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W r k " h 6 3 l! h __gmon_start__ __cxa_finalize _Jv_RegisterClasses gst_property_probe_get_type g_type_add_interface_static g_str_equal g_type_name gst_element_get_type g_type_check_instance_cast gst_element_class_get_pad_template_list snd_ctl_card_info_malloc snd_pcm_info_malloc snd_card_next g_snprintf snd_ctl_open snd_ctl_card_info snd_ctl_pcm_next_device snd_pcm_info_set_device snd_pcm_info_set_subdevice snd_pcm_info_set_stream snd_ctl_pcm_info g_strdup_printf g_list_append g_strdup snd_ctl_close snd_ctl_card_info_free snd_pcm_info_free g_list_length g_value_array_new g_value_init __gst_debug_min gst_debug_log g_value_take_string g_value_array_append g_value_unset g_list_free __stack_chk_fail gst_object_get_type g_type_check_class_cast g_static_rec_mutex_lock g_static_rec_mutex_unlock g_object_class_find_property g_return_if_fail_warning g_object_ref snd_mixer_selem_get_enum_item snd_strerror g_list_nth_data strcmp snd_mixer_selem_set_enum_item snd_mixer_selem_set_capture_volume snd_mixer_selem_set_capture_switch_all g_type_check_instance_is_a snd_mixer_selem_set_playback_volume snd_mixer_selem_set_playback_switch_all snd_mixer_selem_set_playback_volume_all snd_mixer_selem_set_capture_volume_all snd_mixer_selem_get_playback_volume snd_mixer_selem_get_capture_volume snd_mixer_get_count snd_mixer_first_elem snd_mixer_selem_has_playback_volume snd_mixer_selem_get_name snd_mixer_elem_next snd_mixer_selem_has_playback_switch snd_mixer_selem_is_playback_mono snd_mixer_elem_set_callback snd_mixer_elem_set_callback_private snd_mixer_selem_has_capture_switch snd_mixer_selem_has_common_switch snd_mixer_selem_has_capture_volume snd_mixer_selem_has_common_volume snd_mixer_selem_is_enumerated gst_mixer_options_get_type gst_mixer_option_changed g_malloc_n memcpy gst_mixer_record_toggled gst_mixer_mute_toggled g_free gst_mixer_volume_changed snd_mixer_elem_get_callback_private write gst_task_join gst_object_unref g_static_rec_mutex_free g_object_unref g_list_foreach snd_mixer_close g_malloc0 pipe g_malloc g_static_rec_mutex_init gst_task_create gst_task_set_lock snd_mixer_open snd_mixer_attach snd_mixer_selem_register snd_mixer_load snd_mixer_set_callback_private snd_mixer_set_callback snd_ctl_card_info_get_name gst_task_start snd_mixer_get_callback_private snd_mixer_handle_events snd_mixer_poll_descriptors_count snd_mixer_poll_descriptors poll snd_mixer_poll_descriptors_revents g_assertion_message_expr _gst_element_error_printf gst_resource_error_quark gst_element_message_full gst_task_pause __errno_location g_strerror g_once_init_enter_impl g_intern_static_string gst_type_register_static_full gst_implements_interface_get_type gst_mixer_get_type g_once_init_leave g_type_class_peek_parent g_param_spec_string g_object_class_install_property _gst_debug_register_funcptr gst_implements_interface_cast g_threads_got_initialized g_value_dup_string g_thread_functions_for_glib_use g_value_set_string gst_element_class_set_details_simple snd_mixer_selem_get_playback_switch snd_mixer_selem_get_capture_switch gst_mixer_track_get_type g_type_register_static snd_mixer_selem_get_index g_object_new snd_mixer_selem_has_capture_channel g_utf8_collate dcgettext snd_mixer_selem_get_playback_volume_range snd_mixer_selem_has_playback_channel snd_mixer_selem_get_capture_volume_range snd_mixer_selem_has_capture_switch_exclusive snd_mixer_selem_get_capture_group g_assertion_message snd_mixer_selem_get_enum_items snd_mixer_selem_get_enum_item_name gst_element_register bindtextdomain bind_textdomain_codeset snd_lib_error_set_handler _gst_debug_category_new g_strdup_vprintf gst_plugin_desc gst_audio_sink_get_type snd_pcm_drop snd_pcm_prepare snd_pcm_delay snd_pcm_writei snd_pcm_frames_to_bytes snd_pcm_wait g_usleep snd_pcm_resume snd_pcm_close gst_caps_replace snd_pcm_hw_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_pcm_hw_params_malloc snd_pcm_format_name 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_period_time_near snd_pcm_hw_params_any snd_pcm_hw_params_set_access snd_pcm_build_linear_format snd_pcm_hw_params_set_buffer_time_near snd_pcm_hw_params_free snd_pcm_hw_params snd_pcm_hw_params_get_buffer_size snd_pcm_hw_params_get_period_size snd_output_buffer_open snd_pcm_dump_hw_setup snd_output_buffer_string snd_output_close snd_pcm_dump_sw_setup snd_pcm_hw_params_set_buffer_size_near snd_pcm_hw_params_set_period_size_near snd_pcm_open gst_caps_ref gst_element_class_get_pad_template gst_pad_template_get_caps g_thread_use_default_impl g_static_mutex_get_mutex_impl gst_static_pad_template_get gst_element_class_add_pad_template stdout snd_output_stdio_attach gst_audio_src_get_type snd_pcm_readi snd_pcm_nonblock strchr strtol snd_pcm_info_get_name snd_pcm_info snd_ctl_card_info_get_card snd_card_get_name gst_structure_copy gst_structure_remove_field __sprintf_chk snd_pcm_stream gst_caps_copy snd_pcm_format_mask_malloc snd_pcm_hw_params_get_format_mask gst_caps_new_empty gst_caps_get_size gst_caps_get_structure gst_structure_has_name gst_structure_get_int snd_pcm_format_mask_test gst_structure_set gst_caps_append_structure snd_pcm_format_mask_free gst_caps_unref snd_pcm_hw_params_get_rate_min snd_pcm_hw_params_get_rate_max gst_caps_make_writable gst_int_range_get_type snd_pcm_hw_params_get_channels_min snd_pcm_hw_params_get_channels_max gst_structure_get_field_type gst_audio_set_channel_positions gst_structure_get_value gst_value_get_int_range_min gst_value_get_int_range_max gst_caps_new_simple gst_caps_append libgstinterfaces-0.10.so.0 libgstaudio-0.10.so.0 libgstbase-0.10.so.0 libgstreamer-0.10.so.0 libgobject-2.0.so.0 libgmodule-2.0.so.0 libgthread-2.0.so.0 librt.so.1 libxml2.so.2 libglib-2.0.so.0 libasound.so.2 libpthread.so.0 libc.so.6 libgstalsa.so /usr/lib64 GLIBC_2.3.4 GLIBC_2.4 GLIBC_2.2.5 ALSA_0.9.0rc4 ALSA_0.9 @ ti ii
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DŽ$ [ E1 A Aa A A AM܃[Ll$X KD9LH5H E1Hź H1*ILH?H|$(HB~A AMA9 Lc @ LGSHHT HH9uLIEH5 E1ٺ IH1HHL>H|$(Lhandle != NULL Master Front PCM Speaker gstalsamixer.c MASTER [%s] is enumerated (%d) Updating track %P ALSA elem cb stop Cannot join alsamixer task Cannot open CTL: %s Cannot get card info: %s (NULL) Card name = %s Unknown Cannot find card name Cannot open mixer: %s device != NULL mixer->handle == NULL ALSA cb task loop system error: %s alsa error: %s Handling events rnfds <= nfds mixer != NULL && mixer->interface == NULL [%s] probing element #%u, mixer->dir=%u [%s] PLAYBACK: has_playback_volume=%d, has_playback_switch=%d%s [%s] CAPTURE: has_capture_volume=%d, has_common_volume=%d, has_capture_switch=%d, has_common_switch=%d, play_track=%p Cannot send update notifications, no GstMixer * given Cannot send stop to alsamixer task Cannot open mixer for sound device '%s': %s Cannot register mixer elements: %s Cannot load mixer settings: %s Successfully opened mixer for device '%s'. Could not start alsamixer task snd_mixer_poll_descriptors_count <= 0: %d _gst_alsa_mixer_set_interface gst_alsa_mixer_get_mixer_flags gst_alsa_mixer_get_option gst_alsa_mixer_set_option gst_alsa_mixer_set_record gst_alsa_mixer_set_mute gst_alsa_mixer_set_volume gst_alsa_mixer_get_volume gst_alsa_mixer_list_tracks gst_alsa_mixer_ensure_track_list gst_alsa_mixer_elem_handle_callback gst_alsa_mixer_update gst_alsa_mixer_update_track gst_alsa_mixer_update_option gst_alsa_mixer_free gst_alsa_mixer_new gst_alsa_mixer_open gst_alsa_mixer_handle_callback task_monitor_alsa GstAlsaMixerElement gstalsamixerelement.c iface_type == GST_TYPE_MIXER this != NULL this->mixer != NULL default Device Device name device-name gstalsamixerelement.c:166 gstalsamixerelement.c:195 Generic/Audio Alsa mixer file %s: line %d (%s): should not be reached ALSA device, as defined in an asound configuration file Human-readable name of the sound device gst_alsa_mixer_element_change_state Failed to open alsa mixer device '%s' Leif Johnson Control sound input and output levels with ALSA gst_alsa_mixer_element_interface_supported gst_alsa_mixer_element_supported gst_alsa_mixer_element_get_mixer_flags gst_alsa_mixer_element_get_option gst_alsa_mixer_element_set_option gst_alsa_mixer_element_set_record gst_alsa_mixer_element_set_mute gst_alsa_mixer_element_get_volume gst_alsa_mixer_element_set_volume gst_alsa_mixer_element_list_tracks gst_alsa_mixer_element_change_state GstAlsaMixerTrack false true gstalsamixertrack.c index untranslated-label [%s] %d output channels [%s] %d input channels gst-plugins-base-0.10 Capture %s%s%s %d [%s,%u] num=%d,track_num=%d,flags=0x%08x,sw=%s,shared_mute_track=%p [%s] created new mixer track %p Mixer track must be either output or input! [%s] alsa_flags=0x%08x, capture_group=%d gst_alsa_mixer_track_new gst_alsa_mixer_track_update_alsa_capabilities GstAlsaMixerOptions gstalsamixeroptions.c [%s,%u] gst_alsa_mixer_options_new alsamixer alsasrc alsa plugins alsa gstalsaplugin.c /usr/share/locale UTF-8 : alsalib error: %s%s%s ALSA plugin library 0.10.29 LGPL gst-plugins-base binding text domain %s to locale dir %s failed to set alsa error handler Fedora gstreamer-plugins-base package http://download.fedora.redhat.com/fedora plugin_init GstAlsaSink gst_alsasink_getcaps gst_alsasink_open gst_alsasink_prepare gst_alsasink_unprepare gst_alsasink_close gst_alsasink_write gst_alsasink_delay gst_alsasink_reset gstalsasink.c 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 periods min %u, max %u warning: %s buffer time %u period time %u Hardware setup:
%s Software setup:
%s Error parsing spec Opened device %s Device '%s' is busy Returning cached caps returning caps %P pad_template != NULL gstalsasink.c:232 gstalsasink.c:255 Wim Taymans Sink/Audio Audio sink (ALSA) initializing alsasink 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 Can't recovery from underrun, prepare failed: %s Can't recovery from suspend, prepare failed: %s Unable to determine current swparams for playback: %s Unable to set start threshold mode for playback: %s Unable to set avail min for playback: %s Unable to set sw params for playback: %s Linear format : depth=%d, width=%d, sign=%d, bigend=%d Negotiating to %d channels @ %d Hz (format = %s) SPDIF (%d) falling back to little endian with swapping 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 buffer size %lu, period size %lu Broken configuration for playback: no configurations available: %s Access type not available for playback: %s Sample format not available for playback: %s 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. Rate %iHz not available for playback: %s Rate doesn't match (requested %iHz, get %iHz) Unable to get buffer size for playback: %s Unable to get period size for playback: %s Unable to set hw params for playback: %s Could not open IEC958 (SPDIF) device for playback Setting of hwparams failed: %s Setting of swparams failed: %s 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 Output to a sound card via ALSA audio/x-raw-int, endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, signed = (boolean) { TRUE, FALSE }, width = (int) 32, depth = (int) 32, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-raw-int, endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, signed = (boolean) { TRUE, FALSE }, width = (int) 24, depth = (int) 24, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-raw-int, endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, signed = (boolean) { TRUE, FALSE }, width = (int) 32, depth = (int) 24, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-raw-int, endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, signed = (boolean) { TRUE, FALSE }, width = (int) 16, depth = (int) 16, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-raw-int, signed = (boolean) { TRUE, FALSE }, width = (int) 8, depth = (int) 8, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ];audio/x-iec958 {{{{|gst_alsasink_reset gst_alsasink_delay gst_alsasink_write xrun_recovery gst_alsasink_prepare alsasink_parse_spec set_hwparams set_swparams gst_alsasink_open gst_alsasink_getcaps gst_alsasink_init GstAlsaSrc gstalsasrc.c gst_alsasrc_getcaps gst_alsasrc_open gst_alsasrc_prepare gst_alsasrc_unprepare gst_alsasrc_close gst_alsasrc_read gst_alsasrc_delay gst_alsasrc_reset Read error: %s initializing Prepare failed: %s gstalsasrc.c:248 gstalsasrc.c:271 Source/Audio Audio source (ALSA) interface_type == GST_TYPE_MIXER Broken configuration for recording: no configurations available: %s Access type not available for recording: %s Sample format not available 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 Rate %iHz not available for recording: %s Unable to set buffer time %i for recording: %s Unable to get buffer size for recording: %s Unable to set period time %i for recording: %s Unable to get period size for recording: %s Unable to set hw params for recording: %s Could not set device to blocking: %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 Read from a sound card via ALSA audio/x-raw-int, endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, signed = (boolean) { TRUE, FALSE }, width = (int) 32, depth = (int) 32, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-raw-int, endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, signed = (boolean) { TRUE, FALSE }, width = (int) 32, depth = (int) 24, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-raw-int, endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, signed = (boolean) { TRUE, FALSE }, width = (int) 24, depth = (int) 24, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-raw-int, endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, signed = (boolean) { TRUE, FALSE }, width = (int) 16, depth = (int) 16, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ]; audio/x-raw-int, signed = (boolean) { TRUE, FALSE }, width = (int) 8, depth = (int) 8, rate = (int) [ 1, MAX ], channels = (int) [ 1, MAX ] gst_alsasrc_interface_supported gst_alsasrc_mixer_supported gst_alsasrc_mixer_get_mixer_flags gst_alsasrc_mixer_get_option gst_alsasrc_mixer_set_option gst_alsasrc_mixer_set_record gst_alsasrc_mixer_set_mute gst_alsasrc_mixer_get_volume gst_alsasrc_mixer_set_volume gst_alsasrc_mixer_list_tracks gst_alsasrc_reset gst_alsasrc_delay gst_alsasrc_read xrun_recovery gst_alsasrc_prepare set_hwparams set_swparams gst_alsasrc_open gst_alsasrc_getcaps gst_alsasrc_init gstalsa.c [%s] device=%d name from pcminfo: %s no luck so far, trying backup Generated device string "%s" audio/x-raw-int skipping non-int format width depth signed endianness probing sample rates ... Min. rate = %u (%d) Max. rate = %u (%d) rate probing channels ... Min. channels = %d (%d) Max. channels = %d (%d) audio/x-iec958 failed to query formats: %s failed to query formats min_chans >= 1 Trying to get device name from string '%s' Trying to get device name from open handle Device name for device '%s': %s iec958:{AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x} failed opening IEC958 device: %s 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 gst_alsa_find_device_name gst_alsa_find_device_name_no_handle gst_alsa_probe_supported_formats "
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