Current File : //lib64/gstreamer-1.0/libgstalsa.so
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Z p! / r X __gmon_start__ _ITM_deregisterTMCloneTable _ITM_registerTMCloneTable __cxa_finalize g_strdup_vprintf gst_debug_log g_free snd_strerror __stack_chk_fail gst_element_register _gst_debug_min bindtextdomain bind_textdomain_codeset snd_lib_error_set_handler _gst_debug_category_new gst_plugin_alsa_get_desc gst_plugin_alsa_register gst_plugin_register_static g_strdup g_mutex_init g_mutex_lock g_mutex_unlock stdout snd_output_stdio_attach g_type_class_peek_parent gst_element_class_set_static_metadata gst_element_class_add_static_pad_template _gst_debug_register_funcptr g_param_spec_string g_object_class_install_property g_type_class_adjust_private_offset snd_pcm_drop snd_pcm_prepare snd_pcm_delay snd_pcm_wait g_strerror g_usleep snd_pcm_resume gst_audio_base_sink_report_device_failure snd_pcm_writei snd_pcm_frames_to_bytes dcgettext _gst_element_error_printf gst_resource_error_quark gst_element_message_full snd_pcm_close gst_mini_object_replace snd_pcm_hw_free snd_pcm_hw_params_malloc snd_pcm_format_name snd_pcm_hw_params_any snd_pcm_hw_params_set_access snd_pcm_hw_params_set_format snd_pcm_hw_params_set_channels snd_pcm_hw_params_set_rate_near snd_pcm_hw_params_get_buffer_time_min snd_pcm_hw_params_get_buffer_time_max snd_pcm_hw_params_get_period_time_min snd_pcm_hw_params_get_period_time_max snd_pcm_hw_params_get_periods_min snd_pcm_hw_params_get_periods_max snd_pcm_hw_params_set_buffer_size_near snd_pcm_hw_params_set_period_size_near snd_pcm_hw_params_set_buffer_time_near snd_pcm_hw_params_set_period_time_near gst_audio_ring_buffer_set_channel_positions snd_pcm_hw_params snd_pcm_hw_params_get_buffer_size snd_pcm_hw_params_get_period_size snd_pcm_hw_params_free snd_pcm_sw_params_malloc snd_pcm_sw_params_current snd_pcm_sw_params_set_start_threshold snd_pcm_sw_params_set_avail_min snd_pcm_sw_params snd_pcm_sw_params_free snd_output_buffer_open snd_pcm_dump_hw_setup snd_output_buffer_string snd_output_close snd_pcm_dump_sw_setup g_strdup_printf snd_pcm_open gst_pad_query_caps gst_caps_is_empty gst_mini_object_unref gst_caps_is_fixed gst_audio_ring_buffer_parse_caps gst_caps_get_structure gst_structure_get_boolean gst_audio_iec61937_frame_size gst_caps_intersect_full gst_mini_object_ref gst_element_class_get_pad_template gst_pad_template_get_caps g_assertion_message_expr g_mutex_clear gst_buffer_new_allocate gst_buffer_map gst_audio_iec61937_payload gst_buffer_unmap gst_buffer_copy_into gst_query_parse_accept_caps gst_query_set_accept_caps_result g_type_name g_value_dup_string g_value_take_string g_value_set_string g_once_init_enter g_intern_static_string gst_audio_sink_get_type g_type_register_static_simple g_once_init_leave gst_element_get_clock gst_system_clock_get_type g_type_check_instance_is_a g_object_get gst_object_unref snd_pcm_readi snd_pcm_status_malloc snd_pcm_status snd_pcm_status_get_state snd_pcm_status_get_htstamp snd_pcm_status_get_avail gst_util_uint64_scale_int snd_pcm_status_free snd_pcm_nonblock snd_pcm_sw_params_set_tstamp_mode g_return_if_fail_warning gst_audio_src_get_type gst_base_src_set_format gst_base_src_set_live gst_buffer_new g_memdup gst_memory_new_wrapped gst_buffer_append_memory _gst_debug_dump_mem gst_buffer_list_insert gst_poll_set_flushing gst_poll_free snd_midi_event_free snd_seq_close snd_seq_port_info_sizeof memset snd_seq_port_info_set_name snd_seq_port_info_set_type snd_seq_port_info_set_capability snd_seq_alloc_named_queue snd_seq_port_info_set_timestamping snd_seq_port_info_set_timestamp_real snd_seq_port_info_set_timestamp_queue snd_seq_create_port snd_seq_control_queue snd_seq_drain_output snd_seq_open snd_config_update_free_global snd_seq_set_client_name g_strsplit g_strv_length g_try_malloc_n g_strfreev snd_seq_connect_from snd_seq_nonblock snd_midi_event_new snd_midi_event_init snd_midi_event_reset_decode snd_midi_event_no_status g_try_malloc snd_seq_poll_descriptors_count snd_seq_poll_descriptors gst_poll_new gst_poll_add_fd gst_poll_fd_ctl_read gst_poll_fd_ctl_write snd_seq_parse_address snd_seq_client_id snd_seq_event_output gst_buffer_list_new gst_poll_wait snd_midi_event_decode snd_seq_event_input gst_buffer_list_length gst_buffer_list_get gst_mini_object_copy gst_buffer_list_remove gst_pad_push_list __errno_location gst_clock_get_time gst_element_get_base_time snd_seq_queue_status_malloc snd_seq_get_queue_status snd_seq_queue_status_get_real_time snd_seq_queue_status_free gst_push_src_get_type snd_ctl_open snd_ctl_card_info_malloc snd_ctl_card_info snd_ctl_card_info_get_card snd_card_get_name snd_ctl_card_info_free snd_ctl_close snd_ctl_pcm_next_device snd_pcm_info_malloc snd_pcm_info_set_device snd_pcm_info_set_subdevice snd_pcm_info_set_stream snd_ctl_pcm_info snd_pcm_info_get_name snd_pcm_info_free g_type_check_value_holds g_value_get_string gst_audio_format_from_string gst_audio_format_get_info snd_pcm_format_mask_test snd_pcm_format_mask_malloc snd_pcm_hw_params_get_format_mask gst_caps_get_size gst_structure_has_name gst_structure_get_value _gst_value_list_type g_value_init gst_value_list_get_size gst_structure_copy gst_structure_take_value gst_caps_append_structure snd_pcm_format_mask_free gst_value_list_get_value gst_structure_set_value g_value_unset gst_value_list_append_value gst_caps_new_empty __sprintf_chk snd_pcm_stream snd_pcm_hw_params_get_rate_min snd_pcm_hw_params_get_rate_max gst_mini_object_make_writable _gst_int_range_type gst_structure_set snd_pcm_hw_params_get_channels_min snd_pcm_hw_params_get_channels_max gst_structure_get_field_type gst_structure_remove_field _gst_bitmask_type gst_caps_merge_structure gst_structure_get_int gst_value_get_int_range_min gst_value_get_int_range_max gst_caps_from_string gst_caps_append strchr strtol snd_pcm_info snd_pcm_get_chmap gst_audio_channel_positions_to_string libgstaudio-1.0.so.0 libgsttag-1.0.so.0 libz.so.1 libm.so.6 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 libasound.so.2 libpthread.so.0 libc.so.6 _edata __bss_start _end libgstalsa.so GLIBC_2.2.5 GLIBC_2.3.4 GLIBC_2.4 ALSA_0.9.0rc8 ALSA_0.9.0rc4 ALSA_0.9 9 ui y I @ ti ii
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* PHpf H$ H81g[Y^LTHL^^ HH : alsalib error: %s%s%s alsasrc alsamidisrc alsa plugins alsa gstalsaplugin.c /usr/share/locale gst-plugins-base-1.0 UTF-8 1.16.1 ALSA plugin library http://www.redhat.com gst-plugins-base LGPL 2019-09-23 binding text domain %s to locale dir %s failed to set alsa error handler GStreamer-plugins-base package plugin_init gstalsasink.c initializing alsasink default Wim Taymans <wim@fluendo.com> Sink/Audio Audio sink (ALSA) gst_alsasink_getcaps gst_alsasink_query gst_alsasink_payload gst_alsasink_open gst_alsasink_prepare gst_alsasink_unprepare gst_alsasink_close gst_alsasink_write gst_alsasink_delay gst_alsasink_reset Device device Device name device-name Card name card-name drop reset done snd_pcm_delay returned %d swapping bytes wait error, %d written %d frames out of %d Write error: %s xrun recovery %d: %s Error parsing spec periods min %u, max %u warning: %s buffer time %u period time %u Hardware setup:
%s Software setup:
%s Opened device %s Device '%s' is busy framed parsed Returning cached caps %pA returning caps %pA property GstAlsaSink Output to a sound card via ALSA ALSA device, as defined in an asound configuration file Human-readable name of the sound device Human-readable name of the sound card Error %d (%s) calling snd_pcm_drop (alsa->handle) Error %d (%s) calling snd_pcm_prepare (alsa->handle) alsa-reset: pcm drop error: %s alsa-reset: pcm prepare error: %s snd_pcm_delay returned negative delay received audio samples buffer of %u bytes Error outputting to audio device. The device has been disconnected. Can't recover from underrun, prepare failed: %s Can't recover from suspend, prepare failed: %s Could not open IEC958 (SPDIF) device for playback Negotiating to %d channels @ %d Hz (format = %s) SPDIF (%d) Error %d (%s) calling snd_pcm_hw_params_any (alsa->handle, params) Broken configuration for playback: no configurations available: %s Error %d (%s) calling snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access) Access type not available for playback: %s falling back to little endian with swapping Error %d (%s) calling snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) Sample format not available for playback: %s Error %d (%s) calling snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels) Error %d (%s) calling snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL) Rate %iHz not available for playback: %s buffer time %u, min %u, max %u period time %u, min %u, max %u Unable to set buffer time %i for playback: %s Unable to set period time %i for playback: %s Error %d (%s) calling snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params, &buffer_size) Error %d (%s) calling snd_pcm_hw_params_set_period_size_near (alsa->handle, params, &period_size, NULL) Error %d (%s) calling snd_pcm_hw_params (alsa->handle, params) Unable to set hw params for playback: %s Error %d (%s) calling snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size) Error %d (%s) calling snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL) buffer size %lu, period size %lu Could not open device for playback in mono mode. Could not open device for playback in stereo mode. Could not open device for playback in %d-channel mode. Unable to get buffer size for playback: %s Unable to get period size for playback: %s Error %d (%s) calling set_hwparams (alsa) Setting of hwparams failed: %s Error %d (%s) calling snd_pcm_sw_params_current (alsa->handle, params) Unable to determine current swparams for playback: %s Error %d (%s) calling snd_pcm_sw_params_set_start_threshold (alsa->handle, params, (alsa->buffer_size / alsa->period_size) * alsa->period_size) Unable to set start threshold mode for playback: %s Error %d (%s) calling snd_pcm_sw_params_set_avail_min (alsa->handle, params, alsa->period_size) Unable to set avail min for playback: %s Error %d (%s) calling snd_pcm_sw_params (alsa->handle, params) Unable to set sw params for playback: %s Error %d (%s) calling set_swparams (alsa) Setting of swparams failed: %s Error %d (%s) calling snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK) Could not open audio device for playback. Device is being used by another application. Could not open audio device for playback. Playback open error on device '%s': %s device not open, using template caps Returning cached caps %pA with filter %pA applied: %pA %s:%d: invalid %s id %u for "%s" of type '%s' in '%s' 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 }, layout = (string) interleaved, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-ac3, framed = (boolean) true;audio/x-eac3, framed = (boolean) true; audio/x-dts, framed = (boolean) true, block-size = (int) { 512, 1024, 2048 }; audio/mpeg, mpegversion = (int) 1, mpegaudioversion = (int) [ 1, 3 ], parsed = (boolean) true; peee`a``aaa``||||!~~~}}}}}}}}|}m}^}O}@}1}"}}}||||}|n| gst_alsasink_getcaps gst_alsasink_getcaps gst_alsasink_open set_swparams set_hwparams gst_alsasink_prepare xrun_recovery gst_alsasink_write gst_alsasink_delay gst_alsasink_reset gst_alsasink_init gstalsasrc.c initializing Source/Audio Audio source (ALSA) gst_alsasrc_getcaps gst_alsasrc_open gst_alsasrc_prepare gst_alsasrc_unprepare gst_alsasrc_close gst_alsasrc_read gst_alsasrc_delay gst_alsasrc_reset gst_alsasrc_change_state clock-type Using driver timestamps ! Read error: %s snd_pcm_status_malloc failed snd_pcm_status failed Prepare failed: %s Returning cached caps pad_template != NULL template caps %pA GstAlsaSrc Read from a sound card via ALSA Error recording from audio device. The device has been disconnected. Could not recover from xrun condition ! ALSA timestamp : %u:%02u:%02u.%09u, delay %lu Could not set device to blocking: %s Broken configuration for recording: no configurations available: %s Access type not available for recording: %s Sample format not available for recording: %s Rate %iHz not available for recording: %s Rate doesn't match (requested %iHz, get %iHz) Unable to set buffer time %i for recording: %s Unable to set period time %i for recording: %s Unable to set hw params for recording: %s Unable to get buffer size for recording: %s Unable to get period size for recording: %s Could not open device for recording in mono mode. Could not open device for recording in stereo mode. Could not open device for recording in %d-channel mode Unable to set tstamp mode for playback: %s Could not open audio device for recording. Device is being used by another application. Could not open audio device for recording. Recording open error on device '%s': %s 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 }, layout = (string) interleaved, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ] ((Π)8GVet9HWfuϟޟ ݠgst_alsasrc_getcaps gst_alsasrc_getcaps gst_alsasrc_open set_swparams set_hwparams gst_alsasrc_prepare gst_alsasrc_get_timestamp xrun_recovery gst_alsasrc_read gst_alsasrc_delay gst_alsasrc_reset gst_alsasrc_change_state gst_alsasrc_init Comma separated list of sequencer ports (e.g. client:port,...) Push ALSA MIDI sequencer events around gst_alsa_midi_src_state_changed Cannot start timer for queue: %d - %s Cannot connect from port %d:%d - %s Warning: Received non-MIDI message Error decoding event from ALSA to output: %s Ports ports Antonio Ospite <ao2@ao2.it> AlsaMidi Source gst_alsa_midi_src_start gst_alsa_midi_src_stop gst_alsa_midi_src_unlock gst_alsa_midi_src_unlock_stop gst_alsa_midi_src_create MIDI data: gstalsamidisrc.c Cannot allocate queue: %s
Cannot create port - %s Timer event output error: %s
Drain output error: %s
Cannot open sequencer - %s Cannot set client name - %s ports: %s , Out of memory Invalid port %s - %s Cannot set nonblock mode - %s Event output error: %s
Event drain error: %s
flushing ERROR in poll: %s No clock present GstAlsaMidiSrc alsamidisrc element GST_IS_ALSA_MIDI_SRC (object) audio/x-midi-event gst_alsa_midi_src_get_property connect_ports start_queue_timer create_port parse_ports init_seq gst_alsa_midi_src_start push_buffer gst_alsa_midi_src_create schedule_next_tick gst_alsa_midi_src_state_changed gstalsa.c [%s] device=%d name from pcminfo: %s trying card name audio/x-raw skipping non-raw format Generated device string "%s" probing sample rates ... Min. rate = %u (%d) Max. rate = %u (%d) rate probing channels ... Min. channels = %d (%d) Max. channels = %d (%d) channels min_chans >= 1 channel-mask failed to query formats: %s failed to query formats (NULL) got channels mapping %s %s:{AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x} failed opening IEC958 device: %s no formats in native endianness detected failed to query minimum sample rate: %s failed to query maximum sample rate: %s minimum channels > maximum channels (%d > %d), please fix your soundcard drivers min_chans = %u, looks like a pro card failed to query minimum channel count: %s failed to query maximum channel count: %s audio/x-ac3, framed = (boolean) true;audio/x-eac3, framed = (boolean) true; audio/x-dts, framed = (boolean) true, block-size = (int) { 512, 1024, 2048 }; audio/mpeg, mpegversion = (int) 1, mpegaudioversion = (int) [ 1, 3 ], parsed = (boolean) true; Trying to get device name from string '%s' Trying to get device name from open handle Device name for device '%s': %s ALSA driver does not implement channels mapping API got channels mapping for %u channels but stream has %u channels; ignoring failed to convert ALSA channels mapping
! " # alsa_detect_channels_mapping gst_alsa_find_device_name gst_alsa_find_device_name_no_handle caps_add_channel_configuration gst_alsa_detect_channels gst_alsa_detect_rates gst_alsa_detect_formats gst_alsa_probe_supported_formats gst_alsa_open_iec958_pcm
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