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alevel: New audio/video level element

Browse source 
The videoframe-audiolevel element acts like a synchronized audio/video "level"
element. For each video frame, it posts a level-style message containing the
RMS value of the corresponding audio frames. This element needs both video and
audio to pass through it. Furthermore, it needs a queue after its video
source.

https://bugzilla.gnome.org/show_bug.cgi?id=748259
This commit is contained in:
Vivia Nikolaidou 2015-04-21 21:09:19 +03:00 committed by Sebastian Dröge
parent c564a043d1
commit 978bcd7181
6 changed files with 1513 additions and 0 deletions
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2
configure.ac
View file

@ -435,6 +435,7 @@ AG_GST_CHECK_PLUGIN(accurip)
AG_GST_CHECK_PLUGIN(adpcmdec)
AG_GST_CHECK_PLUGIN(adpcmenc)
AG_GST_CHECK_PLUGIN(aiff)
AG_GST_CHECK_PLUGIN(videoframe_audiolevel)
AG_GST_CHECK_PLUGIN(asfmux)
AG_GST_CHECK_PLUGIN(audiofxbad)
AG_GST_CHECK_PLUGIN(audiomixer)
@ -3415,6 +3416,7 @@ gst/accurip/Makefile
gst/adpcmdec/Makefile
gst/adpcmenc/Makefile
gst/aiff/Makefile
gst/videoframe_audiolevel/Makefile
gst/asfmux/Makefile
gst/audiofxbad/Makefile
gst/audiomixer/Makefile

9
gst/videoframe_audiolevel/Makefile.am Normal file
View file

@ -0,0 +1,9 @@
plugin_LTLIBRARIES = libgstvideoframe_audiolevel.la
libgstvideoframe_audiolevel_la_SOURCES = gstvideoframe-audiolevel.c
libgstvideoframe_audiolevel_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstvideoframe_audiolevel_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(GST_BASE_LIBS) $(LIBM)
libgstvideoframe_audiolevel_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstvideoframe_audiolevel_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstvideoframe-audiolevel.h

785
gst/videoframe_audiolevel/gstvideoframe-audiolevel.c Normal file
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@ -0,0 +1,785 @@
/*
* GStreamer
* Copyright (C) 2015 Vivia Nikolaidou <vivia@toolsonair.com>
*
* Based on gstlevel.c:
* Copyright (C) 2000,2001,2002,2003,2005
* Thomas Vander Stichele <thomas at apestaart dot org>
*
* 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-videoframe-audiolevel
*
* This element acts like a synchronized audio/video "level". It gathers
* all audio buffers sent between two video frames, and then sends a message
* that contains the RMS value of all samples for these buffers.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 -m filesrc location="file.mkv" ! decodebin name=d ! "audio/x-raw" ! videoframe-audiolevel name=l ! autoaudiosink d. ! "video/x-raw" ! l. l. ! queue ! autovideosink ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
/* FIXME 2.0: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include "gstvideoframe-audiolevel.h"
#include <math.h>
#define GST_CAT_DEFAULT gst_videoframe_audiolevel_debug
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
# define FORMATS "{ S8, S16LE, S32LE, F32LE, F64LE }"
#else
# define FORMATS "{ S8, S16BE, S32BE, F32BE, F64BE }"
#endif
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static GstStaticPadTemplate audio_sink_template =
GST_STATIC_PAD_TEMPLATE ("asink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (FORMATS))
);
static GstStaticPadTemplate audio_src_template =
GST_STATIC_PAD_TEMPLATE ("asrc",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (FORMATS))
);
static GstStaticPadTemplate video_sink_template =
GST_STATIC_PAD_TEMPLATE ("vsink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-raw")
);
static GstStaticPadTemplate video_src_template =
GST_STATIC_PAD_TEMPLATE ("vsrc",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-raw")
);
#define parent_class gst_videoframe_audiolevel_parent_class
G_DEFINE_TYPE (GstVideoFrameAudioLevel, gst_videoframe_audiolevel,
GST_TYPE_ELEMENT);
static GstFlowReturn gst_videoframe_audiolevel_asink_chain (GstPad * pad,
GstObject * parent, GstBuffer * inbuf);
static GstFlowReturn gst_videoframe_audiolevel_vsink_chain (GstPad * pad,
GstObject * parent, GstBuffer * inbuf);
static gboolean gst_videoframe_audiolevel_asink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_videoframe_audiolevel_vsink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static GstIterator *gst_videoframe_audiolevel_iterate_internal_links (GstPad *
pad, GstObject * parent);
static void gst_videoframe_audiolevel_finalize (GObject * gobject);
static GstStateChangeReturn gst_videoframe_audiolevel_change_state (GstElement *
element, GstStateChange transition);
static void
gst_videoframe_audiolevel_class_init (GstVideoFrameAudioLevelClass * klass)
{
GstElementClass *gstelement_class;
GObjectClass *gobject_class = (GObjectClass *) klass;
GST_DEBUG_CATEGORY_INIT (gst_videoframe_audiolevel_debug,
"videoframe-audiolevel", 0, "Synchronized audio/video level");
gstelement_class = (GstElementClass *) klass;
gst_element_class_set_static_metadata (gstelement_class,
"Video-frame audio level", "Filter/Analyzer/Audio",
"Synchronized audio/video RMS Level messenger for audio/raw",
"Vivia Nikolaidou <vivia@toolsonair.com>");
gobject_class->finalize = gst_videoframe_audiolevel_finalize;
gstelement_class->change_state = gst_videoframe_audiolevel_change_state;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&audio_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&audio_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&video_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&video_sink_template));
}
static void
gst_videoframe_audiolevel_init (GstVideoFrameAudioLevel * self)
{
self->asinkpad =
gst_pad_new_from_static_template (&audio_sink_template, "asink");
gst_pad_set_chain_function (self->asinkpad,
GST_DEBUG_FUNCPTR (gst_videoframe_audiolevel_asink_chain));
gst_pad_set_event_function (self->asinkpad,
GST_DEBUG_FUNCPTR (gst_videoframe_audiolevel_asink_event));
gst_pad_set_iterate_internal_links_function (self->asinkpad,
GST_DEBUG_FUNCPTR (gst_videoframe_audiolevel_iterate_internal_links));
gst_element_add_pad (GST_ELEMENT (self), self->asinkpad);
self->vsinkpad =
gst_pad_new_from_static_template (&video_sink_template, "vsink");
gst_pad_set_chain_function (self->vsinkpad,
GST_DEBUG_FUNCPTR (gst_videoframe_audiolevel_vsink_chain));
gst_pad_set_event_function (self->vsinkpad,
GST_DEBUG_FUNCPTR (gst_videoframe_audiolevel_vsink_event));
gst_pad_set_iterate_internal_links_function (self->vsinkpad,
GST_DEBUG_FUNCPTR (gst_videoframe_audiolevel_iterate_internal_links));
gst_element_add_pad (GST_ELEMENT (self), self->vsinkpad);
self->asrcpad =
gst_pad_new_from_static_template (&audio_src_template, "asrc");
gst_pad_set_iterate_internal_links_function (self->asrcpad,
GST_DEBUG_FUNCPTR (gst_videoframe_audiolevel_iterate_internal_links));
gst_element_add_pad (GST_ELEMENT (self), self->asrcpad);
self->vsrcpad =
gst_pad_new_from_static_template (&video_src_template, "vsrc");
gst_pad_set_iterate_internal_links_function (self->vsrcpad,
GST_DEBUG_FUNCPTR (gst_videoframe_audiolevel_iterate_internal_links));
gst_element_add_pad (GST_ELEMENT (self), self->vsrcpad);
GST_PAD_SET_PROXY_CAPS (self->asinkpad);
GST_PAD_SET_PROXY_ALLOCATION (self->asinkpad);
GST_PAD_SET_PROXY_CAPS (self->asrcpad);
GST_PAD_SET_PROXY_SCHEDULING (self->asrcpad);
GST_PAD_SET_PROXY_CAPS (self->vsinkpad);
GST_PAD_SET_PROXY_ALLOCATION (self->vsinkpad);
GST_PAD_SET_PROXY_CAPS (self->vsrcpad);
GST_PAD_SET_PROXY_SCHEDULING (self->vsrcpad);
self->adapter = gst_adapter_new ();
g_queue_init (&self->vtimeq);
self->first_time = GST_CLOCK_TIME_NONE;
self->total_frames = 0;
/* alignment_threshold and discont_wait should become properties if needed */
self->alignment_threshold = 40 * GST_MSECOND;
self->discont_time = GST_CLOCK_TIME_NONE;
self->next_offset = -1;
self->discont_wait = 1 * GST_SECOND;
self->video_eos_flag = FALSE;
self->audio_flush_flag = FALSE;
self->shutdown_flag = FALSE;
g_mutex_init (&self->mutex);
g_cond_init (&self->cond);
}
static GstStateChangeReturn
gst_videoframe_audiolevel_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret;
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (element);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
g_mutex_lock (&self->mutex);
self->shutdown_flag = TRUE;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
g_mutex_lock (&self->mutex);
self->shutdown_flag = FALSE;
self->video_eos_flag = FALSE;
self->audio_flush_flag = FALSE;
g_mutex_unlock (&self->mutex);
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
g_mutex_lock (&self->mutex);
self->first_time = GST_CLOCK_TIME_NONE;
self->total_frames = 0;
gst_segment_init (&self->asegment, GST_FORMAT_UNDEFINED);
gst_segment_init (&self->vsegment, GST_FORMAT_UNDEFINED);
self->vsegment.position = GST_CLOCK_TIME_NONE;
gst_adapter_clear (self->adapter);
g_queue_foreach (&self->vtimeq, (GFunc) g_free, NULL);
g_queue_clear (&self->vtimeq);
if (self->CS) {
g_free (self->CS);
self->CS = NULL;
}
g_mutex_unlock (&self->mutex);
break;
default:
break;
}
return ret;
}
static void
gst_videoframe_audiolevel_finalize (GObject * object)
{
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (object);
if (self->adapter) {
g_object_unref (self->adapter);
self->adapter = NULL;
}
g_queue_foreach (&self->vtimeq, (GFunc) g_free, NULL);
g_queue_clear (&self->vtimeq);
self->first_time = GST_CLOCK_TIME_NONE;
self->total_frames = 0;
if (self->CS) {
g_free (self->CS);
self->CS = NULL;
}
g_mutex_clear (&self->mutex);
g_cond_clear (&self->cond);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
#define DEFINE_INT_LEVEL_CALCULATOR(TYPE, RESOLUTION) \
static void inline \
gst_videoframe_audiolevel_calculate_##TYPE (gpointer data, guint num, guint channels, \
gdouble *NCS) \
{ \
TYPE * in = (TYPE *)data; \
register guint j; \
gdouble squaresum = 0.0; /* square sum of the input samples */ \
register gdouble square = 0.0; /* Square */ \
gdouble normalizer; /* divisor to get a [-1.0, 1.0] range */ \
\
/* *NCS = 0.0; Normalized Cumulative Square */ \
\
for (j = 0; j < num; j += channels) { \
square = ((gdouble) in[j]) * in[j]; \
squaresum += square; \
} \
\
normalizer = (gdouble) (G_GINT64_CONSTANT(1) << (RESOLUTION * 2)); \
*NCS = squaresum / normalizer; \
}
DEFINE_INT_LEVEL_CALCULATOR (gint32, 31);
DEFINE_INT_LEVEL_CALCULATOR (gint16, 15);
DEFINE_INT_LEVEL_CALCULATOR (gint8, 7);
#define DEFINE_FLOAT_LEVEL_CALCULATOR(TYPE) \
static void inline \
gst_videoframe_audiolevel_calculate_##TYPE (gpointer data, guint num, guint channels, \
gdouble *NCS) \
{ \
TYPE * in = (TYPE *)data; \
register guint j; \
gdouble squaresum = 0.0; /* square sum of the input samples */ \
register gdouble square = 0.0; /* Square */ \
\
/* *NCS = 0.0; Normalized Cumulative Square */ \
\
for (j = 0; j < num; j += channels) { \
square = ((gdouble) in[j]) * in[j]; \
squaresum += square; \
} \
\
*NCS = squaresum; \
}
DEFINE_FLOAT_LEVEL_CALCULATOR (gfloat);
DEFINE_FLOAT_LEVEL_CALCULATOR (gdouble);
static gboolean
gst_videoframe_audiolevel_vsink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (parent);
GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
g_mutex_lock (&self->mutex);
g_queue_foreach (&self->vtimeq, (GFunc) g_free, NULL);
g_queue_clear (&self->vtimeq);
g_mutex_unlock (&self->mutex);
gst_event_copy_segment (event, &self->vsegment);
if (self->vsegment.format != GST_FORMAT_TIME)
return FALSE;
self->vsegment.position = GST_CLOCK_TIME_NONE;
break;
case GST_EVENT_GAP:
return TRUE;
case GST_EVENT_EOS:
g_mutex_lock (&self->mutex);
self->video_eos_flag = TRUE;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex);
break;
case GST_EVENT_FLUSH_STOP:
g_mutex_lock (&self->mutex);
g_queue_foreach (&self->vtimeq, (GFunc) g_free, NULL);
g_queue_clear (&self->vtimeq);
gst_segment_init (&self->vsegment, GST_FORMAT_UNDEFINED);
g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex);
self->vsegment.position = GST_CLOCK_TIME_NONE;
break;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
static gboolean
gst_videoframe_audiolevel_asink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (parent);
GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
self->first_time = GST_CLOCK_TIME_NONE;
self->total_frames = 0;
gst_adapter_clear (self->adapter);
gst_event_copy_segment (event, &self->asegment);
if (self->asegment.format != GST_FORMAT_TIME)
return FALSE;
break;
case GST_EVENT_FLUSH_START:
g_mutex_lock (&self->mutex);
self->audio_flush_flag = TRUE;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex);
break;
case GST_EVENT_FLUSH_STOP:
self->audio_flush_flag = FALSE;
self->total_frames = 0;
self->first_time = GST_CLOCK_TIME_NONE;
gst_adapter_clear (self->adapter);
gst_segment_init (&self->asegment, GST_FORMAT_UNDEFINED);
break;
case GST_EVENT_CAPS:{
GstCaps *caps;
gint channels;
gst_event_parse_caps (event, &caps);
GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps);
if (!gst_audio_info_from_caps (&self->ainfo, caps))
return FALSE;
switch (GST_AUDIO_INFO_FORMAT (&self->ainfo)) {
case GST_AUDIO_FORMAT_S8:
self->process = gst_videoframe_audiolevel_calculate_gint8;
break;
case GST_AUDIO_FORMAT_S16:
self->process = gst_videoframe_audiolevel_calculate_gint16;
break;
case GST_AUDIO_FORMAT_S32:
self->process = gst_videoframe_audiolevel_calculate_gint32;
break;
case GST_AUDIO_FORMAT_F32:
self->process = gst_videoframe_audiolevel_calculate_gfloat;
break;
case GST_AUDIO_FORMAT_F64:
self->process = gst_videoframe_audiolevel_calculate_gdouble;
break;
default:
self->process = NULL;
break;
}
gst_adapter_clear (self->adapter);
channels = GST_AUDIO_INFO_CHANNELS (&self->ainfo);
self->first_time = GST_CLOCK_TIME_NONE;
self->total_frames = 0;
if (self->CS)
g_free (self->CS);
self->CS = g_new0 (gdouble, channels);
break;
}
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
static GstMessage *
update_rms_from_buffer (GstVideoFrameAudioLevel * self, GstBuffer * inbuf)
{
GstMapInfo map;
guint8 *in_data;
gsize in_size;
gdouble CS;
guint i;
guint num_frames, frames;
guint num_int_samples = 0; /* number of interleaved samples
* ie. total count for all channels combined */
gint channels, rate, bps;
GValue v = G_VALUE_INIT;
GValue va = G_VALUE_INIT;
GValueArray *a;
GstStructure *s;
GstMessage *msg;
GstClockTime duration, running_time;
channels = GST_AUDIO_INFO_CHANNELS (&self->ainfo);
bps = GST_AUDIO_INFO_BPS (&self->ainfo);
rate = GST_AUDIO_INFO_RATE (&self->ainfo);
gst_buffer_map (inbuf, &map, GST_MAP_READ);
in_data = map.data;
in_size = map.size;
num_int_samples = in_size / bps;
GST_LOG_OBJECT (self, "analyzing %u sample frames at ts %" GST_TIME_FORMAT,
num_int_samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)));
g_return_val_if_fail (num_int_samples % channels == 0, NULL);
num_frames = num_int_samples / channels;
frames = num_frames;
duration = GST_FRAMES_TO_CLOCK_TIME (frames, rate);
if (num_frames > 0) {
for (i = 0; i < channels; ++i) {
self->process (in_data + (bps * i), num_int_samples, channels, &CS);
GST_LOG_OBJECT (self,
"[%d]: cumulative squares %lf, over %d samples/%d channels",
i, CS, num_int_samples, channels);
self->CS[i] += CS;
}
in_data += num_frames * bps;
self->total_frames += num_frames;
}
running_time =
self->first_time + gst_util_uint64_scale (self->total_frames, GST_SECOND,
rate);
a = g_value_array_new (channels);
s = gst_structure_new ("videoframe-audiolevel", "running-time", G_TYPE_UINT64,
running_time, "duration", G_TYPE_UINT64, duration, NULL);
g_value_init (&v, G_TYPE_DOUBLE);
g_value_init (&va, G_TYPE_VALUE_ARRAY);
for (i = 0; i < channels; i++) {
gdouble rms = sqrt (self->CS[i] / frames);
if (frames == 0 && self->CS[i] == 0) {
rms = 0; /* empty buffer */
}
self->CS[i] = 0.0;
g_value_set_double (&v, rms);
g_value_array_append (a, &v);
}
g_value_take_boxed (&va, a);
gst_structure_take_value (s, "rms", &va);
msg = gst_message_new_element (GST_OBJECT (self), s);
gst_buffer_unmap (inbuf, &map);
return msg;
}
static GstFlowReturn
gst_videoframe_audiolevel_vsink_chain (GstPad * pad, GstObject * parent,
GstBuffer * inbuf)
{
GstClockTime timestamp;
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (parent);
GstClockTime duration;
GstClockTime *ptrtime = g_new (GstClockTime, 1);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
*ptrtime =
gst_segment_to_running_time (&self->vsegment, GST_FORMAT_TIME, timestamp);
g_mutex_lock (&self->mutex);
self->vsegment.position = timestamp;
duration = GST_BUFFER_DURATION (inbuf);
if (duration != GST_CLOCK_TIME_NONE)
self->vsegment.position += duration;
g_queue_push_tail (&self->vtimeq, ptrtime);
g_cond_signal (&self->cond);
GST_DEBUG_OBJECT (pad, "Pushed a frame");
g_mutex_unlock (&self->mutex);
return gst_pad_push (self->vsrcpad, inbuf);
}
static GstFlowReturn
gst_videoframe_audiolevel_asink_chain (GstPad * pad, GstObject * parent,
GstBuffer * inbuf)
{
GstClockTime timestamp, cur_time;
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (parent);
GstBuffer *buf;
gsize inbuf_size;
guint64 start_offset, end_offset;
GstClockTime running_time;
gint rate, bpf;
gboolean discont = FALSE;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
running_time =
gst_segment_to_running_time (&self->asegment, GST_FORMAT_TIME, timestamp);
rate = GST_AUDIO_INFO_RATE (&self->ainfo);
bpf = GST_AUDIO_INFO_BPF (&self->ainfo);
start_offset = gst_util_uint64_scale (timestamp, rate, GST_SECOND);
inbuf_size = gst_buffer_get_size (inbuf);
end_offset = start_offset + inbuf_size / bpf;
g_mutex_lock (&self->mutex);
if (GST_BUFFER_IS_DISCONT (inbuf)
|| GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC)
|| self->first_time == GST_CLOCK_TIME_NONE) {
discont = TRUE;
} else {
guint64 diff, max_sample_diff;
/* Check discont, based on audiobasesink */
if (start_offset <= self->next_offset)
diff = self->next_offset - start_offset;
else
diff = start_offset - self->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (self->alignment_threshold, rate, GST_SECOND);
/* Discont! */
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (self->discont_wait > 0) {
if (self->discont_time == GST_CLOCK_TIME_NONE) {
self->discont_time = timestamp;
} else if (timestamp - self->discont_time >= self->discont_wait) {
discont = TRUE;
self->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (self->discont_time != GST_CLOCK_TIME_NONE)) {
/* we have had a discont, but are now back on track! */
self->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
/* Have discont, need resync */
if (self->next_offset != -1)
GST_INFO_OBJECT (pad, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
self->next_offset, start_offset);
self->total_frames = 0;
self->first_time = running_time;
self->next_offset = end_offset;
} else {
self->next_offset += inbuf_size / bpf;
}
gst_adapter_push (self->adapter, gst_buffer_ref (inbuf));
GST_DEBUG_OBJECT (self, "Queue length %i",
g_queue_get_length (&self->vtimeq));
while (TRUE) {
GstClockTime *vt0, *vt1;
GstClockTime vtemp;
GstMessage *msg;
gsize bytes, available_bytes;
vtemp = GST_CLOCK_TIME_NONE;
while (!(g_queue_get_length (&self->vtimeq) >= 2 || self->video_eos_flag
|| self->audio_flush_flag || self->shutdown_flag))
g_cond_wait (&self->cond, &self->mutex);
if (self->audio_flush_flag || self->shutdown_flag) {
g_mutex_unlock (&self->mutex);
gst_buffer_unref (inbuf);
return GST_FLOW_FLUSHING;
} else if (self->video_eos_flag) {
GST_DEBUG_OBJECT (self, "Video EOS flag alert");
/* nothing to do here if queue is empty */
if (g_queue_get_length (&self->vtimeq) == 0)
break;
if (g_queue_get_length (&self->vtimeq) < 2) {
vtemp = self->vsegment.position;
} else if (self->vsegment.position == GST_CLOCK_TIME_NONE) {
/* g_queue_get_length is surely >= 2 at this point
* so the adapter isn't empty */
buf =
gst_adapter_take_buffer (self->adapter,
gst_adapter_available (self->adapter));
if (buf != NULL) {
GstMessage *msg;
msg = update_rms_from_buffer (self, buf);
g_mutex_unlock (&self->mutex);
gst_element_post_message (GST_ELEMENT (self), msg);
gst_buffer_unref (buf);
g_mutex_lock (&self->mutex); /* we unlock again later */
}
break;
}
} else if (g_queue_get_length (&self->vtimeq) < 2) {
continue;
}
vt0 = g_queue_pop_head (&self->vtimeq);
if (vtemp == GST_CLOCK_TIME_NONE)
vt1 = g_queue_peek_head (&self->vtimeq);
else
vt1 = &vtemp;
cur_time =
self->first_time + gst_util_uint64_scale (self->total_frames,
GST_SECOND, rate);
GST_DEBUG_OBJECT (self,
"Processing: current time is %" GST_TIME_FORMAT,
GST_TIME_ARGS (cur_time));
GST_DEBUG_OBJECT (self, "Total frames is %i with a rate of %d",
self->total_frames, rate);
GST_DEBUG_OBJECT (self, "Start time is %" GST_TIME_FORMAT,
GST_TIME_ARGS (self->first_time));
GST_DEBUG_OBJECT (self, "Time on top is %" GST_TIME_FORMAT,
GST_TIME_ARGS (*vt0));
if (cur_time < *vt0) {
guint num_frames =
gst_util_uint64_scale (*vt0 - cur_time, rate, GST_SECOND);
bytes = num_frames * GST_AUDIO_INFO_BPF (&self->ainfo);
available_bytes = gst_adapter_available (self->adapter);
if (available_bytes == 0) {
g_queue_push_head (&self->vtimeq, vt0);
break;
}
if (bytes == 0) {
cur_time = *vt0;
} else {
GST_DEBUG_OBJECT (self, "Flushed %ld out of %ld bytes", bytes,
available_bytes);
gst_adapter_flush (self->adapter, MIN (bytes, available_bytes));
self->total_frames += num_frames;
if (available_bytes <= bytes) {
g_queue_push_head (&self->vtimeq, vt0);
break;
}
cur_time =
self->first_time + gst_util_uint64_scale (self->total_frames,
GST_SECOND, rate);
}
}
if (*vt1 > cur_time) {
bytes =
GST_AUDIO_INFO_BPF (&self->ainfo) * gst_util_uint64_scale (*vt1 -
cur_time, rate, GST_SECOND);
} else {
bytes = 0; /* We just need to discard vt0 */
}
available_bytes = gst_adapter_available (self->adapter);
GST_DEBUG_OBJECT (self, "Adapter contains %ld out of %ld bytes",
available_bytes, bytes);
if (available_bytes < bytes) {
g_queue_push_head (&self->vtimeq, vt0);
goto done;
}
if (bytes > 0) {
buf = gst_adapter_take_buffer (self->adapter, bytes);
g_assert (buf != NULL);
} else {
/* Just an empty buffer */
buf = gst_buffer_new ();
}
msg = update_rms_from_buffer (self, buf);
g_mutex_unlock (&self->mutex);
gst_element_post_message (GST_ELEMENT (self), msg);
g_mutex_lock (&self->mutex);
gst_buffer_unref (buf);
g_free (vt0);
if (available_bytes == bytes)
break;
}
done:
g_mutex_unlock (&self->mutex);
return gst_pad_push (self->asrcpad, inbuf);
}
static GstIterator *
gst_videoframe_audiolevel_iterate_internal_links (GstPad * pad,
GstObject * parent)
{
GstIterator *it = NULL;
GstPad *opad;
GValue val = { 0, };
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (parent);
if (self->asinkpad == pad)
opad = gst_object_ref (self->asrcpad);
else if (self->asrcpad == pad)
opad = gst_object_ref (self->asinkpad);
else if (self->vsinkpad == pad)
opad = gst_object_ref (self->vsrcpad);
else if (self->vsrcpad == pad)
opad = gst_object_ref (self->vsinkpad);
else
goto out;
g_value_init (&val, GST_TYPE_PAD);
g_value_set_object (&val, opad);
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
g_value_unset (&val);
gst_object_unref (opad);
out:
return it;
}
static gboolean
gst_videoframe_audiolevel_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "videoframe-audiolevel",
GST_RANK_NONE, GST_TYPE_VIDEOFRAME_AUDIOLEVEL);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
videoframe - audiolevel,
"Video frame-synchronized audio level",
gst_videoframe_audiolevel_plugin_init, VERSION, GST_LICENSE,
GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);

73
gst/videoframe_audiolevel/gstvideoframe-audiolevel.h Normal file
View file

@ -0,0 +1,73 @@
/*
* GStreamer
* Copyright (C) 2015 Vivia Nikolaidou <vivia@toolsonair.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_VIDEOFRAME_AUDIOLEVEL_H__
#define __GST_VIDEOFRAME_AUDIOLEVEL_H__
#include <gst/gst.h>
#include <gst/audio/audio.h>
G_BEGIN_DECLS
#define GST_TYPE_VIDEOFRAME_AUDIOLEVEL (gst_videoframe_audiolevel_get_type())
#define GST_VIDEOFRAME_AUDIOLEVEL(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_VIDEOFRAME_AUDIOLEVEL,GstVideoFrameAudioLevel))
#define GST_IS_VIDEOFRAME_AUDIOLEVEL(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_VIDEOFRAME_AUDIOLEVEL))
#define GST_VIDEOFRAME_AUDIOLEVEL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_VIDEOFRAME_AUDIOLEVEL,GstVideoFrameAudioLevelClass))
#define GST_IS_VIDEOFRAME_AUDIOLEVEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_VIDEOFRAME_AUDIOLEVEL))
#define GST_VIDEOFRAME_AUDIOLEVEL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_VIDEOFRAME_AUDIOLEVEL,GstVideoFrameAudioLevelClass))
typedef struct _GstVideoFrameAudioLevel GstVideoFrameAudioLevel;
typedef struct _GstVideoFrameAudioLevelClass GstVideoFrameAudioLevelClass;
struct _GstVideoFrameAudioLevel
{
GstElement parent;
GstPad *asrcpad, *asinkpad, *vsrcpad, *vsinkpad;
GstAudioInfo ainfo;
gdouble *CS; /* normalized Cumulative Square */
GstSegment asegment, vsegment;
void (*process) (gpointer, guint, guint, gdouble *);
GQueue vtimeq;
GstAdapter *adapter;
GstClockTime first_time;
guint total_frames;
guint64 next_offset, alignment_threshold, discont_time, discont_wait;
gboolean video_eos_flag;
gboolean audio_flush_flag;
gboolean shutdown_flag;
GCond cond;
GMutex mutex;
};
struct _GstVideoFrameAudioLevelClass
{
GstElementClass parent_class;
};
GType gst_videoframe_audiolevel_get_type (void);
G_END_DECLS
#endif /* __GST_VIDEOFRAME_AUDIOLEVEL_H__ */

8
tests/check/Makefile.am
View file

@ -247,6 +247,7 @@ check_PROGRAMS = \
$(check_curl) \
$(check_shm) \
elements/aiffparse \
elements/videoframe-audiolevel \
elements/autoconvert \
elements/autovideoconvert \
elements/audiointerleave \
@ -373,6 +374,13 @@ libs_vp8parser_LDADD = \
$(GST_PLUGINS_BAD_LIBS) -lgstcodecparsers-@GST_API_VERSION@ \
$(GST_BASE_LIBS) $(GST_LIBS) $(LDADD)
elements_videoframe_audiolevel_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS)
elements_videoframe_audiolevel_LDADD = \
$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \
-lgstaudio-@GST_API_VERSION@
elements_faad_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS)

636
tests/check/elements/videoframe-audiolevel.c Normal file
View file

@ -0,0 +1,636 @@
/* GStreamer unit test for videoframe-audiolevel
*
* Copyright (C) 2015 Vivia Nikolaidou <vivia@toolsonair.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
/* suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.h>
static gboolean got_eos;
static guint audio_buffer_count, video_buffer_count;
static GstSegment current_audio_segment, current_video_segment;
static guint num_msgs;
static GQueue v_timestamp_q, msg_timestamp_q;
static guint n_abuffers, n_vbuffers;
static guint channels, fill_value;
static gdouble expected_rms;
static gboolean audiodelay, videodelay, per_channel, long_video;
static gboolean early_video, late_video;
static gboolean video_gaps, video_overlaps;
static gboolean audio_nondiscont, audio_drift;
static guint fill_value_per_channel[] = { 0, 1 };
static gdouble expected_rms_per_channel[] = { 0, 0.0078125 };
static void
set_default_params (void)
{
n_abuffers = 40;
n_vbuffers = 15;
channels = 2;
expected_rms = 0.0078125;
fill_value = 1;
audiodelay = FALSE;
videodelay = FALSE;
per_channel = FALSE;
long_video = FALSE;
video_gaps = FALSE;
video_overlaps = FALSE;
audio_nondiscont = FALSE;
audio_drift = FALSE;
early_video = FALSE;
late_video = FALSE;
};
static GstFlowReturn
output_achain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstClockTime timestamp;
guint8 b;
gboolean audio_jitter = audio_nondiscont || audio_drift || early_video;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp,
(audio_buffer_count % n_abuffers) * 1 * GST_SECOND);
timestamp =
gst_segment_to_stream_time (&current_audio_segment, GST_FORMAT_TIME,
timestamp);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp,
(audio_buffer_count % n_abuffers) * 1 * GST_SECOND);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
timestamp =
gst_segment_to_running_time (&current_audio_segment, GST_FORMAT_TIME,
timestamp);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp, audio_buffer_count * 1 * GST_SECOND);
gst_buffer_extract (buffer, 0, &b, 1);
if (per_channel) {
fail_unless_equals_int (b, fill_value_per_channel[0]);
} else {
fail_unless_equals_int (b, fill_value);
}
audio_buffer_count++;
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static gboolean
output_aevent (GstPad * pad, GstObject * parent, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&current_audio_segment, GST_FORMAT_UNDEFINED);
break;
case GST_EVENT_SEGMENT:
gst_event_copy_segment (event, &current_audio_segment);
break;
case GST_EVENT_EOS:
got_eos = TRUE;
break;
default:
break;
}
gst_event_unref (event);
return TRUE;
}
static GstFlowReturn
output_vchain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstClockTime timestamp;
guint8 b;
gboolean jitter = video_gaps || video_overlaps || late_video;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (!jitter)
fail_unless_equals_int64 (timestamp,
(video_buffer_count % n_vbuffers) * 25 * GST_MSECOND);
timestamp =
gst_segment_to_stream_time (&current_video_segment, GST_FORMAT_TIME,
timestamp);
if (!jitter)
fail_unless_equals_int64 (timestamp,
(video_buffer_count % n_vbuffers) * 25 * GST_MSECOND);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
timestamp =
gst_segment_to_running_time (&current_video_segment, GST_FORMAT_TIME,
timestamp);
if (!jitter)
fail_unless_equals_int64 (timestamp, video_buffer_count * 25 * GST_MSECOND);
gst_buffer_extract (buffer, 0, &b, 1);
if (!jitter)
fail_unless_equals_int (b, video_buffer_count % n_vbuffers);
video_buffer_count++;
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static gboolean
output_vevent (GstPad * pad, GstObject * parent, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&current_video_segment, GST_FORMAT_UNDEFINED);
break;
case GST_EVENT_SEGMENT:
gst_event_copy_segment (event, &current_video_segment);
break;
case GST_EVENT_EOS:
got_eos = TRUE;
break;
default:
break;
}
gst_event_unref (event);
return TRUE;
}
static gpointer
push_abuffers (gpointer data)
{
GstSegment segment;
GstPad *pad = data;
gint i, j, k;
GstClockTime timestamp = 0;
GstAudioInfo info;
GstCaps *caps;
guint buf_size = 1000;
if (audiodelay)
g_usleep (2000);
if (early_video)
timestamp = 50 * GST_MSECOND;
gst_pad_send_event (pad, gst_event_new_stream_start ("test"));
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S8, buf_size, channels,
NULL);
caps = gst_audio_info_to_caps (&info);
gst_pad_send_event (pad, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_send_event (pad, gst_event_new_segment (&segment));
for (i = 0; i < n_abuffers; i++) {
GstBuffer *buf = gst_buffer_new_and_alloc (channels * buf_size);
if (per_channel) {
GstMapInfo map;
guint8 *in_data;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
in_data = map.data;
for (j = 0; j < buf_size; j++) {
for (k = 0; k < channels; k++) {
in_data[j * channels + k] = fill_value_per_channel[k];
}
}
gst_buffer_unmap (buf, &map);
} else {
gst_buffer_memset (buf, 0, fill_value, channels * buf_size);
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
timestamp += 1 * GST_SECOND;
if (audio_drift)
timestamp += 50 * GST_MSECOND;
else if (i == 4 && audio_nondiscont)
timestamp += 30 * GST_MSECOND;
GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf);
fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK);
}
gst_pad_send_event (pad, gst_event_new_eos ());
return NULL;
}
static gpointer
push_vbuffers (gpointer data)
{
GstSegment segment;
GstPad *pad = data;
gint i;
GstClockTime timestamp = 0;
if (videodelay)
g_usleep (2000);
if (late_video)
timestamp = 50 * GST_MSECOND;
gst_pad_send_event (pad, gst_event_new_stream_start ("test"));
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_send_event (pad, gst_event_new_segment (&segment));
for (i = 0; i < n_vbuffers; i++) {
GstBuffer *buf = gst_buffer_new_and_alloc (1000);
GstClockTime *rtime = g_new (GstClockTime, 1);
gst_buffer_memset (buf, 0, i, 1);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
timestamp += 25 * GST_MSECOND;
GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf);
*rtime = gst_segment_to_running_time (&segment, GST_FORMAT_TIME, timestamp);
g_queue_push_tail (&v_timestamp_q, rtime);
if (i == 4) {
if (video_gaps)
timestamp += 10 * GST_MSECOND;
else if (video_overlaps)
timestamp -= 10 * GST_MSECOND;
}
fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK);
}
gst_pad_send_event (pad, gst_event_new_eos ());
return NULL;
}
static GstBusSyncReply
on_message (GstBus * bus, GstMessage * message, gpointer user_data)
{
const GstStructure *s = gst_message_get_structure (message);
const gchar *name = gst_structure_get_name (s);
GValueArray *rms_arr;
const GValue *array_val;
const GValue *value;
gdouble rms;
gint channels2;
guint i;
GstClockTime *rtime = g_new (GstClockTime, 1);
if (message->type != GST_MESSAGE_ELEMENT
|| strcmp (name, "videoframe-audiolevel") != 0)
goto done;
num_msgs++;
if (!gst_structure_get_clock_time (s, "running-time", rtime))
g_warning ("Could not parse running time");
else
g_queue_push_tail (&msg_timestamp_q, rtime);
/* the values are packed into GValueArrays with the value per channel */
array_val = gst_structure_get_value (s, "rms");
rms_arr = (GValueArray *) g_value_get_boxed (array_val);
channels2 = rms_arr->n_values;
fail_unless_equals_int (channels2, channels);
for (i = 0; i < channels; ++i) {
value = g_value_array_get_nth (rms_arr, i);
rms = g_value_get_double (value);
if (per_channel) {
fail_unless_equals_float (rms, expected_rms_per_channel[i]);
} else if (early_video && *rtime <= 50 * GST_MSECOND) {
fail_unless_equals_float (rms, 0);
} else {
fail_unless_equals_float (rms, expected_rms);
}
}
done:
return GST_BUS_PASS;
}
static void
test_videoframe_audiolevel_generic (void)
{
GstElement *alevel;
GstPad *asink, *vsink, *asrc, *vsrc, *aoutput_sink, *voutput_sink;
GThread *athread, *vthread;
GstBus *bus;
guint i;
got_eos = FALSE;
audio_buffer_count = 0;
video_buffer_count = 0;
num_msgs = 0;
g_queue_init (&v_timestamp_q);
g_queue_init (&msg_timestamp_q);
alevel = gst_element_factory_make ("videoframe-audiolevel", NULL);
fail_unless (alevel != NULL);
bus = gst_bus_new ();
gst_element_set_bus (alevel, bus);
gst_bus_set_sync_handler (bus, on_message, NULL, NULL);
asink = gst_element_get_static_pad (alevel, "asink");
fail_unless (asink != NULL);
vsink = gst_element_get_static_pad (alevel, "vsink");
fail_unless (vsink != NULL);
asrc = gst_element_get_static_pad (alevel, "asrc");
aoutput_sink = gst_pad_new ("sink", GST_PAD_SINK);
fail_unless (aoutput_sink != NULL);
fail_unless (gst_pad_link (asrc, aoutput_sink) == GST_PAD_LINK_OK);
vsrc = gst_element_get_static_pad (alevel, "vsrc");
voutput_sink = gst_pad_new ("sink", GST_PAD_SINK);
fail_unless (voutput_sink != NULL);
fail_unless (gst_pad_link (vsrc, voutput_sink) == GST_PAD_LINK_OK);
gst_pad_set_chain_function (aoutput_sink, output_achain);
gst_pad_set_event_function (aoutput_sink, output_aevent);
gst_pad_set_chain_function (voutput_sink, output_vchain);
gst_pad_set_event_function (voutput_sink, output_vevent);
gst_pad_set_active (aoutput_sink, TRUE);
gst_pad_set_active (voutput_sink, TRUE);
fail_unless (gst_element_set_state (alevel,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
athread = g_thread_new ("athread", (GThreadFunc) push_abuffers, asink);
vthread = g_thread_new ("vthread", (GThreadFunc) push_vbuffers, vsink);
g_thread_join (vthread);
g_thread_join (athread);
fail_unless (got_eos);
fail_unless_equals_int (audio_buffer_count, n_abuffers);
fail_unless_equals_int (video_buffer_count, n_vbuffers);
if (!long_video)
fail_unless_equals_int (num_msgs, n_vbuffers);
fail_unless_equals_int (g_queue_get_length (&v_timestamp_q), n_vbuffers);
/* num_msgs is equal to n_vbuffers except in the case of long_video */
fail_unless_equals_int (g_queue_get_length (&msg_timestamp_q), num_msgs);
for (i = 0; i < g_queue_get_length (&msg_timestamp_q); i++) {
GstClockTime *vt = g_queue_pop_head (&v_timestamp_q);
GstClockTime *mt = g_queue_pop_head (&msg_timestamp_q);
fail_unless (vt != NULL);
fail_unless (mt != NULL);
if (!video_gaps && !video_overlaps && !early_video)
fail_unless_equals_uint64 (*vt, *mt);
g_free (vt);
g_free (mt);
}
/* teardown */
gst_element_set_state (alevel, GST_STATE_NULL);
gst_bus_set_flushing (bus, TRUE);
gst_object_unref (bus);
g_queue_foreach (&v_timestamp_q, (GFunc) g_free, NULL);
g_queue_foreach (&msg_timestamp_q, (GFunc) g_free, NULL);
g_queue_clear (&v_timestamp_q);
g_queue_clear (&msg_timestamp_q);
gst_pad_unlink (asrc, aoutput_sink);
gst_object_unref (asrc);
gst_pad_unlink (vsrc, voutput_sink);
gst_object_unref (vsrc);
gst_object_unref (asink);
gst_object_unref (vsink);
gst_pad_set_active (aoutput_sink, FALSE);
gst_object_unref (aoutput_sink);
gst_pad_set_active (voutput_sink, FALSE);
gst_object_unref (voutput_sink);
gst_object_unref (alevel);
}
GST_START_TEST (test_videoframe_audiolevel_16chan_1)
{
set_default_params ();
channels = 16;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_8chan_1)
{
set_default_params ();
channels = 8;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_2chan_1)
{
set_default_params ();
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_1chan_1)
{
set_default_params ();
channels = 1;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_16chan_0)
{
set_default_params ();
channels = 16;
expected_rms = 0;
fill_value = 0;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_8chan_0)
{
set_default_params ();
channels = 8;
expected_rms = 0;
fill_value = 0;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_2chan_0)
{
set_default_params ();
channels = 2;
expected_rms = 0;
fill_value = 0;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_1chan_0)
{
set_default_params ();
channels = 1;
expected_rms = 0;
fill_value = 0;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_adelay)
{
set_default_params ();
audiodelay = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_vdelay)
{
set_default_params ();
videodelay = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_per_channel)
{
set_default_params ();
per_channel = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_long_video)
{
set_default_params ();
n_abuffers = 6;
n_vbuffers = 255;
long_video = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_video_gaps)
{
set_default_params ();
video_gaps = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_video_overlaps)
{
set_default_params ();
video_overlaps = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_audio_nondiscont)
{
set_default_params ();
audio_nondiscont = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_audio_drift)
{
set_default_params ();
audio_drift = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_early_video)
{
set_default_params ();
early_video = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_late_video)
{
set_default_params ();
late_video = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
static Suite *
videoframe_audiolevel_suite (void)
{
Suite *s = suite_create ("videoframe-audiolevel");
TCase *tc_chain;
tc_chain = tcase_create ("videoframe-audiolevel");
tcase_add_test (tc_chain, test_videoframe_audiolevel_16chan_1);
tcase_add_test (tc_chain, test_videoframe_audiolevel_8chan_1);
tcase_add_test (tc_chain, test_videoframe_audiolevel_2chan_1);
tcase_add_test (tc_chain, test_videoframe_audiolevel_1chan_1);
tcase_add_test (tc_chain, test_videoframe_audiolevel_16chan_0);
tcase_add_test (tc_chain, test_videoframe_audiolevel_8chan_0);
tcase_add_test (tc_chain, test_videoframe_audiolevel_2chan_0);
tcase_add_test (tc_chain, test_videoframe_audiolevel_1chan_0);
tcase_add_test (tc_chain, test_videoframe_audiolevel_adelay);
tcase_add_test (tc_chain, test_videoframe_audiolevel_vdelay);
tcase_add_test (tc_chain, test_videoframe_audiolevel_per_channel);
tcase_add_test (tc_chain, test_videoframe_audiolevel_long_video);
tcase_add_test (tc_chain, test_videoframe_audiolevel_video_gaps);
tcase_add_test (tc_chain, test_videoframe_audiolevel_video_overlaps);
tcase_add_test (tc_chain, test_videoframe_audiolevel_audio_nondiscont);
tcase_add_test (tc_chain, test_videoframe_audiolevel_audio_drift);
tcase_add_test (tc_chain, test_videoframe_audiolevel_early_video);
tcase_add_test (tc_chain, test_videoframe_audiolevel_late_video);
suite_add_tcase (s, tc_chain);
return s;
}
GST_CHECK_MAIN (videoframe_audiolevel);