| /third_party/gstreamer/gstplugins_bad/gst/inter/ |
| D | gstinteraudiosrc.c | 164 interaudiosrc->latency_time = DEFAULT_AUDIO_LATENCY_TIME; in gst_inter_audio_src_init()
183 interaudiosrc->latency_time = g_value_get_uint64 (value); in gst_inter_audio_src_set_property()
208 g_value_set_uint64 (value, interaudiosrc->latency_time); in gst_inter_audio_src_get_property()
290 interaudiosrc->surface->audio_latency_time = interaudiosrc->latency_time; in gst_inter_audio_src_start()
449 min_latency = interaudiosrc->latency_time; in gst_inter_audio_src_query()
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| D | gstinteraudiosrc.h | 48 guint64 buffer_time, latency_time, period_time; member
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| /third_party/gstreamer/gstplugins_base/gst-libs/gst/audio/ |
| D | gstaudiobasesrc.c | 219 audiobasesrc->latency_time = DEFAULT_LATENCY_TIME; in gst_audio_base_src_init()
435 src->latency_time = g_value_get_int64 (value); in gst_audio_base_src_set_property()
462 g_value_set_int64 (value, src->latency_time); in gst_audio_base_src_get_property()
475 g_value_set_int64 (value, src->ringbuffer->spec.latency_time); in gst_audio_base_src_get_property()
532 spec->latency_time = src->latency_time; in gst_audio_base_src_setcaps()
544 spec->segsize = rate * bpf * spec->latency_time / GST_MSECOND; in gst_audio_base_src_setcaps()
547 spec->segtotal = spec->buffer_time / spec->latency_time; in gst_audio_base_src_setcaps()
559 spec->latency_time = spec->segsize * GST_MSECOND / (rate * bpf); in gst_audio_base_src_setcaps()
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| D | gstaudiobasesrc.h | 103 GstClockTime latency_time; member
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| D | gstaudiobasesink.h | 181 guint64 latency_time; member
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| D | gstaudiobasesink.c | 285 audiobasesink->latency_time = DEFAULT_LATENCY_TIME; in gst_audio_base_sink_init()
838 sink->latency_time = g_value_get_int64 (value); in gst_audio_base_sink_set_property()
878 g_value_set_int64 (value, sink->latency_time); in gst_audio_base_sink_get_property()
943 spec->latency_time = sink->latency_time; in gst_audio_base_sink_setcaps()
979 spec->latency_time = gst_util_uint64_scale (spec->segsize, in gst_audio_base_sink_setcaps()
982 spec->buffer_time = spec->segtotal * spec->latency_time; in gst_audio_base_sink_setcaps()
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| D | gstaudioringbuffer.h | 138 guint64 latency_time; /* the required/actual latency time, this is the member
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| D | gstaudioringbuffer.c | 183 spec->latency_time); in gst_audio_ring_buffer_debug_spec_buff()
322 g_return_val_if_fail (spec->latency_time != 0, FALSE); in gst_audio_ring_buffer_parse_caps()
328 spec->latency_time, GST_SECOND / GST_USECOND); in gst_audio_ring_buffer_parse_caps()
332 spec->segtotal = spec->buffer_time / spec->latency_time; in gst_audio_ring_buffer_parse_caps()
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| /third_party/gstreamer/gstplugins_bad/sys/directsound/ |
| D | gstdirectsoundsrc.c | 500 G_GUINT64_FORMAT, spec->latency_time, spec->buffer_time); in gst_directsound_src_prepare()
503 if (spec->buffer_time < spec->latency_time * 2) { in gst_directsound_src_prepare()
504 spec->buffer_time = spec->latency_time * 2; in gst_directsound_src_prepare()
516 gst_util_uint64_scale (spec->latency_time, wfx.nAvgBytesPerSec, in gst_directsound_src_prepare()
532 dsoundsrc->latency_time = in gst_directsound_src_prepare()
537 spec->latency_time); in gst_directsound_src_prepare()
694 sleep_time_ms = gst_util_uint64_scale (dsoundsrc->latency_time, in gst_directsound_src_read()
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| D | gstdirectsoundsrc.h | 87 guint latency_time; member
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| /third_party/gstreamer/gstplugins_good/sys/osxaudio/ |
| D | gstosxaudioringbuffer.c | 225 (spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) / in gst_osx_audio_ring_buffer_acquire()
227 spec->segtotal = spec->buffer_time / spec->latency_time; in gst_osx_audio_ring_buffer_acquire()
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| D | gstosxaudiosink.c | 406 spec.latency_time = GST_SECOND; in gst_osx_audio_sink_acceptcaps()
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| /third_party/gstreamer/gstplugins_good/ext/jack/ |
| D | gstjackaudiosrc.c | 441 spec->latency_time = gst_util_uint64_scale (spec->segsize, in gst_jack_ring_buffer_acquire()
444 spec->segtotal = spec->buffer_time / spec->latency_time; in gst_jack_ring_buffer_acquire()
449 spec->buffer_time = spec->latency_time * spec->segtotal; in gst_jack_ring_buffer_acquire()
455 spec->latency_time); in gst_jack_ring_buffer_acquire()
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| D | gstjackaudiosink.c | 433 spec->latency_time = gst_util_uint64_scale (spec->segsize, in gst_jack_ring_buffer_acquire()
436 spec->segtotal = spec->buffer_time / spec->latency_time; in gst_jack_ring_buffer_acquire()
441 spec->buffer_time = spec->latency_time * spec->segtotal; in gst_jack_ring_buffer_acquire()
447 spec->latency_time); in gst_jack_ring_buffer_acquire()
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| /third_party/gstreamer/gstplugins_bad/sys/tinyalsa/ |
| D | tinyalsasink.c | 273 GST_FORMAT_TIME, spec->latency_time * GST_USECOND, in pcm_config_from_spec()
277 config->period_count = spec->buffer_time / spec->latency_time; in pcm_config_from_spec()
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| /third_party/gstreamer/gstplugins_bad/sys/dshowsrcwrapper/ |
| D | gstdshowaudiosrc.cpp | 543 spec->segsize = (gint) (spec->info.bpf * spec->info.rate * spec->latency_time / in gst_dshowaudiosrc_prepare()
546 (gfloat) spec->latency_time + 0.5); in gst_dshowaudiosrc_prepare()
670 Sleep (GST_AUDIO_BASE_SRC(src)->ringbuffer->spec.latency_time / in gst_dshowaudiosrc_read()
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| /third_party/gstreamer/gstplugins_bad/sys/wasapi/ |
| D | gstwasapiutil.c | 809 if (spec->latency_time * 10 > default_period) in gst_wasapi_util_get_best_buffer_sizes()
821 use_period = use_period * MAX ((spec->latency_time * 10) / use_period, 1); in gst_wasapi_util_get_best_buffer_sizes()
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| /third_party/gstreamer/gstplugins_good/sys/directsound/ |
| D | gstdirectsoundsink.c | 352 spec.latency_time = GST_SECOND; in gst_directsound_sink_acceptcaps()
506 gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time, in gst_directsound_sink_prepare()
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| /third_party/gstreamer/gstplugins_base/ext/alsa/ |
| D | gstalsasink.c | 371 spec.latency_time = GST_SECOND; in gst_alsasink_acceptcaps()
848 alsa->period_time = spec->latency_time; in alsasink_parse_spec()
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| D | gstalsasrc.c | 752 alsa->period_time = spec->latency_time; in alsasrc_parse_spec()
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| /third_party/gstreamer/gstplugins_good/ext/pulse/ |
| D | pulsesink.c | 2276 spec.latency_time = GST_AUDIO_BASE_SINK (psink)->latency_time; in gst_pulsesink_query_acceptcaps()
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| D | pulsesrc.c | 1282 new_spec.latency_time = GST_SECOND; in gst_pulsesrc_create_stream()
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| /third_party/pulseaudio/ |
| D | NEWS | 174 * Removed the "latency_time" option from module-null-source
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| /third_party/gstreamer/gstplugins_base/ |
| D | ChangeLog | 149138 …gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored …
149146 Document better the fact that latency_time and buffer_time are values
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