Current File : //lib64/gstreamer-1.0/libgstopus.so
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J X m Z h J a h __gmon_start__ _ITM_deregisterTMCloneTable _ITM_registerTMCloneTable __cxa_finalize gst_element_register gst_tag_register_musicbrainz_tags gst_plugin_opus_get_desc gst_plugin_opus_register gst_plugin_register_static opus_multistream_decoder_destroy gst_mini_object_replace gst_audio_decoder_set_needs_format gst_audio_decoder_set_use_default_pad_acceptcaps g_type_class_peek_parent _gst_debug_register_funcptr gst_element_class_add_static_pad_template gst_element_class_set_static_metadata g_param_spec_boolean g_object_class_install_property g_type_class_adjust_private_offset _gst_debug_category_new gst_caps_get_size gst_caps_get_structure gst_structure_get_int _gst_int_range_type g_value_init gst_value_set_int_range gst_structure_set_value g_value_unset __stack_chk_fail g_value_set_int gst_value_list_append_value _gst_value_list_type gst_mini_object_copy gst_audio_decoder_proxy_getcaps gst_mini_object_unref gst_mini_object_make_writable gst_caps_intersect gst_pad_get_allowed_caps gst_caps_from_string gst_caps_set_simple gst_caps_is_empty _gst_debug_min gst_debug_log gst_caps_truncate gst_structure_fixate_field_nearest_int gst_audio_channel_positions_to_valid_order gst_audio_info_init gst_audio_info_set_format gst_audio_decoder_set_output_format gst_codec_utils_opus_parse_header pow _gst_element_error_printf gst_stream_error_quark gst_element_message_full gst_pad_get_current_caps gst_caps_is_equal gst_structure_get_value _gst_value_array_type g_type_check_value_holds gst_value_array_get_size gst_codec_utils_opus_parse_caps gst_value_array_get_value _gst_buffer_type g_value_get_boxed gst_buffer_get_size gst_audio_decoder_allocate_output_buffer gst_buffer_map opus_multistream_decode gst_buffer_unmap gst_buffer_set_size gst_audio_clipping_meta_api_get_type gst_buffer_get_meta g_assertion_message_expr gst_audio_decoder_finish_frame opus_multistream_decoder_create opus_multistream_decoder_ctl opus_strerror gst_util_uint64_scale_int gst_flow_get_name gst_audio_buffer_reorder_channels gst_buffer_resize _gst_audio_decoder_error gst_library_error_quark gst_buffer_memcmp gst_audio_decoder_set_plc_aware gst_audio_decoder_set_latency g_type_name g_value_set_boolean g_value_get_boolean g_once_init_enter g_intern_static_string gst_audio_decoder_get_type g_type_register_static_simple g_once_init_leave g_mutex_init g_mutex_clear gst_audio_encoder_proxy_getcaps gst_caps_new_empty gst_structure_new gst_structure_copy gst_structure_set gst_caps_append_structure _gst_bitmask_type gst_structure_free gst_event_type_get_name gst_tag_setter_get_tag_merge_mode gst_event_parse_tag gst_tag_setter_merge_tags g_mutex_lock g_mutex_unlock g_malloc0 memcpy gst_audio_encoder_allocate_output_buffer opus_multistream_encode gst_audio_encoder_finish_frame gst_buffer_add_audio_clipping_meta gst_util_uint64_scale gst_audio_encoder_set_latency gst_audio_encoder_set_frame_samples_min gst_audio_encoder_set_frame_samples_max gst_audio_encoder_set_frame_max opus_multistream_encoder_destroy gst_tag_setter_reset_tags g_value_set_uint g_value_set_enum g_value_get_uint g_value_get_enum g_value_get_int opus_multistream_encoder_ctl g_param_spec_enum g_param_spec_int g_param_spec_uint g_enum_register_static opus_multistream_encoder_create gst_codec_utils_opus_create_header gst_tag_setter_get_tag_list gst_tag_list_to_vorbiscomment_buffer gst_codec_utils_opus_create_caps_from_header gst_audio_encoder_set_output_format gst_tag_list_new_empty gst_tag_list_add gst_audio_encoder_merge_tags gst_audio_encoder_get_type gst_tag_setter_get_type g_type_add_interface_static gst_preset_get_type gst_debug_category_get_threshold g_string_new g_string_append_printf g_string_append g_string_free libgstpbutils-1.0.so.0 libgstvideo-1.0.so.0 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 libopus.so.0 libpthread.so.0 libc.so.6 _edata __bss_start _end libgstopus.so GLIBC_2.2.5 GLIBC_2.14 GLIBC_2.4 ui P ii
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LPLL$(H 1kH HH߾ []A\A]A^A_ HH opusenc opusdec 1.16.1 OPUS plugin library http://www.redhat.com gst-plugins-base LGPL 2019-09-23 GStreamer-plugins-base package gst_opus_dec_start gst_opus_dec_stop gst_opus_dec_handle_frame gst_opus_dec_set_format gst_opus_dec_getcaps decode opus streams to audio Codec/Decoder/Audio Opus audio decoder Use in-band FEC use-inband-fec Apply gain apply-gain Control Phase Inversion phase-inversion opus decoding element audio/x-raw gstopusdec.c Using a default of 2 channels Negotiated %d channels, %d Hz error: %s warning: %s Channel mapping family %d set_format: %pA caps didn't change Mapping table %d streams, %d stereo Failed to create Opus decoder Received buffer of size %lu Received missing buffer Using buffer of size %lu Using NULL buffer FEC disabled, decoding buffer Decoding error (%d): %s decoded %d samples Applying gain: volume %f flow: %s found streamheader found vorbiscomments OpusHead OpusTags counted vorbiscomments property GstOpusDec sink src Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> Use forward error correction if available (needs PLC enabled) Apply gain if any is specified in the header Set to true to enable phase inversion, this will slightly improve stereo quality, but will have side effects when downmixed to mono. Empty intersection, failed to negotiate Using a default of 48kHz sample rate Header is not an Opus ID header Failed to parse Opus ID header Found pre-skip of %u samples, R128 gain %d (volume %f) Channel mapping family 1, Vorbis mapping Using NONE channel layout for more than 8 channels caps have changed, resetting decoder No header, assuming single stream Creating decoder with %d channels, %d Hz Failed to create Opus decoder (%d): %s Could not configure phase inversion: %s First buffer received in FEC mode, early out Missing duration, using last duration %u:%02u:%02u.%09u Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms current duration %u:%02u:%02u.%09u of missing data not enough for PLC (minimum needed: %u:%02u:%02u.%09u) - skipping calculated PLC frame length: %u:%02u:%02u.%09u num frame samples: %d new leftover: %u:%02u:%02u.%09u Code 3 Opus packet has less than 2 bytes Opus packet duration > 120 ms, invalid Opus packet: frame size %.1f ms, %d frames, duration %.1f ms FEC enabled, decoding last delayed buffer FEC enabled, reconstructing last buffer !cmeta || cmeta->format == GST_FORMAT_DEFAULT Skipping %u samples at the beginning (%u at 48000 Hz) Skipping %u samples at the end (%u at 48000 Hz) Failed to create %u byte buffer missing buffer, doing PLC duration %u:%02u:%02u.%09u plus leftover %u:%02u:%02u.%09u Got buffer ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u %s:%d: invalid %s id %u for "%s" of type '%s' in '%s' audio/x-opus, channel-mapping-family = (int) 0; audio/x-opus, channel-mapping-family = (int) [1, 255], channels = (int) [1, 255], stream-count = (int) [1, 255], coupled-count = (int) [0, 255] audio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) { 48000, 24000, 16000, 12000, 8000 }, channels = (int) [ 1, 8 ] packet_duration_opus @ @ @ x x x x opus_dec_chain_parse_data opus_dec_chain_parse_data gst_opus_dec_handle_frame gst_opus_dec_negotiate gst_opus_dec_parse_header gst_opus_dec_set_format p? 4@ $@ ? @B $tIgstopusenc.c Unsupported frame size: %d init sink getcaps layout S16LE interleaved channel-mask Returning caps: %pA sink event: %s bsize < bytes draining %d samples nothing to drain size == bytes Encoding failed (%d): %s Output packet is %u bytes Latency: %u:%02u:%02u.%09u Encodes audio in Opus format Codec/Encoder/Audio Opus audio encoder gst_opus_enc_start gst_opus_enc_stop gst_opus_enc_set_format gst_opus_enc_handle_frame gst_opus_enc_sink_event gst_opus_enc_sink_getcaps GstOpusEncAudioType audio-type Encoding Bit-rate bitrate GstOpusEncBandwidth Audio Band Width bandwidth GstOpusEncFrameSize Frame Size frame-size GstOpusEncBitrateType Bitrate type bitrate-type Complexity complexity In-band FEC DTX dtx Packet loss percentage Loss percentage packet-loss-percentage Maximum payload size in bytes Max payload size max-payload-size gst_opus_enc_finalize Opus encoder Mono, trivial RTP mapping Stereo, trivial RTP mapping Encoding mapping table Decoding mapping table Unknown mapping Setup with %d channels, %d Hz here are the caps: %pA Encoder creation failed GstOpusEnc CBR cbr Constrained VBR constrained-vbr 2.5 10 20 40 60 Narrow band narrowband Medium band mediumband Wide band Super wide band superwideband Full band fullband Auto auto Generic audio generic Voice voice audio/x-opus p н H 4Ld|Pp0 ` gst_opus_enc_start gst_opus_enc_stop gst_opus_enc_get_latency gst_opus_enc_get_frame_samples gst_opus_enc_setup gst_opus_enc_find_channel_position_in_vorbis_order gst_opus_enc_setup_channel_mappings
gst_opus_enc_setup_channel_mappings gst_opus_enc_set_format gst_opus_enc_encode gst_opus_enc_encode gst_opus_enc_handle_frame gst_opus_enc_sink_event gst_opus_enc_sink_getcaps gst_opus_enc_init received buffer %p of %lu bytes draining; adding silence samples %li extra samples of padding in this frame Need to add %li extra samples in the next frame encoding %d samples (%d bytes) Adding trim-start %lu trim-end %lu Encoded size %d is higher than max payload size (%d bytes) What type of audio to optimize for Specify an encoding bit-rate (in bps). The duration of an audio frame, in ms Enable forward error correction Channel position %s maps to index %d in Vorbis order Channel position %s is not representable in Vorbis order Setting up channel mapping for %d channels In range for the Vorbis mapping, building channel mapping tables Natural pair '%s/%s' found at %d %d Cannot map channel positions to Vorbis order, using unknown mapping Channel position %s is not mapped yet, adding Mapping tables built: %d channels, %d stereo streams Unpositioned mapping, all channels mono setup: %d Hz, %d channels, %d stereo streams, family %d we have frame size %d, lookahead %d Encoded with GStreamer opusenc audio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) 48000, channels = (int) [ 1, 8 ]; audio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) { 8000, 12000, 16000, 24000 }, channels = (int) [ 1, 8 ] [ %d ] gstopuscommon.c %s: %s 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 none gst_opus_common_log_channel_mapping_table
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