| /third_party/gstreamer/gstplugins_base/tests/check/libs/ |
| D | audio.c | 74 ASSERT_CRITICAL (gst_audio_buffer_clip (buf, &s, 100, 1)); in GST_START_TEST()
99 ret = gst_audio_buffer_clip (buf, &s, 100, 1); in GST_START_TEST()
141 ret = gst_audio_buffer_clip (buf, &s, 100, 1); in GST_START_TEST()
177 ret = gst_audio_buffer_clip (buf, &s, 100, 1); in GST_START_TEST()
220 ret = gst_audio_buffer_clip (buf, &s, 100, 2); in GST_START_TEST()
263 ret = gst_audio_buffer_clip (buf, &s, 100, 1); in GST_START_TEST()
306 ret = gst_audio_buffer_clip (buf, &s, 100, 2); in GST_START_TEST()
347 ret = gst_audio_buffer_clip (buf, &s, 100, 1); in GST_START_TEST()
371 ret = gst_audio_buffer_clip (buf, &s, 100, 1); in GST_START_TEST()
408 ret = gst_audio_buffer_clip (buf, &s, 100, 1); in GST_START_TEST()
[all …]
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| /third_party/gstreamer/gstplugins_base/gst-libs/gst/audio/ |
| D | audio.h | 96 GstBuffer * gst_audio_buffer_clip (GstBuffer *buffer,
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| D | gstaudiofilter.c | 184 gst_audio_buffer_clip (input, &btrans->segment, filter->info.rate, in gst_audio_filter_submit_input_buffer()
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| D | audio.c | 79 gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment, in gst_audio_buffer_clip() function
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| D | gstaudioencoder.c | 1286 buffer = gst_audio_buffer_clip (buffer, &enc->input_segment, ctx->info.rate, in gst_audio_encoder_chain()
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| D | gstaudioaggregator.c | 1691 buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf); in gst_audio_aggregator_do_clip()
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| D | gstaudiodecoder.c | 994 buf = gst_audio_buffer_clip (buf, &dec->output_segment, ctx->info.rate, in gst_audio_decoder_push_forward()
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| /third_party/gstreamer/gstplugins_bad/gst/segmentclip/ |
| D | gstaudiosegmentclip.c | 125 gst_audio_buffer_clip (buffer, segment, self->rate, self->framesize); in gst_audio_segment_clip_clip_buffer()
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| /third_party/gstreamer/gstplugins_base/gst/audiorate/ |
| D | gstaudiorate.c | 599 fill = gst_audio_buffer_clip (fill, &audiorate->src_segment, rate, bpf); in gst_audio_rate_chain()
677 buf = gst_audio_buffer_clip (buf, &audiorate->src_segment, rate, bpf); in gst_audio_rate_chain()
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| /third_party/gstreamer/gstplugins_bad/gst/timecode/ |
| D | gstavwait.c | 1299 gst_audio_buffer_clip (inbuf, &asegment2, self->ainfo.rate, in gst_avwait_asink_chain()
1323 gst_audio_buffer_clip (inbuf, &asegment2, self->ainfo.rate, in gst_avwait_asink_chain()
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| /third_party/gstreamer/gstplugins_base/gst/audioconvert/ |
| D | gstaudioconvert.c | 922 gst_audio_buffer_clip (input, &base->segment, this->in_info.rate, in gst_audio_convert_submit_input_buffer()
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| /third_party/gstreamer/gstplugins_bad/gst/audiobuffersplit/ |
| D | gstaudiobuffersplit.c | 733 buffer = gst_audio_buffer_clip (buffer, &self->in_segment, rate, bpf); in gst_audio_buffer_split_sink_chain()
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| /third_party/gstreamer/gstplugins_good/gst/audiofx/ |
| D | gstscaletempo.c | 555 gst_audio_buffer_clip (input, &scaletempo->in_segment, in gst_scaletempo_submit_input_buffer()
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| /third_party/gstreamer/gstplugins_base/gst/audioresample/ |
| D | gstaudioresample.c | 981 gst_audio_buffer_clip (input, &base->segment, resample->in.rate, in gst_audio_resample_submit_input_buffer()
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| /third_party/gstreamer/gstplugins_base/gst/adder/ |
| D | gstadder.c | 1066 buffer = gst_audio_buffer_clip (buffer, &data->segment, rate, bpf); in gst_adder_do_clip()
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| /third_party/gstreamer/gst_libav/ |
| D | ChangeLog | 6291 avdec: Use gst_audio_buffer_clip() instead of manual clipping
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| /third_party/gstreamer/gstplugins_base/ |
| D | ChangeLog | 18564 Essentially this moves the truncation logic out of gst_audio_buffer_clip()
18571 tests: audio: add unit tests for planar audio support in gst_audio_buffer_clip()
18577 libs: audio: implement support for non-interleaved audio in gst_audio_buffer_clip()
50251 audio: Add const to segment parameter of gst_audio_buffer_clip()
140262 * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
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