Current File : //lib64/gstreamer-1.0/libgstaudioresample.so
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U 0 B ? p a C q > , + , F " ] Q w ( $ W Q v 4 C V ( 3 n q @ L ^ X Y X __gmon_start__ _ITM_deregisterTMCloneTable _ITM_registerTMCloneTable __cxa_finalize gst_base_transform_set_gap_aware gst_pad_set_query_function_full g_type_class_peek_parent g_param_spec_int g_object_class_install_property gst_audio_resampler_method_get_type g_param_spec_enum gst_audio_resampler_filter_mode_get_type g_param_spec_uint gst_audio_resampler_filter_interpolation_get_type gst_element_class_add_static_pad_template gst_element_class_set_static_metadata _gst_debug_register_funcptr g_type_class_adjust_private_offset gst_audio_buffer_clip gst_meta_api_type_get_tags g_strv_length g_quark_from_string gst_meta_api_type_has_tag gst_audio_converter_get_out_frames gst_buffer_new_allocate gst_audio_buffer_map gst_audio_converter_samples gst_audio_buffer_unmap _gst_debug_min gst_debug_log gst_pad_push gst_flow_get_name g_assertion_message_expr gst_util_uint64_scale_int_round gst_buffer_add_audio_meta __stack_chk_fail gst_audio_converter_get_max_latency gst_audio_converter_reset gst_mini_object_is_writable gst_buffer_set_size memset gst_buffer_n_memory gst_buffer_peek_memory gst_util_uint64_scale_int_ceil g_malloc g_free gst_buffer_get_size gst_audio_converter_free gst_structure_new_empty gst_structure_set gst_structure_free gst_audio_converter_new gst_audio_resampler_options_set_quality gst_message_new_latency gst_element_post_message gst_audio_converter_update_config gst_audio_info_from_caps gst_caps_get_structure gst_structure_get_int gst_caps_truncate gst_mini_object_make_writable gst_structure_fixate_field_nearest_int gst_caps_new_empty gst_caps_get_size gst_structure_copy gst_caps_append_structure _gst_int_range_type gst_caps_is_subset_structure gst_structure_get_value gst_caps_intersect_full gst_mini_object_unref gst_audio_converter_get_in_frames g_value_get_enum g_value_get_int g_value_get_uint g_type_name gst_base_transform_is_passthrough gst_pad_peer_query gst_pad_query_default gst_query_parse_latency gst_query_set_latency gst_util_uint64_scale_round g_value_set_enum g_value_set_int g_value_set_uint g_once_init_enter g_intern_static_string gst_base_transform_get_type g_type_register_static_simple g_once_init_leave gst_element_register _gst_debug_category_new gst_plugin_audioresample_get_desc gst_plugin_audioresample_register gst_plugin_register_static libgstaudio-1.0.so.0 libgsttag-1.0.so.0 libz.so.1 liborc-0.4.so.0 libgstbase-1.0.so.0 libgstreamer-1.0.so.0 libgobject-2.0.so.0 libglib-2.0.so.0 libm.so.6 libpthread.so.0 libc.so.6 _edata __bss_start _end libgstaudioresample.so GLIBC_2.4 GLIBC_2.2.5 ii
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PHE H P}H( HH Resample quality with 0 being the lowest and 10 being the best What sinc filter table mode to use Memory usage threshold to use if sinc filter mode is AUTO, given in bytes Sinc filter auto mode threshold How to interpolate the sinc filter table Sebastian Dröge <sebastian.droege@collabora.co.uk> gst_audio_resample_transform_size gst_audio_resample_get_unit_size gst_audio_resample_transform_caps gst_audio_resample_fixate_caps gst_audio_resample_transform_meta gst_audio_resample_submit_input_buffer Pushing drain buffer of %u bytes with timestamp %u:%02u:%02u.%09u duration %u:%02u:%02u.%09u offset %lu offset_end %lu Converted to buffer of %lu samples (%lu bytes) with timestamp %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u, offset %lu, offset_end %lu transforming buffer of %lu bytes, ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u, offset %li, offset_end %li encountered timestamp discontinuity of %lu samples = %u:%02u:%02u.%09u found discontinuity; resyncing ... but new timestamp is invalid marking this buffer with the DISCONT flag input = samples [%lu, %lu) = [%lu, %lu) ns; output = samples [%lu, %lu) = [%lu, %lu) ns GstAudioConverter.resampler-method GstAudioResampler.filter-interpolation GstAudioResampler.filter-mode-threshold asked to transform size %lu in direction %s %s:%d: invalid %s id %u for "%s" of type '%s' in '%s' Peer latency: min %u:%02u:%02u.%09u max %u:%02u:%02u.%09u Our latency: %u:%02u:%02u.%09u Calculated total latency : min %u:%02u:%02u.%09u max %u:%02u:%02u.%09u GStreamer-plugins-base package audio/x-raw, format = (string) { S8, U8, S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE }, rate = (int) [ 1, max ], channels = (int) [ 1, max ], layout = (string) { interleaved, non-interleaved } Quality quality What resample method to use Resample method to use resample-method Sinc filter table mode sinc-filter-mode sinc-filter-auto-threshold Sinc filter interpolation sinc-filter-interpolation Resamples audio Filter/Converter/Audio Audio resampler gst_audio_resample_start gst_audio_resample_stop gst_audio_resample_set_caps gst_audio_resample_transform gst_audio_resample_sink_event resample->converter != NULL gstaudioresample.c Failed to push drain: %s ... but new offset is invalid resampler-options GstAudioResampler.filter-mode failed to create resampler failed to update resampler incaps %pA, outcaps %pA invalid incaps invalid outcaps rate SINK SRC transformed size %lu to %lu new quality %d property invalid caps GstAudioResample audio resampling element audioresample 1.16.1 http://www.redhat.com gst-plugins-base LGPL 2019-09-23 src sink x xh gst_audio_resample_set_property gst_audio_resample_transform_size gst_audio_resample_get_unit_size gst_audio_resample_update_state gst_audio_resample_set_caps gst_audio_resample_process gst_audio_resample_check_discont gst_audio_resample_transform gst_audio_resample_push_drain gst_audio_resample_push_drain gst_audio_resample_query ; ĺ d , @ t\ 4 | 4 $0 dL T p t H 4 T t D D T0 zR x $ й FJ w ?:*3$" D H \ \ p V OB 4 L Eje XDM aF
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