stop");
res = gst_collect_pads_event_default (pads, pad, event, discard);
audiomixer->flush_stop_pending = FALSE;
event = NULL;
gst_buffer_replace (&audiomixer->current_buffer, NULL);
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
} else {
discard = TRUE;
GST_DEBUG_OBJECT (pad->pad, "eating flush stop");
}
GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect);
/* Clear pending tags */
if (audiomixer->pending_events) {
g_list_foreach (audiomixer->pending_events, (GFunc) gst_event_unref,
NULL);
g_list_free (audiomixer->pending_events);
audiomixer->pending_events = NULL;
}
adata->position = adata->size = 0;
adata->output_offset = adata->next_offset = -1;
gst_buffer_replace (&adata->buffer, NULL);
break;
case GST_EVENT_TAG:
/* collect tags here so we can push them out when we collect data */
audiomixer->pending_events =
g_list_append (audiomixer->pending_events, event);
event = NULL;
break;
case GST_EVENT_SEGMENT:{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
if (segment->rate != audiomixer->segment.rate) {
GST_ERROR_OBJECT (pad->pad,
"Got segment event with wrong rate %lf, expected %lf",
segment->rate, audiomixer->segment.rate);
res = FALSE;
gst_event_unref (event);
event = NULL;
} else if (segment->rate < 0.0) {
GST_ERROR_OBJECT (pad->pad, "Negative rates not supported yet");
res = FALSE;
gst_event_unref (event);
event = NULL;
}
discard = TRUE;
break;
}
default:
break;
}
if (G_LIKELY (event))
return gst_collect_pads_event_default (pads, pad, event, discard);
else
return res;
}
static void
gst_audiomixer_class_init (GstAudioMixerClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audiomixer_set_property;
gobject_class->get_property = gst_audiomixer_get_property;
gobject_class->dispose = gst_audiomixer_dispose;
g_object_class_install_property (gobject_class, PROP_FILTER_CAPS,
g_param_spec_boxed ("caps", "Target caps",
"Set target format for mixing (NULL means ANY). "
"Setting this property takes a reference to the supplied GstCaps "
"object", GST_TYPE_CAPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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));
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));
g_object_class_install_property (gobject_class, PROP_BLOCKSIZE,
g_param_spec_uint ("blocksize", "Block Size",
"Output block size in number of samples", 0,
G_MAXUINT, DEFAULT_BLOCKSIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audiomixer_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audiomixer_sink_template));
gst_element_class_set_static_metadata (gstelement_class, "AudioMixer",
"Generic/Audio",
"Mixes multiple audio streams",
"Sebastian Dröge <sebastian@centricular.com>");
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_audiomixer_request_new_pad);
gstelement_class->release_pad =
GST_DEBUG_FUNCPTR (gst_audiomixer_release_pad);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_audiomixer_change_state);
}
static void
gst_audiomixer_init (GstAudioMixer * audiomixer)
{
GstPadTemplate *template;
template = gst_static_pad_template_get (&gst_audiomixer_src_template);
audiomixer->srcpad = gst_pad_new_from_template (template, "src");
gst_object_unref (template);
gst_pad_set_query_function (audiomixer->srcpad,
GST_DEBUG_FUNCPTR (gst_audiomixer_src_query));
gst_pad_set_event_function (audiomixer->srcpad,
GST_DEBUG_FUNCPTR (gst_audiomixer_src_event));
GST_PAD_SET_PROXY_CAPS (audiomixer->srcpad);
gst_element_add_pad (GST_ELEMENT (audiomixer), audiomixer->srcpad);
audiomixer->current_caps = NULL;
gst_audio_info_init (&audiomixer->info);
audiomixer->padcount = 0;
audiomixer->filter_caps = NULL;
audiomixer->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
audiomixer->discont_wait = DEFAULT_DISCONT_WAIT;
audiomixer->blocksize = DEFAULT_BLOCKSIZE;
/* keep track of the sinkpads requested */
audiomixer->collect = gst_collect_pads_new ();
gst_collect_pads_set_function (audiomixer->collect,
GST_DEBUG_FUNCPTR (gst_audiomixer_collected), audiomixer);
gst_collect_pads_set_clip_function (audiomixer->collect,
GST_DEBUG_FUNCPTR (gst_audiomixer_do_clip), audiomixer);
gst_collect_pads_set_event_function (audiomixer->collect,
GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event), audiomixer);
gst_collect_pads_set_query_function (audiomixer->collect,
GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query), audiomixer);
}
static void
gst_audiomixer_dispose (GObject * object)
{
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object);
if (audiomixer->collect) {
gst_object_unref (audiomixer->collect);
audiomixer->collect = NULL;
}
gst_caps_replace (&audiomixer->filter_caps, NULL);
gst_caps_replace (&audiomixer->current_caps, NULL);
if (audiomixer->pending_events) {
g_list_foreach (audiomixer->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (audiomixer->pending_events);
audiomixer->pending_events = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audiomixer_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object);
switch (prop_id) {
case PROP_FILTER_CAPS:{
GstCaps *new_caps = NULL;
GstCaps *old_caps;
const GstCaps *new_caps_val = gst_value_get_caps (value);
if (new_caps_val != NULL) {
new_caps = (GstCaps *) new_caps_val;
gst_caps_ref (new_caps);
}
GST_OBJECT_LOCK (audiomixer);
old_caps = audiomixer->filter_caps;
audiomixer->filter_caps = new_caps;
GST_OBJECT_UNLOCK (audiomixer);
if (old_caps)
gst_caps_unref (old_caps);
GST_DEBUG_OBJECT (audiomixer, "set new caps %" GST_PTR_FORMAT, new_caps);
break;
}
case PROP_ALIGNMENT_THRESHOLD:
audiomixer->alignment_threshold = g_value_get_uint64 (value);
break;
case PROP_DISCONT_WAIT:
audiomixer->discont_wait = g_value_get_uint64 (value);
break;
case PROP_BLOCKSIZE:
audiomixer->blocksize = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audiomixer_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object);
switch (prop_id) {
case PROP_FILTER_CAPS:
GST_OBJECT_LOCK (audiomixer);
gst_value_set_caps (value, audiomixer->filter_caps);
GST_OBJECT_UNLOCK (audiomixer);
break;
case PROP_ALIGNMENT_THRESHOLD:
g_value_set_uint64 (value, audiomixer->alignment_threshold);
break;
case PROP_DISCONT_WAIT:
g_value_set_uint64 (value, audiomixer->discont_wait);
break;
case PROP_BLOCKSIZE:
g_value_set_uint (value, audiomixer->blocksize);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
free_pad (GstCollectData * data)
{
GstAudioMixerCollect *adata = (GstAudioMixerCollect *) data;
gst_buffer_replace (&adata->buffer, NULL);
}
static GstPad *
gst_audiomixer_request_new_pad (GstElement * element, GstPadTemplate * templ,
const gchar * unused, const GstCaps * caps)
{
gchar *name;
GstAudioMixer *audiomixer;
GstPad *newpad;
gint padcount;
GstCollectData *cdata;
GstAudioMixerCollect *adata;
if (templ->direction != GST_PAD_SINK)
goto not_sink;
audiomixer = GST_AUDIO_MIXER (element);
/* increment pad counter */
padcount = g_atomic_int_add (&audiomixer->padcount, 1);
name = g_strdup_printf ("sink_%u", padcount);
newpad = g_object_new (GST_TYPE_AUDIO_MIXER_PAD, "name", name, "direction",
templ->direction, "template", templ, NULL);
GST_DEBUG_OBJECT (audiomixer, "request new pad %s", name);
g_free (name);
cdata =
gst_collect_pads_add_pad (audiomixer->collect, newpad,
sizeof (GstAudioMixerCollect), free_pad, TRUE);
adata = (GstAudioMixerCollect *) cdata;
adata->buffer = NULL;
adata->position = 0;
adata->size = 0;
adata->output_offset = -1;
adata->next_offset = -1;
/* takes ownership of the pad */
if (!gst_element_add_pad (GST_ELEMENT (audiomixer), newpad))
goto could_not_add;
gst_child_proxy_child_added (GST_CHILD_PROXY (audiomixer), G_OBJECT (newpad),
GST_OBJECT_NAME (newpad));
return newpad;
/* errors */
not_sink:
{
g_warning ("gstaudiomixer: request new pad that is not a SINK pad\n");
return NULL;
}
could_not_add:
{
GST_DEBUG_OBJECT (audiomixer, "could not add pad");
gst_collect_pads_remove_pad (audiomixer->collect, newpad);
gst_object_unref (newpad);
return NULL;
}
}
static void
gst_audiomixer_release_pad (GstElement * element, GstPad * pad)
{
GstAudioMixer *audiomixer;
audiomixer = GST_AUDIO_MIXER (element);
GST_DEBUG_OBJECT (audiomixer, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad));
gst_child_proxy_child_removed (GST_CHILD_PROXY (audiomixer), G_OBJECT (pad),
GST_OBJECT_NAME (pad));
if (audiomixer->collect)
gst_collect_pads_remove_pad (audiomixer->collect, pad);
gst_element_remove_pad (element, pad);
}
static GstFlowReturn
gst_audiomixer_do_clip (GstCollectPads * pads, GstCollectData * data,
GstBuffer * buffer, GstBuffer ** out, gpointer user_data)
{
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (user_data);
gint rate, bpf;
rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
buffer = gst_audio_buffer_clip (buffer, &data->segment, rate, bpf);
*out = buffer;
return GST_FLOW_OK;
}
static gboolean
gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstCollectPads * pads,
GstCollectData * collect_data, GstAudioMixerCollect * adata,
GstBuffer * inbuf)
{
GstClockTime start_time, end_time;
gboolean discont = FALSE;
guint64 start_offset, end_offset;
GstClockTime timestamp, stream_time;
gint rate, bpf;
g_assert (adata->buffer == NULL);
rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
stream_time =
gst_segment_to_stream_time (&collect_data->segment, GST_FORMAT_TIME,
timestamp);
/* sync object properties on stream time */
/* TODO: Ideally we would want to do that on every sample */
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (collect_data->pad), stream_time);
adata->position = 0;
adata->size = gst_buffer_get_size (inbuf);
start_time = GST_BUFFER_TIMESTAMP (inbuf);
end_time =
start_time + gst_util_uint64_scale_ceil (adata->size / bpf,
GST_SECOND, rate);
start_offset = gst_util_uint64_scale (start_time, rate, GST_SECOND);
end_offset = start_offset + adata->size / bpf;
if (GST_BUFFER_IS_DISCONT (inbuf)
|| GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC)
|| adata->next_offset == -1) {
discont = TRUE;
} else {
guint64 diff, max_sample_diff;
/* Check discont, based on audiobasesink */
if (start_offset <= adata->next_offset)
diff = adata->next_offset - start_offset;
else
diff = start_offset - adata->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (audiomixer->alignment_threshold, rate,
GST_SECOND);
/* Discont! */
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (audiomixer->discont_wait > 0) {
if (audiomixer->discont_time == GST_CLOCK_TIME_NONE) {
audiomixer->discont_time = start_time;
} else if (start_time - audiomixer->discont_time >=
audiomixer->discont_wait) {
discont = TRUE;
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (audiomixer->discont_time != GST_CLOCK_TIME_NONE)) {
/* we have had a discont, but are now back on track! */
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
/* Have discont, need resync */
if (adata->next_offset != -1)
GST_INFO_OBJECT (collect_data->pad, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
adata->next_offset, start_offset);
adata->output_offset = -1;
} else {
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
}
adata->next_offset = end_offset;
if (adata->output_offset == -1) {
GstClockTime start_running_time;
GstClockTime end_running_time;
guint64 start_running_time_offset;
guint64 end_running_time_offset;
start_running_time =
gst_segment_to_running_time (&collect_data->segment,
GST_FORMAT_TIME, start_time);
end_running_time =
gst_segment_to_running_time (&collect_data->segment,
GST_FORMAT_TIME, end_time);
start_running_time_offset =
gst_util_uint64_scale (start_running_time, rate, GST_SECOND);
end_running_time_offset =
gst_util_uint64_scale (end_running_time, rate, GST_SECOND);
if (end_running_time_offset < audiomixer->offset) {
/* Before output segment, drop */
gst_buffer_unref (inbuf);
adata->buffer = NULL;
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
adata->position = 0;
adata->size = 0;
adata->output_offset = -1;
GST_DEBUG_OBJECT (collect_data->pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GUINT64_FORMAT, end_running_time_offset, audiomixer->offset);
return FALSE;
}
if (start_running_time_offset < audiomixer->offset) {
guint diff = (audiomixer->offset - start_running_time_offset) * bpf;
adata->position += diff;
adata->size -= diff;
/* FIXME: This could only happen due to rounding errors */
if (adata->size == 0) {
/* Empty buffer, drop */
gst_buffer_unref (inbuf);
adata->buffer = NULL;
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
adata->position = 0;
adata->size = 0;
adata->output_offset = -1;
GST_DEBUG_OBJECT (collect_data->pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GUINT64_FORMAT, end_running_time_offset,
audiomixer->offset);
return FALSE;
}
}
adata->output_offset = MAX (start_running_time_offset, audiomixer->offset);
GST_DEBUG_OBJECT (collect_data->pad,
"Buffer resynced: Pad offset %" G_GUINT64_FORMAT
", current mixer offset %" G_GUINT64_FORMAT, adata->output_offset,
audiomixer->offset);
}
GST_LOG_OBJECT (collect_data->pad,
"Queued new buffer at offset %" G_GUINT64_FORMAT, adata->output_offset);
adata->buffer = inbuf;
return TRUE;
}
static void
gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstCollectPads * pads,
GstCollectData * collect_data, GstAudioMixerCollect * adata,
GstMapInfo * outmap)
{
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (adata->collect.pad);
guint overlap;
guint out_start;
GstBuffer *inbuf;
GstMapInfo inmap;
gint bpf;
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
/* Overlap => mix */
if (audiomixer->offset < adata->output_offset)
out_start = adata->output_offset - audiomixer->offset;
else
out_start = 0;
overlap = adata->size / bpf - adata->position / bpf;
if (overlap > audiomixer->blocksize - out_start)
overlap = audiomixer->blocksize - out_start;
inbuf = gst_collect_pads_peek (pads, collect_data);
g_assert (inbuf != NULL && inbuf == adata->buffer);
GST_OBJECT_LOCK (pad);
if (pad->mute || pad->volume < G_MINDOUBLE) {
GST_DEBUG_OBJECT (pad, "Skipping muted pad");
gst_buffer_unref (inbuf);
adata->position += overlap * bpf;
adata->output_offset += overlap;
if (adata->position >= adata->size) {
/* Buffer done, drop it */
gst_buffer_replace (&adata->buffer, NULL);
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
}
GST_OBJECT_UNLOCK (pad);
return;
}
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
/* skip gap buffer */
GST_LOG_OBJECT (pad, "skipping GAP buffer");
gst_buffer_unref (inbuf);
adata->output_offset += adata->size / bpf;
/* Buffer done, drop it */
gst_buffer_replace (&adata->buffer, NULL);
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
GST_OBJECT_UNLOCK (pad);
return;
}
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
GST_LOG_OBJECT (pad, "mixing %u bytes at offset %u from offset %u",
overlap * bpf, out_start * bpf, adata->position);
/* further buffers, need to add them */
if (pad->volume == 1.0) {
switch (audiomixer->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
audiomixer_orc_add_u8 ((gpointer) (outmap->data + out_start * bpf),
(gpointer) (inmap.data + adata->position),
overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_S8:
audiomixer_orc_add_s8 ((gpointer) (outmap->data + out_start * bpf),
(gpointer) (inmap.data + adata->position),
overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_U16:
audiomixer_orc_add_u16 ((gpointer) (outmap->data + out_start * bpf),
(gpointer) (inmap.data + adata->position),
overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_S16:
audiomixer_orc_add_s16 ((gpointer) (outmap->data + out_start * bpf),
(gpointer) (inmap.data + adata->position),
overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_U32:
audiomixer_orc_add_u32 ((gpointer) (outmap->data + out_start * bpf),
(gpointer) (inmap.data + adata->position),
overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_S32:
audiomixer_orc_add_s32 ((gpointer) (outmap->data + out_start * bpf),
(gpointer) (inmap.data + adata->position),
overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_F32:
audiomixer_orc_add_f32 ((gpointer) (outmap->data + out_start * bpf),
(gpointer) (inmap.data + adata->position),
overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_F64:
audiomixer_orc_add_f64 ((gpointer) (outmap->data + out_start * bpf),
(gpointer) (inmap.data + adata->position),
overlap * audiomixer->info.channels);
break;
default:
g_assert_not_reached ();
break;
}
} else {
switch (audiomixer->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
audiomixer_orc_add_volume_u8 ((gpointer) (outmap->data +
out_start * bpf), (gpointer) (inmap.data + adata->position),
pad->volume_i8, overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_S8:
audiomixer_orc_add_volume_s8 ((gpointer) (outmap->data +
out_start * bpf), (gpointer) (inmap.data + adata->position),
pad->volume_i8, overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_U16:
audiomixer_orc_add_volume_u16 ((gpointer) (outmap->data +
out_start * bpf), (gpointer) (inmap.data + adata->position),
pad->volume_i16, overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_S16:
audiomixer_orc_add_volume_s16 ((gpointer) (outmap->data +
out_start * bpf), (gpointer) (inmap.data + adata->position),
pad->volume_i16, overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_U32:
audiomixer_orc_add_volume_u32 ((gpointer) (outmap->data +
out_start * bpf), (gpointer) (inmap.data + adata->position),
pad->volume_i32, overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_S32:
audiomixer_orc_add_volume_s32 ((gpointer) (outmap->data +
out_start * bpf), (gpointer) (inmap.data + adata->position),
pad->volume_i32, overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_F32:
audiomixer_orc_add_volume_f32 ((gpointer) (outmap->data +
out_start * bpf), (gpointer) (inmap.data + adata->position),
pad->volume, overlap * audiomixer->info.channels);
break;
case GST_AUDIO_FORMAT_F64:
audiomixer_orc_add_volume_f64 ((gpointer) (outmap->data +
out_start * bpf), (gpointer) (inmap.data + adata->position),
pad->volume, overlap * audiomixer->info.channels);
break;
default:
g_assert_not_reached ();
break;
}
}
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unref (inbuf);
adata->position += overlap * bpf;
adata->output_offset += overlap;
if (adata->position == adata->size) {
/* Buffer done, drop it */
gst_buffer_replace (&adata->buffer, NULL);
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
GST_DEBUG_OBJECT (pad, "Finished mixing buffer, waiting for next");
}
GST_OBJECT_UNLOCK (pad);
}
static GstFlowReturn
gst_audiomixer_collected (GstCollectPads * pads, gpointer user_data)
{
/* Get all pads that have data for us and store them in a
* new list.
*
* Calculate the current output offset/timestamp and
* offset_end/timestamp_end. Allocate a silence buffer
* for this and store it.
*
* For all pads:
* 1) Once per input buffer (cached)
* 1) Check discont (flag and timestamp with tolerance)
* 2) If discont or new, resync. That means:
* 1) Drop all start data of the buffer that comes before
* the current position/offset.
* 2) Calculate the offset (output segment!) that the first
* frame of the input buffer corresponds to. Base this on
* the running time.
*
* 2) If the current pad's offset/offset_end overlaps with the output
* offset/offset_end, mix it at the appropiate position in the output
* buffer and advance the pad's position. Remember if this pad needs
* a new buffer to advance behind the output offset_end.
*
* 3) If we had no pad with a buffer, go EOS.
*
* 4) If we had at least one pad that did not advance behind output
* offset_end, let collected be called again for the current
* output offset/offset_end.
*/
GstAudioMixer *audiomixer;
GSList *collected;
GstFlowReturn ret;
GstBuffer *outbuf = NULL;
GstMapInfo outmap;
gint64 next_offset;
gint64 next_timestamp;
gint rate, bpf;
gboolean dropped = FALSE;
gboolean is_eos = TRUE;
gboolean is_done = TRUE;
audiomixer = GST_AUDIO_MIXER (user_data);
/* this is fatal */
if (G_UNLIKELY (audiomixer->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN))
goto not_negotiated;
if (audiomixer->flush_stop_pending == TRUE) {
GST_INFO_OBJECT (audiomixer->srcpad, "send pending flush stop event");
if (!gst_pad_push_event (audiomixer->srcpad,
gst_event_new_flush_stop (TRUE))) {
GST_WARNING_OBJECT (audiomixer->srcpad,
"Sending flush stop event failed");
}
audiomixer->flush_stop_pending = FALSE;
gst_buffer_replace (&audiomixer->current_buffer, NULL);
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
}
if (audiomixer->send_stream_start) {
gchar s_id[32];
GST_INFO_OBJECT (audiomixer->srcpad, "send pending stream start event");
/* stream-start (FIXME: create id based on input ids) */
g_snprintf (s_id, sizeof (s_id), "audiomixer-%08x", g_random_int ());
if (!gst_pad_push_event (audiomixer->srcpad,
gst_event_new_stream_start (s_id))) {
GST_WARNING_OBJECT (audiomixer->srcpad,
"Sending stream start event failed");
}
audiomixer->send_stream_start = FALSE;
}
if (audiomixer->send_caps) {
GstEvent *caps_event;
caps_event = gst_event_new_caps (audiomixer->current_caps);
GST_INFO_OBJECT (audiomixer->srcpad,
"send pending caps event %" GST_PTR_FORMAT, caps_event);
if (!gst_pad_push_event (audiomixer->srcpad, caps_event)) {
GST_WARNING_OBJECT (audiomixer->srcpad, "Sending caps event failed");
}
audiomixer->send_caps = FALSE;
}
rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
if (g_atomic_int_compare_and_exchange (&audiomixer->segment_pending, TRUE,
FALSE)) {
GstEvent *event;
/*
* When seeking we set the start and stop positions as given in the seek
* event. We also adjust offset & timestamp accordingly.
* This basically ignores all newsegments sent by upstream.
*
* FIXME: We require that all inputs have the same rate currently
* as we do no rate conversion!
*/
event = gst_event_new_segment (&audiomixer->segment);
if (audiomixer->segment.rate > 0.0) {
audiomixer->segment.position = audiomixer->segment.start;
} else {
audiomixer->segment.position = audiomixer->segment.stop;
}
audiomixer->offset = gst_util_uint64_scale (audiomixer->segment.position,
rate, GST_SECOND);
GST_INFO_OBJECT (audiomixer->srcpad, "sending pending new segment event %"
GST_SEGMENT_FORMAT, &audiomixer->segment);
if (event) {
if (!gst_pad_push_event (audiomixer->srcpad, event)) {
GST_WARNING_OBJECT (audiomixer->srcpad,
"Sending new segment event failed");
}
} else {
GST_WARNING_OBJECT (audiomixer->srcpad, "Creating new segment event for "
"start:%" G_GINT64_FORMAT " end:%" G_GINT64_FORMAT " failed",
audiomixer->segment.start, audiomixer->segment.stop);
}
}
if (G_UNLIKELY (audiomixer->pending_events)) {
GList *tmp = audiomixer->pending_events;
while (tmp) {
GstEvent *ev = (GstEvent *) tmp->data;
gst_pad_push_event (audiomixer->srcpad, ev);
tmp = g_list_next (tmp);
}
g_list_free (audiomixer->pending_events);
audiomixer->pending_events = NULL;
}
/* for the next timestamp, use the sample counter, which will
* never accumulate rounding errors */
/* FIXME: Reverse mixing does not work at all yet */
if (audiomixer->segment.rate > 0.0) {
next_offset = audiomixer->offset + audiomixer->blocksize;
} else {
next_offset = audiomixer->offset - audiomixer->blocksize;
}
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
if (audiomixer->current_buffer) {
outbuf = audiomixer->current_buffer;
} else {
outbuf = gst_buffer_new_and_alloc (audiomixer->blocksize * bpf);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
gst_audio_format_fill_silence (audiomixer->info.finfo, outmap.data,
outmap.size);
gst_buffer_unmap (outbuf, &outmap);
audiomixer->current_buffer = outbuf;
}
GST_LOG_OBJECT (audiomixer,
"Starting to mix %u samples for offset %" G_GUINT64_FORMAT
" with timestamp %" GST_TIME_FORMAT, audiomixer->blocksize,
audiomixer->offset, GST_TIME_ARGS (audiomixer->segment.position));
gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
for (collected = pads->data; collected; collected = collected->next) {
GstCollectData *collect_data;
GstAudioMixerCollect *adata;
GstBuffer *inbuf;
collect_data = (GstCollectData *) collected->data;
adata = (GstAudioMixerCollect *) collect_data;
inbuf = gst_collect_pads_peek (pads, collect_data);
if (!inbuf)
continue;
/* New buffer? */
if (!adata->buffer || adata->buffer != inbuf) {
/* Takes ownership of buffer */
if (!gst_audio_mixer_fill_buffer (audiomixer, pads, collect_data, adata,
inbuf)) {
dropped = TRUE;
continue;
}
} else {
gst_buffer_unref (inbuf);
}
if (!adata->buffer && !dropped
&& GST_COLLECT_PADS_STATE_IS_SET (&adata->collect,
GST_COLLECT_PADS_STATE_EOS)) {
GST_DEBUG_OBJECT (collect_data->pad, "Pad is in EOS state");
} else {
is_eos = FALSE;
}
/* At this point adata->output_offset >= audiomixer->offset or we have no buffer anymore */
if (adata->output_offset >= audiomixer->offset
&& adata->output_offset <
audiomixer->offset + audiomixer->blocksize && adata->buffer) {
GST_LOG_OBJECT (collect_data->pad, "Mixing buffer for current offset");
gst_audio_mixer_mix_buffer (audiomixer, pads, collect_data, adata,
&outmap);
if (adata->output_offset >= next_offset) {
GST_DEBUG_OBJECT (collect_data->pad,
"Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
G_GUINT64_FORMAT, adata->output_offset, next_offset);
} else {
is_done = FALSE;
}
}
}
gst_buffer_unmap (outbuf, &outmap);
if (dropped) {
/* We dropped a buffer, retry */
GST_DEBUG_OBJECT (audiomixer,
"A pad dropped a buffer, wait for the next one");
return GST_FLOW_OK;
}
if (!is_done && !is_eos) {
/* Get more buffers */
GST_DEBUG_OBJECT (audiomixer,
"We're not done yet for the current offset," " waiting for more data");
return GST_FLOW_OK;
}
if (is_eos) {
gint64 max_offset = 0;
gboolean empty_buffer = TRUE;
GST_DEBUG_OBJECT (audiomixer, "We're EOS");
for (collected = pads->data; collected; collected = collected->next) {
GstCollectData *collect_data;
GstAudioMixerCollect *adata;
collect_data = (GstCollectData *) collected->data;
adata = (GstAudioMixerCollect *) collect_data;
max_offset = MAX (max_offset, adata->output_offset);
if (adata->output_offset > audiomixer->offset)
empty_buffer = FALSE;
}
/* This means EOS or no pads at all */
if (empty_buffer) {
gst_buffer_replace (&audiomixer->current_buffer, NULL);
goto eos;
}
if (max_offset <= next_offset) {
GST_DEBUG_OBJECT (audiomixer,
"Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
G_GUINT64_FORMAT, max_offset, next_offset);
next_offset = max_offset;
gst_buffer_resize (outbuf, 0, (next_offset - audiomixer->offset) * bpf);
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
}
}
/* set timestamps on the output buffer */
if (audiomixer->segment.rate > 0.0) {
GST_BUFFER_TIMESTAMP (outbuf) = audiomixer->segment.position;
GST_BUFFER_OFFSET (outbuf) = audiomixer->offset;
GST_BUFFER_OFFSET_END (outbuf) = next_offset;
GST_BUFFER_DURATION (outbuf) =
next_timestamp - audiomixer->segment.position;
} else {
GST_BUFFER_TIMESTAMP (outbuf) = next_timestamp;
GST_BUFFER_OFFSET (outbuf) = next_offset;
GST_BUFFER_OFFSET_END (outbuf) = audiomixer->offset;
GST_BUFFER_DURATION (outbuf) =
audiomixer->segment.position - next_timestamp;
}
audiomixer->offset = next_offset;
audiomixer->segment.position = next_timestamp;
/* send it out */
GST_LOG_OBJECT (audiomixer,
"pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_BUFFER_OFFSET (outbuf));
ret = gst_pad_push (audiomixer->srcpad, outbuf);
audiomixer->current_buffer = NULL;
GST_LOG_OBJECT (audiomixer, "pushed outbuf, result = %s",
gst_flow_get_name (ret));
if (ret == GST_FLOW_OK && is_eos)
goto eos;
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (audiomixer, STREAM, FORMAT, (NULL),
("Unknown data received, not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
eos:
{
GST_DEBUG_OBJECT (audiomixer, "EOS");
gst_pad_push_event (audiomixer->srcpad, gst_event_new_eos ());
return GST_FLOW_EOS;
}
}
static GstStateChangeReturn
gst_audiomixer_change_state (GstElement * element, GstStateChange transition)
{
GstAudioMixer *audiomixer;
GstStateChangeReturn ret;
audiomixer = GST_AUDIO_MIXER (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
audiomixer->offset = 0;
audiomixer->flush_stop_pending = FALSE;
audiomixer->segment_pending = TRUE;
audiomixer->send_stream_start = TRUE;
audiomixer->send_caps = TRUE;
gst_caps_replace (&audiomixer->current_caps, NULL);
gst_segment_init (&audiomixer->segment, GST_FORMAT_TIME);
gst_collect_pads_start (audiomixer->collect);
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* need to unblock the collectpads before calling the
* parent change_state so that streaming can finish */
gst_collect_pads_stop (audiomixer->collect);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_buffer_replace (&audiomixer->current_buffer, NULL);
break;
default:
break;
}
return ret;
}
/* GstChildProxy implementation */
static GObject *
gst_audiomixer_child_proxy_get_child_by_index (GstChildProxy * child_proxy,
guint index)
{
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy);
GObject *obj = NULL;
GST_OBJECT_LOCK (audiomixer);
obj = g_list_nth_data (GST_ELEMENT_CAST (audiomixer)->sinkpads, index);
if (obj)
gst_object_ref (obj);
GST_OBJECT_UNLOCK (audiomixer);
return obj;
}
static guint
gst_audiomixer_child_proxy_get_children_count (GstChildProxy * child_proxy)
{
guint count = 0;
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy);
GST_OBJECT_LOCK (audiomixer);
count = GST_ELEMENT_CAST (audiomixer)->numsinkpads;
GST_OBJECT_UNLOCK (audiomixer);
GST_INFO_OBJECT (audiomixer, "Children Count: %d", count);
return count;
}
static void
gst_audiomixer_child_proxy_init (gpointer g_iface, gpointer iface_data)
{
GstChildProxyInterface *iface = g_iface;
GST_INFO ("intializing child proxy interface");
iface->get_child_by_index = gst_audiomixer_child_proxy_get_child_by_index;
iface->get_children_count = gst_audiomixer_child_proxy_get_children_count;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "audiomixer", 0,
"audio mixing element");
if (!gst_element_register (plugin, "audiomixer", GST_RANK_NONE,
GST_TYPE_AUDIO_MIXER))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audiomixer,
"Mixes multiple audio streams",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
Reference in a new issue View git blame Copy permalink