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B py j 9 z u t @ ( x ] y В \ J H __gmon_start__ __cxa_finalize _Jv_RegisterClasses gst_audio_buffer_clip g_return_if_fail_warning gst_buffer_get_type g_type_check_instance_is_a gst_segment_clip gst_util_uint64_scale gst_buffer_make_metadata_writable 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_assertion_message_expr gst_caps_get_structure gst_structure_get_int gst_audio_frame_byte_size gst_audio_is_buffer_framed gst_audio_frame_length g_thread_functions_for_glib_use gst_ring_buffer_debug_spec_buff __gst_debug_min gst_debug_log gst_ring_buffer_debug_spec_caps g_atomic_int_compare_and_exchange g_threads_got_initialized gst_ring_buffer_convert gst_format_get_name gst_util_uint64_scale_int gst_ring_buffer_parse_caps gst_structure_get_name gst_caps_replace gst_structure_get_boolean gst_ring_buffer_get_type gst_object_get_type g_type_register_static _gst_debug_category_new gst_ring_buffer_may_start gst_ring_buffer_clear memcpy gst_ring_buffer_advance g_atomic_int_add gst_ring_buffer_prepare_read gst_ring_buffer_commit_full gst_ring_buffer_commit gst_ring_buffer_clear_all gst_ring_buffer_set_sample gst_ring_buffer_samples_done gst_ring_buffer_stop gst_ring_buffer_pause gst_ring_buffer_start gst_ring_buffer_read gst_ring_buffer_set_flushing gst_ring_buffer_is_active gst_ring_buffer_activate gst_ring_buffer_is_acquired gst_ring_buffer_delay gst_ring_buffer_release g_free gst_ring_buffer_acquire g_malloc gst_ring_buffer_device_is_open gst_ring_buffer_close_device gst_ring_buffer_open_device gst_ring_buffer_set_callback g_type_class_peek_parent _gst_debug_register_funcptr g_type_check_instance_cast g_type_check_class_cast gst_audio_clock_adjust gst_audio_clock_get_time gst_audio_clock_reset gst_audio_clock_get_type gst_system_clock_get_type gst_audio_clock_new g_object_new gst_element_set_state gst_element_get_state GST_CAT_DEFAULT gst_mixer_get_type gst_implements_interface_check gst_implements_interface_cast gst_mixer_list_tracks 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_element_factory_get_longname gst_property_probe_get_type gst_object_unref gst_property_probe_get_property gst_property_probe_probe_and_get_values g_value_array_get_nth g_object_set_property g_value_get_string g_value_array_free gst_plugin_feature_list_free g_list_reverse 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_n g_value_get_enum gst_audio_set_channel_positions gst_audio_get_channel_positions g_memdup gst_base_audio_sink_create_ringbuffer gst_object_set_parent gst_base_audio_sink_get_type g_once_init_enter_impl g_intern_static_string gst_base_sink_get_type gst_type_register_static_full g_once_init_leave 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_sink_wait_eos g_static_rec_mutex_lock gst_pad_pull_range gst_flow_get_name g_static_rec_mutex_unlock gst_base_sink_do_preroll gst_message_new_eos gst_element_post_message gst_structure_fixate_field_nearest_int gst_structure_has_field gst_structure_fixate_field_boolean gst_clock_get_time _gst_element_error_printf gst_stream_error_quark gst_element_get_type gst_element_message_full gst_clock_get_calibration gst_base_sink_wait_clock gst_base_sink_wait_preroll gst_base_sink_get_latency gst_base_sink_get_ts_offset gst_base_sink_get_render_delay gst_segment_to_running_time gst_clock_set_calibration __stack_chk_fail gst_clock_set_master gst_event_parse_new_segment_full gst_query_parse_convert gst_query_set_convert gst_base_sink_query_latency gst_query_set_latency gst_object_ref gst_object_unparent g_value_set_int64 g_value_get_int64 g_value_get_boolean g_type_instance_get_private gst_pad_set_query_function gst_plugin_feature_get_name gst_plugin_feature_check_version gst_object_get_parent gst_base_audio_sink_slave_method_get_type g_enum_register_static g_type_class_add_private g_param_spec_int64 g_object_class_install_property g_param_spec_boolean g_param_spec_enum g_type_class_ref gst_base_audio_src_create_ringbuffer gst_base_audio_src_get_type gst_push_src_get_type bindtextdomain bind_textdomain_codeset 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_buffer_new_and_alloc gst_base_src_wait_playing dcgettext gst_core_error_quark gst_resource_error_quark gst_base_src_get_type g_object_notify gst_base_src_set_live gst_base_src_set_format gst_base_audio_src_slave_method_get_type gst_audio_filter_get_type gst_base_transform_get_type g_type_register_static_simple gst_audio_filter_class_add_pad_templates g_type_check_class_is_a gst_caps_get_type gst_caps_copy gst_pad_template_new gst_element_class_add_pad_template gst_audio_sink_get_type g_thread_create_full g_thread_join g_value_set_pointer gst_message_new_stream_status gst_message_set_stream_status_object __errno_location g_strerror _gst_debug_nameof_funcptr memset 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GST_IS_BUFFER (buffer) structure != NULL rate channels endianness width depth signed (NULL) '' GST_IS_BUFFER (buf) audio.c bytes != 0 width != 0 channels != 0 rate != 0 gst_audio_buffer_clip _gst_audio_structure_set_list gst_audio_duration_from_pad_buffer gstringbuffer.c spec caps: %p %P parsed caps: width: %d parsed caps: depth: %d parsed caps: sign: %d parsed caps: bigend: %d parsed caps: rate: %d parsed caps: channels: %d parsed caps: sample bytes: %d silence byte %d: %02x pausing ringbuffer signal waiter failed to pause was not started no rate or bps configured ret=%d result %li audio/x-raw-int audio/x-raw-float audio/x-alaw audio/x-mulaw audio/x-iec958 audio/x-ac3 could not parse caps spec->latency_time != 0 GstRingBuffer ringbuffer class may start: %d GST_IS_RING_BUFFER (buf) clear segment %d @%p clear all segments buf->data != NULL segment != NULL readptr != NULL len != NULL set sample to %lu, segbase %d stopping was not started, try paused failed to stop we are flushing starting ringbuffer was not stopped, try paused resuming failed to start we are not acquired start! waiting.. stopped processing not allowed to start copy %u bytes fwd_up end %d/%d fwd_down end %d/%d rev_up end %d/%d rev_down end %d/%d device not acquired Device for %p not acquired device was active in mode %d failed to activate device releasing device released device device was released failed to release device buf->open == TRUE acquiring device acquired device device not opened Device for %p not opened device was acquired failed to acquire device closed device error closing device opened device failed opening device !buf->acquired default_clear_all default_commit linear mu law a law ima adpcm mpeg gsm eac3 dts unknown s8 u8 s16_le s16_be u16_le u16_be s24_le s24_be u24_le u24_be s32_le s32_be u32_le u32_be s24_3le s24_3be u24_3le u24_3be s20_3le s20_3be u20_3le u20_3be s18_3le s18_3be u18_3le u18_3be float32_le float32_be float64_le float64_be mu_law a_law ima_adpcm acquire ringbuffer: buffer time: %li usec acquire ringbuffer: latency time: %li usec acquire ringbuffer: total segments: %d acquire ringbuffer: latency segments: %d acquire ringbuffer: segment size: %d bytes = %d samples acquire ringbuffer: buffer size: %d bytes = %d samples parsed caps: type: %d, '%s' parsed caps: format: %d, '%s' converting value %li from %s (%d) to %s (%d) prepare read from segment %d (real %d) @%p was not paused, must have been stopped was not paused, must have been started pointer at %d, write to %d-%d, diff %d, segtotal %d, segsize %d, base %d write @%p seg %d, sps %d, off %d, avail %d pointer at %d, sample %lu, read from %d-%d, to_read %d, diff %d, segtotal %d, segsize %d read @%p seg %d, off %d, sampleslen %d invalid bytes_per_sample from acquire ringbuffer, fix the element Device for ring buffer already closed Device for ring buffer %p already closed, fix your code Resources for ring buffer still acquired Resources for ring buffer %p still acquired Device for ring buffer already open Device for ring buffer %p already open, fix your code 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_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_active gst_ring_buffer_activate gst_ring_buffer_activate 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_convert gst_ring_buffer_parse_caps gst_ring_buffer_parse_caps
gst_ring_buffer_debug_spec_buff gst_ring_buffer_debug_spec_caps default_commit default_commit default_clear_all gstaudioclock.c no time, reuse last init GstAudioClock name audioclock result %u:%02u:%02u.%09u, last_time %u:%02u:%02u.%09u reset clock to %u:%02u:%02u.%09u, last %u:%02u:%02u.%09u, offset %u:%02u:%02u.%09u gst_audio_clock_get_time gst_audio_clock_reset gst_audio_clock_init gst_audio_clock_get_internal_time mixerutils.c element is not a mixer Generic/Audio probing %s ... created element %s (%p) probing available devices ... trying device %s .. 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 gst_audio_default_registry_mixer_filter gst_audio_mixer_filter_probe_feature gst_audio_mixer_filter_check_element channel-positions str != NULL num_positions > 0 pos != NULL res channels > 0 caps != NULL 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 Pre-fixated on both %d/%d and %d - conflict! Pre-fixated one side, but other side n/a of %d/%d %d unfixed channel positions left after fixation! gst_value_array_get_size (pos_val_arr) == channels t == GST_TYPE_LIST || t == GST_TYPE_AUDIO_CHANNEL_POSITION 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 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 GstBaseAudioSink baseaudiosink element baseaudiosink GST_IS_BASE_AUDIO_SINK (sink) gstbaseaudiosink.c deactivating pull drained audio EOS error %s ringbuffer may start now release old ringbuffer time was %u:%02u:%02u.%09u activate ringbuffer cannot parse audio format. error: %s could not acquire ringbuffer ringbuffer in wrong state sink not negotiated. checking preroll latency synced late preroll, adding jitter Setting clock as master we have no clock we are not slaved correct clock skew %li > %li correct clock skew %li < %li unknown slaving method %d time offset %u:%02u:%02u.%09u resync after discont rendering at %lu %d/%d wrote %u of %u next sample expected at %lu ringbuffer not negotiated wrong size failed waiting for latency new segment rate of %f latency query query convert ringbuffer not acquired clock provide disabled open failed property gstbaseaudiosink.c:672 gstbaseaudiosink.c:639 GstAudioSinkClock gst_base_audio_sink_query_pad created from factory %p pulsesink disable time offset GstBaseAudioSinkSlaveMethod Audio latency in microseconds Provide Clock provide-clock Slave Method slave-method Allow pull-based scheduling Allow Pull Scheduling can-activate-pull Drift Tolerance drift-tolerance 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 Resampling slaving resample Skew slaving skew No slaving none last sample time %u:%02u:%02u.%09u pulling %d bytes offset %lu to fill audio buffer got different size than requested from sink pad: %u != %u Got flow '%s' but can't return it: %d 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 %lu, start %u:%02u:%02u.%09u, samples %u Buffer of size %u has no time. Using render_start=%lu sync-offset %li, render-delay %u:%02u:%02u.%09u, ts-offset %li clipping start to %u:%02u:%02u.%09u %lu samples clipping stop to %u:%02u:%02u.%09u %lu samples no sync needed. Using render_start=%lu running: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u compensating for sync-offset %u:%02u:%02u.%09u adding base_time %u:%02u:%02u.%09u after compensation: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u internal %u:%02u:%02u.%09u external %u:%02u:%02u.%09u %lu/%lu = %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 %li avg %li last_align %li driftsamples %u, next %lu final timestamps: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u no align possible: no previous sample position known align with prev sample, ABS (%li) < %li discont timestamp (%li) >= %li 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. preroll got interrupted: %d (%s) 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' processed samples: raw %lu, delay %u, real %lu, time %u:%02u:%02u.%09u 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 Tolerance for timestamp and clock drift in microseconds gst_base_audio_sink_change_state gst_base_audio_sink_provide_clock gst_base_audio_sink_async_play gst_base_audio_sink_activate_pull nllllllllollllllmllllllllPncp q0q@qhqqpq|rrrr sdrgst_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_activate_pull 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_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 gst_base_audio_sink_init gst_base_audio_sink_query_pad gst_base_audio_sink_get_time ? GstBaseAudioSrc baseaudiosrc element baseaudiosrc gstbaseaudiosrc.c /usr/share/locale gst-plugins-base-0.10 UTF-8 GST_IS_BASE_AUDIO_SRC (src) at segment %d and sample %lu dropped, align to segment %d reading from sample %lu read %u of %u wait playing continue playing warning: %s ringbuffer stopped flush-start flush-stop refuse to seek dropping event %p acquire new ringbuffer actual-buffer-time actual-latency-time NULL->READY READY->PAUSED PAUSED->PLAYING PLAYING->PAUSED PAUSED->READY READY->NULL gstbaseaudiosrc.c:515 gstbaseaudiosrc.c:472 GstAudioSrcClock GstBaseAudioSrcSlaveMethod Actual Buffer Time Actual Latency Time 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 Re-timestamp re-timestamp Skew binding text domain %s to locale dir %s first sample, align to current %d reading from %d, we are at %d, sample %lu create DISCONT of %lu samples at sample %lu Can't record audio fast enough Dropped %lu samples. This is most likely because downstream can't keep up and is consuming samples too slowly.
running_time = %u:%02u:%02u.%09u
timestamp = %u:%02u:%02u.%09u
running_time_segment = %d
last_written_segment = %d
segment_skew (running time segment - last_written_segment) = %d
last_read_segment = %d Timeshifted the ringbuffer with %d segments: Updating the timestamp to %u:%02u:%02u.%09u, and src->next_sample to %lu buffer timestamp %u:%02u:%02u.%09u (base_time %u:%02u:%02u.%09u) buffer timestamp 0, ts %u:%02u:%02u.%09u <= base_time %u:%02u:%02u.%09u resource can only be operated on sequentially but offset %lu was given report latency min %u:%02u:%02u.%09u max %u:%02u:%02u.%09u 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 `0`p(Xh( 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 _do_init gst_base_audio_src_create gst_base_audio_src_get_offset gst_base_audio_src_query gst_base_audio_src_event gst_base_audio_src_setcaps gst_base_audio_src_provide_clock gst_base_audio_src_change_state gst_base_audio_src_get_time GstAudioFilter audiofilter 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_audio_filter_class_add_pad_templates gst_audio_filter_set_caps GstAudioSink gst_audioringbuffer_acquire gst_audioringbuffer_release gst_audioringbuffer_start gst_audioringbuffer_pause gst_audioringbuffer_stop gst_audioringbuffer_delay gst_audioringbuffer_activate GstAudioSinkRingBuffer gstaudiosink.c reset... reset done start, sending signal could not close device could not open device starting thread waiting for thread thread is started signal wait could not create thread %s enter thread posting ENTER stream status wait for action got signal continue running stop running, exit thread posting LEAVE stream status unprepared could not unprepare device could not prepare device creating ringbuffer created ringbuffer @%p gst_audioringbuffer_open_device gst_audioringbuffer_close_device gst_audio_sink_create_ringbuffer could not create thread for unknown reason transfered %d bytes of %d from segment %d error writing data in %s (reason: %s), skipping segment (left: %d, written: %d) no write function, exit thread gst_audioringbuffer_activate audioringbuffer_thread_func gst_audioringbuffer_stop gst_audioringbuffer_pause gst_audioringbuffer_start gst_audioringbuffer_release gst_audioringbuffer_acquire gst_audioringbuffer_close_device gst_audioringbuffer_open_device gst_audio_sink_create_ringbuffer GstAudioSrc GstAudioSrcRingBuffer gstaudiosrc.c gst_audio_src_create_ringbuffer transfered %d bytes of %d to segment %d error reading data %d (reason: %s), skipping segment gst_audioringbuffer_stop gst_audioringbuffer_start audioringbuffer_thread_func gst_audio_src_create_ringbuffer GstRingBufferState GstAudioChannelPosition GstBufferFormat GstBufferFormatType GstRingBufferSegState GST_UNKNOWN GST_S8 GST_U8 GST_S16_LE s16-le GST_S16_BE s16-be GST_U16_LE u16-le GST_U16_BE u16-be GST_S24_LE s24-le GST_S24_BE s24-be GST_U24_LE u24-le GST_U24_BE u24-be GST_S32_LE s32-le GST_S32_BE s32-be GST_U32_LE u32-le GST_U32_BE u32-be GST_S24_3LE s24-3le GST_S24_3BE s24-3be GST_U24_3LE u24-3le GST_U24_3BE u24-3be GST_S20_3LE s20-3le GST_S20_3BE s20-3be GST_U20_3LE u20-3le GST_U20_3BE u20-3be GST_S18_3LE s18-3le GST_S18_3BE s18-3be GST_U18_3LE u18-3le GST_U18_3BE u18-3be GST_FLOAT32_LE float32-le GST_FLOAT32_BE float32-be GST_FLOAT64_LE float64-le GST_FLOAT64_BE float64-be GST_MU_LAW mu-law GST_A_LAW a-law GST_IMA_ADPCM ima-adpcm GST_MPEG GST_GSM GST_IEC958 GST_AC3 GST_EAC3 GST_DTS GST_BUFTYPE_LINEAR GST_BUFTYPE_FLOAT GST_BUFTYPE_MU_LAW GST_BUFTYPE_A_LAW GST_BUFTYPE_IMA_ADPCM GST_BUFTYPE_MPEG GST_BUFTYPE_GSM GST_BUFTYPE_IEC958 GST_BUFTYPE_AC3 GST_BUFTYPE_EAC3 GST_BUFTYPE_DTS GST_SEGSTATE_INVALID invalid GST_SEGSTATE_EMPTY empty GST_SEGSTATE_FILLED filled GST_SEGSTATE_PARTIAL partial GST_RING_BUFFER_STATE_STOPPED GST_RING_BUFFER_STATE_PAUSED GST_RING_BUFFER_STATE_STARTED front-mono front-left front-right rear-center rear-left rear-right lfe front-center front-left-of-center front-right-of-center side-left side-right num GST_AUDIO_CHANNEL_POSITION_INVALID GST_AUDIO_CHANNEL_POSITION_FRONT_MONO GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT GST_AUDIO_CHANNEL_POSITION_REAR_CENTER GST_AUDIO_CHANNEL_POSITION_REAR_LEFT GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT GST_AUDIO_CHANNEL_POSITION_LFE GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT GST_AUDIO_CHANNEL_POSITION_NONE GST_AUDIO_CHANNEL_POSITION_NUM ;< X P @ А@ p Б p@ 0h P 0 P PX @ P ` ( X P `
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