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__gmon_start__ __cxa_finalize _Jv_RegisterClasses gst_audio_buffer_clip gst_buffer_get_type g_type_check_instance_is_a gst_segment_clip gst_util_uint64_scale gst_buffer_make_metadata_writable g_return_if_fail_warning gst_mini_object_unref gst_value_list_get_type g_value_init g_value_peek_pointer g_array_append_vals g_value_set_boolean g_value_set_int gst_structure_set_value gst_audio_structure_set_int gst_int_range_get_type gst_structure_set gst_audio_duration_from_pad_buffer g_assert_warning gst_caps_get_structure gst_structure_get_int gst_audio_frame_byte_size gst_audio_is_buffer_framed gst_audio_frame_length gst_audio_clock_reset gst_audio_clock_get_type gst_system_clock_get_type g_type_register_static g_type_check_instance_cast gst_audio_clock_new g_object_new g_type_class_peek_parent gst_element_set_state gst_element_get_state __gst_debug_min gst_mixer_get_type gst_implements_interface_check gst_implements_interface_cast gst_mixer_list_tracks GST_CAT_DEFAULT gst_debug_log g_list_prepend gst_element_factory_create gst_audio_default_registry_mixer_filter gst_element_factory_get_type gst_registry_get_default gst_registry_get_feature_list g_list_sort gst_element_factory_get_klass gst_property_probe_get_type gst_property_probe_get_property gst_property_probe_probe_and_get_values g_value_array_get_nth g_object_set_property g_value_array_free gst_object_unref gst_plugin_feature_list_free g_list_reverse g_value_get_string gst_element_factory_get_longname gst_plugin_feature_get_type gst_plugin_feature_get_rank g_ascii_strcasecmp gst_audio_set_structure_channel_positions_list gst_value_array_get_type gst_audio_channel_position_get_type g_value_set_enum gst_value_list_append_value gst_value_array_append_value g_value_unset gst_caps_new_empty gst_structure_get_value g_type_name gst_value_list_get_size gst_value_list_get_value gst_caps_append gst_structure_copy gst_caps_append_structure gst_value_get_int_range_min gst_value_get_int_range_max gst_audio_set_caps_channel_positions_list gst_caps_get_size gst_audio_check_channel_positions gst_audio_fixate_channel_positions gst_value_array_get_size gst_value_array_get_value g_malloc g_value_get_enum g_free gst_audio_set_channel_positions gst_audio_get_channel_positions g_memdup gst_audio_filter_get_type gst_base_transform_get_type _gst_debug_category_new gst_audio_filter_class_add_pad_templates gst_element_get_type g_type_check_class_cast g_type_check_class_is_a gst_caps_get_type gst_caps_copy gst_pad_template_new gst_element_class_add_pad_template _gst_debug_register_funcptr gst_ring_buffer_get_type g_type_class_ref gst_caps_is_equal gst_ring_buffer_parse_caps gst_caps_replace memset g_thread_functions_for_glib_use gst_audio_sink_get_type g_intern_static_string gst_base_audio_sink_get_type gst_type_register_static_full g_threads_got_initialized g_thread_join gst_buffer_new_and_alloc g_thread_create_full gst_ring_buffer_prepare_read gst_ring_buffer_clear gst_ring_buffer_advance __errno_location g_strerror gst_audio_src_get_type gst_base_audio_src_get_type gst_base_audio_sink_create_ringbuffer gst_object_get_type gst_object_set_parent gst_base_sink_get_type gst_base_audio_sink_get_slave_method gst_base_audio_sink_set_slave_method gst_base_audio_sink_get_provide_clock gst_base_audio_sink_set_provide_clock g_type_class_add_private g_param_spec_int64 g_object_class_install_property g_param_spec_boolean g_enum_register_static g_param_spec_enum gst_ring_buffer_set_callback gst_ring_buffer_start gst_ring_buffer_release gst_ring_buffer_is_acquired gst_util_uint64_scale_int gst_base_sink_wait_eos gst_pad_pull_range memcpy gst_message_new_eos gst_element_post_message gst_ring_buffer_may_start gst_structure_fixate_field_nearest_int gst_structure_has_field gst_structure_fixate_field_boolean _gst_element_error_printf gst_stream_error_quark gst_element_message_full gst_ring_buffer_debug_spec_buff gst_ring_buffer_acquire gst_clock_get_calibration gst_ring_buffer_samples_done gst_ring_buffer_delay gst_base_sink_wait_preroll gst_ring_buffer_commit_full gst_segment_to_running_time gst_element_get_base_time gst_base_sink_get_latency gst_clock_get_time gst_clock_get_internal_time gst_clock_set_calibration gst_base_sink_wait_clock gst_clock_set_master gst_core_error_quark gst_event_parse_new_segment_full gst_ring_buffer_set_flushing gst_base_sink_query_latency gst_query_set_latency gst_object_ref gst_ring_buffer_pause gst_ring_buffer_open_device gst_ring_buffer_close_device gst_object_unparent g_value_set_int64 g_value_get_int64 g_value_get_boolean g_type_instance_get_private gst_base_audio_src_create_ringbuffer gst_push_src_get_type bindtextdomain gst_base_audio_src_get_slave_method gst_base_audio_src_set_slave_method gst_base_audio_src_get_provide_clock gst_base_audio_src_set_provide_clock gst_base_src_wait_playing gst_ring_buffer_read gst_resource_error_quark dcgettext gst_base_src_get_type gst_ring_buffer_clear_all g_object_notify gst_base_src_set_live gst_base_src_set_format gst_ring_buffer_debug_spec_caps g_atomic_int_compare_and_exchange gst_structure_get_name gst_structure_get_boolean g_atomic_int_add gst_ring_buffer_set_sample gst_ring_buffer_stop gst_ring_buffer_device_is_open gst_ring_buffer_commit libgstbase-0.10.so.0 libgstreamer-0.10.so.0 libgobject-2.0.so.0 libgmodule-2.0.so.0 libdl.so.2 libgthread-2.0.so.0 libxml2.so.2 libz.so.1 libm.so.6 libglib-2.0.so.0 libgstinterfaces-0.10.so.0 libpthread.so.0 libc.so.6 libgstaudio-0.10.so.0 GLIBC_2.1.3 GLIBC_2.0 8 0 s i X ii
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GST_IS_BUFFER (buffer) structure != NULL rate channels endianness width depth signed GST_IS_BUFFER (buf) audio.c (NULL) '' bytes != 0 width != 0 channels != 0 rate != 0 gst_audio_buffer_clip _gst_audio_structure_set_list gst_audio_duration_from_pad_buffer GstAudioClock name could not open device / set element to READY element is a mixer, but has no tracks element is a mixer with mixer tracks could not create element from factory there are %d available devices Stopping after first found mixer, as requested element does not support the property probe interface mixerutils.c element is not a mixer Generic/Audio probing %s ... created element %s (%p) probing available devices ... trying device %s .. gst_audio_default_registry_mixer_filter gst_audio_mixer_filter_probe_feature gst_audio_mixer_filter_check_element str != NULL num_positions > 0 pos != NULL res channels > 0 channel-positions multichannel.c pos_val_arr != NULL Unexpected value type '%s' for channels field Channel position %d for channel %d is invalid Either all channel positions must be defined, or all be set to NONE, having only some defined is not allowed Channel position %d occurred %d times, not allowed Found conflicting channel positions %d/%d and %d gst_value_array_get_size (pos_val_arr) == channels t == GST_TYPE_LIST || t == GST_TYPE_AUDIO_CHANNEL_POSITION Pre-fixated on both %d/%d and %d - conflict! Pre-fixated one side, but other side n/a of %d/%d conf[i].pos1[0] != GST_AUDIO_CHANNEL_POSITION_INVALID && conf[i].pos1[1] != GST_AUDIO_CHANNEL_POSITION_INVALID conf[i].pos2[0] != GST_AUDIO_CHANNEL_POSITION_INVALID %d unfixed channel positions left after fixation! Failed to retrieve channel layout from caps. This usually means there is a GStreamer element that does not implement multichannel audio correctly. Please file a bug. t == GST_TYPE_AUDIO_CHANNEL_POSITION gst_audio_fixate_channel_positions
gst_audio_set_caps_channel_positions_list gst_audio_set_structure_channel_positions_list gst_audio_set_channel_positions
gst_audio_get_channel_positions gst_audio_check_channel_positions audiofilter GstAudioFilter allowed_caps != NULL GST_IS_CAPS (allowed_caps) gst_audio_filter_change_state gst_audio_filter_set_caps gstaudiofilter.c caps: %P couldn't parse %P GST_IS_AUDIO_FILTER_CLASS (klass) gst_audio_filter_get_unit_size gst_caps_is_equal (incaps, outcaps) gst_audio_filter_class_add_pad_templates gst_audio_filter_set_caps GstAudioSink gst_audioringbuffer_dispose gst_audioringbuffer_finalize gst_audioringbuffer_acquire gst_audioringbuffer_release gst_audioringbuffer_start gst_audioringbuffer_pause gst_audioringbuffer_stop gst_audioringbuffer_delay creating ringbuffer gstaudiosink.c GstAudioSinkRingBuffer created ringbuffer @%p reset... reset done stop, waiting... start, sending signal could not close device could not open device signal wait unprepared could not unprepare device could not prepare device enter thread wait for action got signal continue running stop running, exit thread gst_audio_sink_create_ringbuffer gst_audioringbuffer_open_device gst_audioringbuffer_close_device transfered %d bytes of %d from segment %d error writing data (reason: %s), skipping segment no write function, exit thread gst_audio_sink_create_ringbuffer gst_audioringbuffer_stop gst_audioringbuffer_pause gst_audioringbuffer_start gst_audioringbuffer_release gst_audioringbuffer_acquire audioringbuffer_thread_func gst_audioringbuffer_close_device gst_audioringbuffer_open_device GstAudioSrc GstAudioSrcRingBuffer gstaudiosrc.c stoped gst_audio_src_create_ringbuffer transfered %d bytes of %d to segment %d error reading data (reason: %s), skipping segment gst_audio_src_create_ringbuffer gst_audioringbuffer_stop gst_audioringbuffer_start audioringbuffer_thread_func GstBaseAudioSink baseaudiosink element baseaudiosink GST_IS_BASE_AUDIO_SINK (sink) gst_base_audio_sink_dispose Audio latency in microseconds Provide Clock provide-clock GstBaseAudioSinkSlaveMethod Slave Method slave-method gst_base_audio_sink_query gst_base_audio_sink_event gst_base_audio_sink_preroll gst_base_audio_sink_render gst_base_audio_sink_get_times gst_base_audio_sink_setcaps gst_base_audio_sink_fixate gstbaseaudiosink.c EOS ringbuffer may start now ringbuffer in wrong state sink not negotiated. error: %s release old ringbuffer acquire new ringbuffer could not parse caps cannot parse audio format. could not acquire ringbuffer checking preroll latency synced late preroll, adding jitter Setting clock as master we have no clock we are not slaved we are flushing base_time %u:%02u:%02u.%09u unknown slaving method %d resync after discont warning: %s rendering at %llu %d/%d wrote %u of %u next sample expected at %llu ringbuffer not negotiated wrong size ringbuffer is stopping failed waiting for latency new segment rate of %f latency query ringbuffer not acquired clock provide disabled open failed property gstbaseaudiosink.c:543 gstbaseaudiosink.c:516 GstAudioSinkClock Resampling slaving resample Skew slaving skew No slaving none gst_base_audio_sink_set_property gst_base_audio_sink_get_property Size of audio buffer in microseconds Provide a clock to be used as the global pipeline clock Algorithm to use to match the rate of the masterclock gst_base_audio_sink_change_state gst_base_audio_sink_provide_clock gst_base_audio_sink_async_play gst_base_audio_sink_activate_pull last sample %llu, time %u:%02u:%02u.%09u pulling %d bytes offset %llu to fill audio buffer short read pulling from sink pad: %d<%d Got flow error but can't return it: %d processed samples: raw %llu, delay %u, real %llu, time %u:%02u:%02u.%09u possibly waiting for clock to reach %u:%02u:%02u.%09u clock returned %d %u:%02u:%02u.%09u internal time: %u:%02u:%02u.%09u external time: %u:%02u:%02u.%09u time %u:%02u:%02u.%09u, offset %llu, start %u:%02u:%02u.%09u, samples %u Buffer of size %u has no time. Using render_start=%llu clipping start to %u:%02u:%02u.%09u %llu samples clipping stop to %u:%02u:%02u.%09u %llu samples no sync needed. Using render_start=%llu running: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u compensating for latency %u:%02u:%02u.%09u after latency: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u internal %u:%02u:%02u.%09u external %u:%02u:%02u.%09u %llu/%llu = %f after slaving: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u internal %u:%02u:%02u.%09u external %u:%02u:%02u.%09u cinternal %u:%02u:%02u.%09u cexternal %u:%02u:%02u.%09u internal %u:%02u:%02u.%09u external %u:%02u:%02u.%09u skew %lli avg %lli correct clock skew %lli > %lli last_align %lli segsamples %u, next %llu correct clock skew %lli < %lli no align possible: no previous sample position known align with prev sample, ABS (%lli) < %d Compensating for audio synchronisation problems Unexpected discontinuity in audio timestamps of more than half a second (%u:%02u:%02u.%09u), resyncing no stop time align needed: we are slaved start playback because we are at the end of segment dropping sample out of segment time %u:%02u:%02u.%09u, start %u:%02u:%02u.%09u sink received buffer of wrong size. we are not yet negotiated, can't report latency yet peer min %u:%02u:%02u.%09u, our min latency: %u:%02u:%02u.%09u peer or we are not live, don't care about latency %s: invalid %s id %u for "%s" of type `%s' in `%s' _ ? N! ! g ! gst_base_audio_sink_get_slave_method gst_base_audio_sink_set_slave_method gst_base_audio_sink_get_provide_clock gst_base_audio_sink_set_provide_clock gst_base_audio_sink_callback gst_base_audio_sink_drain gst_base_audio_sink_async_play gst_base_audio_sink_setcaps gst_base_audio_sink_render gst_base_audio_sink_sync_latency gst_base_audio_sink_get_time gst_base_audio_sink_resample_slaving gst_base_audio_sink_skew_slaving gst_base_audio_sink_preroll gst_base_audio_sink_event gst_base_audio_sink_query gst_base_audio_sink_provide_clock gst_base_audio_sink_change_state GstBaseAudioSrc baseaudiosrc element baseaudiosrc /usr/share/locale gst-plugins-base-0.10 gstbaseaudiosrc.c GST_IS_BASE_AUDIO_SRC (src) gst_base_audio_src_dispose Actual Buffer Time actual-buffer-time Actual Latency Time actual-latency-time GstBaseAudioSrcSlaveMethod gst_base_audio_src_setcaps gst_base_audio_src_event gst_base_audio_src_query gst_base_audio_src_get_times gst_base_audio_src_create gst_base_audio_src_fixate reading from %d, we are at %d dropped, align to segment %d reading from sample %llu read %u of %u wait playing continue playing dropped %llu samples ringbuffer stopped NULL->READY READY->PAUSED PAUSED->PLAYING PLAYING->PAUSED PAUSED->READY READY->NULL gstbaseaudiosrc.c:519 gstbaseaudiosrc.c:476 GstAudioSrcClock Re-timestamp re-timestamp Skew binding text domain %s to locale dir %s gst_base_audio_src_set_property gst_base_audio_src_get_property Actual configured size of audio buffer in microseconds Actual configured audio latency in microseconds gst_base_audio_src_change_state gst_base_audio_src_provide_clock gst_base_audio_src_check_get_range create DISCONT of %llu samples at sample %llu Can't record audio fast enough resource can only be operated on sequentially but offset %llu was given report latency min %u:%02u:%02u.%09u max %u:%02u:%02u.%09u GEEEEEEEEFEEEEEEWEEFEEEEEE1Flatency_time != 0 GstRingBuffer ringbuffer class GST_IS_RING_BUFFER (buf) may start: %d clear segment %d @%p buf->data != NULL segment != NULL readptr != NULL len != NULL clear all segments failed to stop starting ringbuffer was started resuming failed to start we are not acquired start! waiting.. stopped processing not allowed to start releasing device buf->open == TRUE released device device was released failed to release device acquiring device acquired device device not opened Device for %p not opened device was acquired failed to acquire device closed device error closing device !buf->acquired opened device failed opening device gst_ring_buffer_dispose gst_ring_buffer_finalize copy %u bytes fwd_up end %d/%d fwd_down end %d/%d rev_up end %d/%d rev_down end %d/%d gst_ring_buffer_may_start gst_ring_buffer_may_start gst_ring_buffer_clear gst_ring_buffer_clear gst_ring_buffer_advance gst_ring_buffer_advance gst_ring_buffer_prepare_read gst_ring_buffer_prepare_read gst_ring_buffer_read gst_ring_buffer_read wait_segment gst_ring_buffer_commit_full gst_ring_buffer_commit_full gst_ring_buffer_clear_all gst_ring_buffer_clear_all gst_ring_buffer_set_sample gst_ring_buffer_set_sample gst_ring_buffer_samples_done gst_ring_buffer_delay gst_ring_buffer_delay gst_ring_buffer_stop gst_ring_buffer_stop gst_ring_buffer_pause gst_ring_buffer_pause gst_ring_buffer_pause_unlocked gst_ring_buffer_start gst_ring_buffer_start gst_ring_buffer_set_flushing gst_ring_buffer_is_acquired gst_ring_buffer_release gst_ring_buffer_release gst_ring_buffer_acquire gst_ring_buffer_acquire gst_ring_buffer_device_is_open gst_ring_buffer_close_device gst_ring_buffer_close_device gst_ring_buffer_open_device gst_ring_buffer_open_device gst_ring_buffer_set_callback gst_ring_buffer_parse_caps gst_ring_buffer_parse_caps
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