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audioaggregator: remove GstAudioAggregator->info

Browse source 
As we now require subclasses to use a subclass of
GstAudioAggregatorPad, we can reuse its info field

https://bugzilla.gnome.org/show_bug.cgi?id=793943
This commit is contained in:
Mathieu Duponchelle 2018-02-28 22:12:57 +01:00
parent 556bc04f1c
commit 318eb61e23
4 changed files with 52 additions and 46 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

43
gst-libs/gst/audio/gstaudioaggregator.c
View file

@ -548,7 +548,6 @@ gst_audio_aggregator_init (GstAudioAggregator * aagg)
aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT;
aagg->current_caps = NULL;
gst_audio_info_init (&aagg->info);
gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration);
@ -834,6 +833,7 @@ gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GstAudioInfo info;
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
GST_INFO_OBJECT (agg, "src caps negotiated %" GST_PTR_FORMAT, caps);
@ -849,7 +849,7 @@ gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
gst_audio_aggregator_update_converters (aagg, &info);
if (aagg->priv->current_buffer
&& !gst_audio_info_is_equal (&aagg->info, &info)) {
&& !gst_audio_info_is_equal (&srcpad->info, &info)) {
GstBuffer *converted;
GstAudioAggregatorPadClass *klass =
GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad);
@ -858,18 +858,18 @@ gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->srcpad));
converted =
gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &aagg->info,
gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &srcpad->info,
&info, aagg->priv->current_buffer);
gst_buffer_unref (aagg->priv->current_buffer);
aagg->priv->current_buffer = converted;
}
}
if (!gst_audio_info_is_equal (&info, &aagg->info)) {
if (!gst_audio_info_is_equal (&info, &srcpad->info)) {
GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps);
gst_caps_replace (&aagg->current_caps, caps);
memcpy (&aagg->info, &info, sizeof (info));
memcpy (&srcpad->info, &info, sizeof (info));
}
GST_OBJECT_UNLOCK (aagg);
@ -1148,6 +1148,7 @@ static gboolean
gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
@ -1170,9 +1171,9 @@ gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
res = TRUE;
break;
case GST_FORMAT_BYTES:
if (GST_AUDIO_INFO_BPF (&aagg->info)) {
if (GST_AUDIO_INFO_BPF (&srcpad->info)) {
gst_query_set_position (query, format, aagg->priv->offset *
GST_AUDIO_INFO_BPF (&aagg->info));
GST_AUDIO_INFO_BPF (&srcpad->info));
res = TRUE;
}
break;
@ -1228,7 +1229,7 @@ gst_audio_aggregator_reset (GstAudioAggregator * aagg)
GST_OBJECT_LOCK (aagg);
agg->segment.position = -1;
aagg->priv->offset = -1;
gst_audio_info_init (&aagg->info);
gst_audio_info_init (&GST_AUDIO_AGGREGATOR_PAD (agg->srcpad)->info);
gst_caps_replace (&aagg->current_caps, NULL);
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
GST_OBJECT_UNLOCK (aagg);
@ -1305,10 +1306,11 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) {
rate = GST_AUDIO_INFO_RATE (&aagg->info);
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
rate = GST_AUDIO_INFO_RATE (&srcpad->info);
bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
} else {
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
@ -1581,20 +1583,22 @@ gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
GstAllocationParams params;
GstBuffer *outbuf;
GstMapInfo outmap;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, &params);
GST_DEBUG ("Creating output buffer with size %d",
num_frames * GST_AUDIO_INFO_BPF (&aagg->info));
num_frames * GST_AUDIO_INFO_BPF (&srcpad->info));
outbuf = gst_buffer_new_allocate (allocator, num_frames *
GST_AUDIO_INFO_BPF (&aagg->info), &params);
GST_AUDIO_INFO_BPF (&srcpad->info), &params);
if (allocator)
gst_object_unref (allocator);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size);
gst_audio_format_fill_silence (srcpad->info.finfo, outmap.data, outmap.size);
gst_buffer_unmap (outbuf, &outmap);
return outbuf;
@ -1663,6 +1667,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
gboolean is_eos = TRUE;
gboolean is_done = TRUE;
guint blocksize;
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
element = GST_ELEMENT (agg);
aagg = GST_AUDIO_AGGREGATOR (agg);
@ -1681,7 +1686,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
agg->segment.position = agg->segment.stop;
}
if (G_UNLIKELY (aagg->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
if (G_UNLIKELY (srcpad->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
if (timeout) {
GST_DEBUG_OBJECT (aagg,
"Got timeout before receiving any caps, don't output anything");
@ -1703,8 +1708,8 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
}
}
rate = GST_AUDIO_INFO_RATE (&aagg->info);
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
rate = GST_AUDIO_INFO_RATE (&srcpad->info);
bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
if (aagg->priv->offset == -1) {
aagg->priv->offset =
@ -1764,7 +1769,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
GST_DEBUG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT
" frames (%" GST_TIME_FORMAT ")", diff,
GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
GST_AUDIO_INFO_RATE (&aagg->info))));
GST_AUDIO_INFO_RATE (&srcpad->info))));
}
} else if (!pad_eos) {
is_done = FALSE;
@ -1778,7 +1783,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
pad->priv->buffer =
gst_audio_aggregator_convert_buffer
(aagg, GST_PAD (pad), &pad->info, &aagg->info,
(aagg, GST_PAD (pad), &pad->info, &srcpad->info,
pad->priv->input_buffer);
else
pad->priv->buffer = gst_buffer_ref (pad->priv->input_buffer);
@ -1820,7 +1825,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
GST_DEBUG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
", dropping %" GST_PTR_FORMAT,
GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer);
GST_AUDIO_INFO_RATE (&srcpad->info))), pad->priv->buffer);
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
gst_buffer_replace (&pad->priv->input_buffer, NULL);

3
gst-libs/gst/audio/gstaudioaggregator.h
View file

@ -172,9 +172,6 @@ struct _GstAudioAggregator
{
GstAggregator parent;
/* All member are read only for subclasses, must hold OBJECT lock */
GstAudioInfo info;
GstCaps *current_caps;
/*< private >*/

12
gst/audiomixer/gstaudiointerleave.c
View file

@ -535,12 +535,12 @@ static gboolean
gst_audio_interleave_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
{
GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg);
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
if (!GST_AGGREGATOR_CLASS (parent_class)->negotiated_src_caps (agg, caps))
return FALSE;
gst_audio_interleave_set_process_function (self, &aagg->info);
gst_audio_interleave_set_process_function (self, &srcpad->info);
return TRUE;
}
@ -818,14 +818,16 @@ gst_audio_interleave_aggregate_one_buffer (GstAudioAggregator * aagg,
GstMapInfo outmap;
gint out_width, in_bpf, out_bpf, out_channels, channel;
guint8 *outdata;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
GST_OBJECT_LOCK (aagg);
GST_OBJECT_LOCK (aaggpad);
out_width = GST_AUDIO_INFO_WIDTH (&aagg->info) / 8;
out_width = GST_AUDIO_INFO_WIDTH (&srcpad->info) / 8;
in_bpf = GST_AUDIO_INFO_BPF (&aaggpad->info);
out_bpf = GST_AUDIO_INFO_BPF (&aagg->info);
out_channels = GST_AUDIO_INFO_CHANNELS (&aagg->info);
out_bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
out_channels = GST_AUDIO_INFO_CHANNELS (&srcpad->info);
gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);

40
gst/audiomixer/gstaudiomixer.c
View file

@ -295,6 +295,8 @@ gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg,
GstMapInfo inmap;
GstMapInfo outmap;
gint bpf;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
GST_OBJECT_LOCK (aagg);
GST_OBJECT_LOCK (aaggpad);
@ -306,7 +308,7 @@ gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg,
return FALSE;
}
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
@ -315,92 +317,92 @@ gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg,
/* further buffers, need to add them */
if (pad->volume == 1.0) {
switch (aagg->info.finfo->format) {
switch (srcpad->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
audiomixer_orc_add_u8 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_S8:
audiomixer_orc_add_s8 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_U16:
audiomixer_orc_add_u16 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_S16:
audiomixer_orc_add_s16 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_U32:
audiomixer_orc_add_u32 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_S32:
audiomixer_orc_add_s32 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_F32:
audiomixer_orc_add_f32 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_F64:
audiomixer_orc_add_f64 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
num_frames * srcpad->info.channels);
break;
default:
g_assert_not_reached ();
break;
}
} else {
switch (aagg->info.finfo->format) {
switch (srcpad->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
audiomixer_orc_add_volume_u8 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i8, num_frames * aagg->info.channels);
pad->volume_i8, num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_S8:
audiomixer_orc_add_volume_s8 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i8, num_frames * aagg->info.channels);
pad->volume_i8, num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_U16:
audiomixer_orc_add_volume_u16 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i16, num_frames * aagg->info.channels);
pad->volume_i16, num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_S16:
audiomixer_orc_add_volume_s16 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i16, num_frames * aagg->info.channels);
pad->volume_i16, num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_U32:
audiomixer_orc_add_volume_u32 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i32, num_frames * aagg->info.channels);
pad->volume_i32, num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_S32:
audiomixer_orc_add_volume_s32 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i32, num_frames * aagg->info.channels);
pad->volume_i32, num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_F32:
audiomixer_orc_add_volume_f32 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume, num_frames * aagg->info.channels);
pad->volume, num_frames * srcpad->info.channels);
break;
case GST_AUDIO_FORMAT_F64:
audiomixer_orc_add_volume_f64 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume, num_frames * aagg->info.channels);
pad->volume, num_frames * srcpad->info.channels);
break;
default:
g_assert_not_reached ();