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audiomixer: Change blocksize property to output-buffer-duration in time format

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This makes the interface of audiomixer independent of the actual caps.
This commit is contained in:
Sebastian Dröge 2014-12-23 11:45:50 +01:00
parent 6771db209d
commit 7c575af6df
3 changed files with 36 additions and 24 deletions
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54
gst/audiomixer/gstaudiomixer.c
View file

@ -174,7 +174,7 @@ gst_audiomixer_pad_init (GstAudioMixerPad * pad)
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND) #define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND) #define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
#define DEFAULT_BLOCKSIZE (1024) #define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND)
enum enum
{ {
@ -182,7 +182,7 @@ enum
PROP_FILTER_CAPS, PROP_FILTER_CAPS,
PROP_ALIGNMENT_THRESHOLD, PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT, PROP_DISCONT_WAIT,
PROP_BLOCKSIZE PROP_OUTPUT_BUFFER_DURATION
}; };
/* elementfactory information */ /* elementfactory information */
@ -792,10 +792,10 @@ gst_audiomixer_class_init (GstAudioMixerClass * klass)
G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT, G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BLOCKSIZE, g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION,
g_param_spec_uint ("blocksize", "Block Size", g_param_spec_uint64 ("output-buffer-duration", "Output Buffer Duration",
"Output block size in number of samples", 1, "Output block size in number of samples", 1,
G_MAXUINT, DEFAULT_BLOCKSIZE, G_MAXUINT64, DEFAULT_OUTPUT_BUFFER_DURATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class, gst_element_class_add_pad_template (gstelement_class,
@ -841,7 +841,9 @@ gst_audiomixer_init (GstAudioMixer * audiomixer)
audiomixer->filter_caps = NULL; audiomixer->filter_caps = NULL;
audiomixer->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD; audiomixer->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
audiomixer->discont_wait = DEFAULT_DISCONT_WAIT; audiomixer->discont_wait = DEFAULT_DISCONT_WAIT;
audiomixer->blocksize = DEFAULT_BLOCKSIZE; audiomixer->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION;
gst_aggregator_set_latency (GST_AGGREGATOR (audiomixer),
audiomixer->output_buffer_duration, GST_CLOCK_TIME_NONE);
} }
static void static void
@ -889,8 +891,10 @@ gst_audiomixer_set_property (GObject * object, guint prop_id,
case PROP_DISCONT_WAIT: case PROP_DISCONT_WAIT:
audiomixer->discont_wait = g_value_get_uint64 (value); audiomixer->discont_wait = g_value_get_uint64 (value);
break; break;
case PROP_BLOCKSIZE: case PROP_OUTPUT_BUFFER_DURATION:
audiomixer->blocksize = g_value_get_uint (value); audiomixer->output_buffer_duration = g_value_get_uint64 (value);
gst_aggregator_set_latency (GST_AGGREGATOR (audiomixer),
audiomixer->output_buffer_duration, GST_CLOCK_TIME_NONE);
break; break;
default: default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -916,8 +920,8 @@ gst_audiomixer_get_property (GObject * object, guint prop_id, GValue * value,
case PROP_DISCONT_WAIT: case PROP_DISCONT_WAIT:
g_value_set_uint64 (value, audiomixer->discont_wait); g_value_set_uint64 (value, audiomixer->discont_wait);
break; break;
case PROP_BLOCKSIZE: case PROP_OUTPUT_BUFFER_DURATION:
g_value_set_uint (value, audiomixer->blocksize); g_value_set_uint64 (value, audiomixer->output_buffer_duration);
break; break;
default: default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -1149,9 +1153,15 @@ gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
GstBuffer *inbuf; GstBuffer *inbuf;
GstMapInfo inmap; GstMapInfo inmap;
gint bpf; gint bpf;
guint blocksize;
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad); GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
blocksize =
gst_util_uint64_scale (audiomixer->output_buffer_duration,
GST_AUDIO_INFO_RATE (&audiomixer->info), GST_SECOND);
blocksize = MAX (1, blocksize);
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info); bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
/* Overlap => mix */ /* Overlap => mix */
@ -1161,8 +1171,8 @@ gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
out_start = 0; out_start = 0;
overlap = pad->size / bpf - pad->position / bpf; overlap = pad->size / bpf - pad->position / bpf;
if (overlap > audiomixer->blocksize - out_start) if (overlap > blocksize - out_start)
overlap = audiomixer->blocksize - out_start; overlap = blocksize - out_start;
inbuf = gst_aggregator_pad_get_buffer (aggpad); inbuf = gst_aggregator_pad_get_buffer (aggpad);
if (inbuf == NULL) if (inbuf == NULL)
@ -1360,6 +1370,7 @@ gst_audiomixer_aggregate (GstAggregator * agg, gboolean timeout)
gboolean dropped = FALSE; gboolean dropped = FALSE;
gboolean is_eos = TRUE; gboolean is_eos = TRUE;
gboolean is_done = TRUE; gboolean is_done = TRUE;
guint blocksize;
audiomixer = GST_AUDIO_MIXER (agg); audiomixer = GST_AUDIO_MIXER (agg);
@ -1367,11 +1378,13 @@ gst_audiomixer_aggregate (GstAggregator * agg, gboolean timeout)
if (G_UNLIKELY (audiomixer->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) if (G_UNLIKELY (audiomixer->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN))
goto not_negotiated; goto not_negotiated;
blocksize =
gst_util_uint64_scale (audiomixer->output_buffer_duration,
GST_AUDIO_INFO_RATE (&audiomixer->info), GST_SECOND);
blocksize = MAX (1, blocksize);
if (audiomixer->send_caps) { if (audiomixer->send_caps) {
gst_aggregator_set_src_caps (agg, audiomixer->current_caps); gst_aggregator_set_src_caps (agg, audiomixer->current_caps);
gst_aggregator_set_latency (agg,
gst_util_uint64_scale (audiomixer->blocksize, GST_SECOND,
GST_AUDIO_INFO_RATE (&audiomixer->info)), GST_CLOCK_TIME_NONE);
if (agg->segment.rate > 0.0) if (agg->segment.rate > 0.0)
agg->segment.position = agg->segment.start; agg->segment.position = agg->segment.start;
@ -1392,16 +1405,16 @@ gst_audiomixer_aggregate (GstAggregator * agg, gboolean timeout)
/* FIXME: Reverse mixing does not work at all yet */ /* FIXME: Reverse mixing does not work at all yet */
if (agg->segment.rate > 0.0) { if (agg->segment.rate > 0.0) {
next_offset = audiomixer->offset + audiomixer->blocksize; next_offset = audiomixer->offset + blocksize;
} else { } else {
next_offset = audiomixer->offset - audiomixer->blocksize; next_offset = audiomixer->offset - blocksize;
} }
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate); next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
if (audiomixer->current_buffer) { if (audiomixer->current_buffer) {
outbuf = audiomixer->current_buffer; outbuf = audiomixer->current_buffer;
} else { } else {
outbuf = gst_buffer_new_and_alloc (audiomixer->blocksize * bpf); outbuf = gst_buffer_new_and_alloc (blocksize * bpf);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE); gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
gst_audio_format_fill_silence (audiomixer->info.finfo, outmap.data, gst_audio_format_fill_silence (audiomixer->info.finfo, outmap.data,
outmap.size); outmap.size);
@ -1411,7 +1424,7 @@ gst_audiomixer_aggregate (GstAggregator * agg, gboolean timeout)
GST_LOG_OBJECT (agg, GST_LOG_OBJECT (agg,
"Starting to mix %u samples for offset %" G_GUINT64_FORMAT "Starting to mix %u samples for offset %" G_GUINT64_FORMAT
" with timestamp %" GST_TIME_FORMAT, audiomixer->blocksize, " with timestamp %" GST_TIME_FORMAT, blocksize,
audiomixer->offset, GST_TIME_ARGS (agg->segment.position)); audiomixer->offset, GST_TIME_ARGS (agg->segment.position));
gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE); gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
@ -1476,8 +1489,7 @@ gst_audiomixer_aggregate (GstAggregator * agg, gboolean timeout)
} }
if (pad->output_offset >= audiomixer->offset if (pad->output_offset >= audiomixer->offset
&& pad->output_offset < && pad->output_offset < audiomixer->offset + blocksize && pad->buffer) {
audiomixer->offset + audiomixer->blocksize && pad->buffer) {
GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset"); GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset");
gst_audio_mixer_mix_buffer (audiomixer, pad, &outmap); gst_audio_mixer_mix_buffer (audiomixer, pad, &outmap);

4
gst/audiomixer/gstaudiomixer.h
View file

@ -71,8 +71,8 @@ struct _GstAudioMixer {
gint64 base_time; gint64 base_time;
/* Size in samples that is output per buffer */ /* Duration of every output buffer */
guint blocksize; GstClockTime output_buffer_duration;
}; };
struct _GstAudioMixerClass { struct _GstAudioMixerClass {

2
tests/check/elements/audiomixer.c
View file

@ -1373,7 +1373,7 @@ run_sync_test (SendBuffersFunction send_buffers,
queue1 = gst_element_factory_make ("queue", "queue1"); queue1 = gst_element_factory_make ("queue", "queue1");
queue2 = gst_element_factory_make ("queue", "queue2"); queue2 = gst_element_factory_make ("queue", "queue2");
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
g_object_set (audiomixer, "blocksize", 500, NULL); g_object_set (audiomixer, "output-buffer-duration", 500 * GST_MSECOND, NULL);
sink = gst_element_factory_make ("fakesink", "sink"); sink = gst_element_factory_make ("fakesink", "sink");
g_object_set (sink, "signal-handoffs", TRUE, NULL); g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb, g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb,