gst/volume/gstvolume.c: memset buffers to zero if we get a GAP buffer. We usually see a buffer as one unit so let's h...

Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_base_init), (gst_volume_class_init),
(volume_process_double), (volume_process_float),
(volume_transform_ip), (plugin_init):
memset buffers to zero if we get a GAP buffer. We usually see a
buffer as one unit so let's handle it as one and don't care about
volume changes while processing one buffer.
Also clean up some stuff a bit.
This commit is contained in:
Sebastian Dröge 2008-03-21 16:46:33 +00:00
parent 88136fc11a
commit 2034387d4d
2 changed files with 47 additions and 27 deletions

View file

@ -1,3 +1,14 @@
2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_base_init), (gst_volume_class_init),
(volume_process_double), (volume_process_float),
(volume_transform_ip), (plugin_init):
memset buffers to zero if we get a GAP buffer. We usually see a
buffer as one unit so let's handle it as one and don't care about
volume changes while processing one buffer.
Also clean up some stuff a bit.
2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),

View file

@ -84,11 +84,6 @@
#define GST_CAT_DEFAULT gst_volume_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails volume_details = GST_ELEMENT_DETAILS ("Volume",
"Filter/Effect/Audio",
"Set volume on audio/raw streams",
"Andy Wingo <wingo@pobox.com>");
/* Filter signals and args */
enum
{
@ -293,7 +288,7 @@ volume_update_real_volume (GstVolume * this)
static gboolean
gst_volume_interface_supported (GstImplementsInterface * iface, GType type)
{
g_assert (type == GST_TYPE_MIXER);
g_return_val_if_fail (type == GST_TYPE_MIXER, FALSE);
return TRUE;
}
@ -391,7 +386,9 @@ gst_volume_base_init (gpointer g_class)
GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details (element_class, &volume_details);
gst_element_class_set_details_simple (element_class, "Volume",
"Filter/Effect/Audio",
"Set volume on audio/raw streams", "Andy Wingo <wingo@pobox.com>");
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (filter_class, caps);
@ -422,8 +419,6 @@ gst_volume_class_init (GstVolumeClass * klass)
0.0, VOLUME_MAX_DOUBLE, 1.0,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "volume", 0, "Volume gain");
trans_class->transform_ip = GST_DEBUG_FUNCPTR (volume_transform_ip);
filter_class->setup = GST_DEBUG_FUNCPTR (volume_setup);
}
@ -455,11 +450,6 @@ gst_volume_init (GstVolume * this, GstVolumeClass * g_class)
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (this), TRUE);
}
/* NOTE: although it might be tempting to have volume_process_mute() which uses
* memset(bytes, 0, nbytes) for the vol=0 case, this has the downside that
* unmuting would only take place after processing a buffer.
*/
static void
volume_process_double (GstVolume * this, gpointer bytes, guint n_bytes)
{
@ -471,6 +461,8 @@ volume_process_double (GstVolume * this, gpointer bytes, guint n_bytes)
oil_scalarmultiply_f64_ns (data, data, &vol, num_samples);
#else
gint i;
for (i = 0; i < num_samples; i++) {
*data++ *= this->real_vol_f;
}
@ -483,15 +475,16 @@ volume_process_float (GstVolume * this, gpointer bytes, guint n_bytes)
gfloat *data = (gfloat *) bytes;
guint num_samples = n_bytes / sizeof (gfloat);
/*
guint i;
for (i = 0; i < num_samples; i++) {
*data++ *= this->real_vol_f;
}
*/
/* time gst-launch 2>/dev/null audiotestsrc wave=7 num-buffers=10000 ! audio/x-raw-float ! volume volume=1.5 ! fakesink
* goes from 0m0.850s -> 0m0.717s with liboil
#if 0
guint i;
for (i = 0; i < num_samples; i++) {
*data++ *= this->real_vol_f;
}
/* time "gst-launch 2>/dev/null audiotestsrc wave=7 num-buffers=10000 ! audio/x-raw-float !
* volume volume=1.5 ! fakesink" goes from 0m0.850s -> 0m0.717s with liboil
*/
#endif
oil_scalarmultiply_f32_ns (data, data, &this->real_vol_f, num_samples);
}
@ -530,10 +523,22 @@ volume_process_int32_clamp (GstVolume * this, gpointer bytes, guint n_bytes)
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
#define get_unaligned_i24(_x) ( (((guint8*)_x)[0]) | ((((guint8*)_x)[1]) << 8) | ((((gint8*)_x)[2]) << 16) )
#define write_unaligned_u24(_x,samp) do { *(_x)++ = samp & 0xFF; *(_x)++ = (samp >> 8) & 0xFF; *(_x)++ = (samp >> 16) & 0xFF; } while (0)
#define write_unaligned_u24(_x,samp) \
G_STMT_START { \
*(_x)++ = samp & 0xFF; \
*(_x)++ = (samp >> 8) & 0xFF; \
*(_x)++ = (samp >> 16) & 0xFF; \
} G_STMT_END
#else /* BIG ENDIAN */
#define get_unaligned_i24(_x) ( (((guint8*)_x)[2]) | ((((guint8*)_x)[1]) << 8) | ((((gint8*)_x)[0]) << 16) )
#define write_unaligned_u24(_x,samp) do { *(_x)++ = (samp >> 16) & 0xFF; *(_x)++ = (samp >> 8) & 0xFF; *(_x)++ = samp & 0xFF; } while (0)
#define write_unaligned_u24(_x,samp) \
G_STMT_START { \
*(_x)++ = (samp >> 16) & 0xFF; \
*(_x)++ = (samp >> 8) & 0xFF; \
*(_x)++ = samp & 0xFF; \
} G_STMT_END
#endif
static void
@ -709,10 +714,12 @@ volume_transform_ip (GstBaseTransform * base, GstBuffer * outbuf)
GST_BUFFER_FLAG_IS_SET (outbuf, GST_BUFFER_FLAG_GAP))
return GST_FLOW_OK;
if (this->real_vol_f == 0.0)
if (this->real_vol_f == 0.0) {
this->silent_buffer = TRUE;
this->process (this, GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
memset (GST_BUFFER_DATA (outbuf), 0, GST_BUFFER_SIZE (outbuf));
} else if (this->real_vol_f != 1.0) {
this->process (this, GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
}
if (this->silent_buffer)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
@ -799,6 +806,8 @@ plugin_init (GstPlugin * plugin)
/* initialize gst controller library */
gst_controller_init (NULL, NULL);
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "volume", 0, "Volume gain");
return gst_element_register (plugin, "volume", GST_RANK_NONE,
GST_TYPE_VOLUME);
}