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gst_audio_stream_align_get_discont_wait gst_audio_stream_align_mark_discont gst_audio_stream_align_new gst_audio_stream_align_set_rate gst_audio_stream_align_get_timestamp_at_discont gst_audio_stream_align_get_samples_since_discont gst_audio_stream_align_process orc_once_mutex_lock orc_once_mutex_unlock orc_program_new_from_static_bytecode orc_program_set_backup_function orc_program_compile orc_program_take_code orc_program_free gst_audio_channel_mixer_flags_get_type g_flags_register_static gst_audio_channel_position_get_type g_enum_register_static gst_audio_converter_flags_get_type gst_audio_format_get_type gst_audio_format_flags_get_type gst_audio_pack_flags_get_type gst_audio_flags_get_type gst_audio_layout_get_type gst_audio_quantize_flags_get_type gst_audio_resampler_flags_get_type gst_audio_base_sink_discont_reason_get_type gst_audio_ring_buffer_state_get_type gst_audio_ring_buffer_format_type_get_type libgsttag-1.0.so.0 libz.so.1 libgstbase-1.0.so.0 libgstreamer-1.0.so.0 libgobject-2.0.so.0 libglib-2.0.so.0 libm.so.6 liborc-0.4.so.0 libpthread.so.0 libc.so.6 _edata __bss_start _end libgstaudio-1.0.so.0 GLIBC_2.2.5 GLIBC_2.14 GLIBC_2.4 GLIBC_2.3.4 GLIBC_2.7 A ui B A ` B ii
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h@0пhX gst_audio_channel_mixer_samples gst_audio_channel_mixer_new gst_audio_channel_mixer_new_with_matrix gst_audio_channel_mixer_new_with_matrix format == GST_AUDIO_FORMAT_S16 || format == GST_AUDIO_FORMAT_S32 || format == GST_AUDIO_FORMAT_F32 || format == GST_AUDIO_FORMAT_F64 in_channels > 0 && in_channels < 64 out_channels > 0 && out_channels < 64 Matrix for %d -> %d: } , { audio-channel-mixer object audio-channel-mixer audio-channel-mixer.c mix != NULL mix->matrix != NULL ? ? D5?> audio-converter object audio-converter audio-converter.c pack %p, %p %lu output samples %p %lu convert->in_place alloc samples %d %lu %lu temp samples %p %lu set samples %p %lu convert in %p, %p, %lu convert out %p, %p %lu deinterleaving %p, %p %lu unpack in-place %p, %lu unpack to tmp %p, %lu copy %p, %p, %lu unpack %p, %p, %lu get in samples %p resample %p %p,%lu %lu mix %p, %p, %lu quantize %p, %p %lu convert != NULL free chain %p GstAudioConverter new rate %d -> %d in_info != NULL out_info != NULL GstAudioConverter.mix-matrix unitsizes: %d -> %d unpack format %s to %s convert S32 to F64 convert F64 to S32 depth in %d, out %d chain %p: %d %d unpositioned channels Invalid mix matrix out != NULL skipping empty buffer convert endian: %lu / %lu samples passthrough: %lu / %lu samples (in_rate == 0 && out_rate == 0) || convert->flags & GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE Invalid mix matrix size, should be %d Invalid mix matrix row size, should be %d Invalid mix matrix element type, should be float mix format %s, passthrough %d, in_channels %d, out_channels %d GstAudioConverter.resampler-method GstAudioConverter.dither-method GstAudioConverter.noise-shaping-method using no dither and noise shaping using dither %d and noise shaping %d quantize to %d bits, dither %d, ns %d same formats, same layout, no resampler and passthrough mixing -> passthrough same formats, and passthrough mixing -> only resampling no resampler, passthrough mixing -> only endian conversion initializing 16-bit endian conversion initializing 24-bit endian conversion initializing 32-bit endian conversion initializing 64-bit endian conversion unsupported sample width for endian conversion flags ^ GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE gst_audio_converter_convert gst_audio_converter_samples gst_audio_converter_samples audio_chain_free gst_audio_converter_free get_output_samples get_temp_samples setup_allocators chain_pack do_change_layout do_change_layout do_quantize chain_quantize do_convert_out chain_convert_out do_resample do_mix chain_mix do_convert_in chain_convert_in audio_chain_set_samples do_unpack chain_unpack check_mix_matrix converter_generic converter_passthrough converter_passthrough converter_endian gst_audio_converter_new gst_audio_converter_new gst_audio_converter_get_config gst_audio_converter_update_config gst_audio_converter_update_config audio-info object audio-info GstAudioInfo Invalid channel positions caps != NULL gst_caps_is_fixed (caps) audio-info.c parsing caps %pA audio/x-raw layout non-interleaved channel-mask no format given unknown format given no layout given unknown layout given no rate property given no channels property given info->finfo != NULL no rate or bpf configured ret=%d result %li channels <= 64 || position == NULL wrong name, expected audio/x-raw Invalid channel mask 0x%016lx for %d channels info->finfo->format != GST_AUDIO_FORMAT_UNKNOWN file %s: line %d (%s): should not be reached Unpositioned audio channel position flag set but channel positions present Unpositioned audio channel position flag not set but no channel positions present converting value %li from %s (%d) to %s (%d) gst_audio_info_convert gst_audio_info_to_caps gst_audio_info_to_caps gst_audio_info_from_caps gst_audio_info_from_caps gst_audio_info_set_format gst_audio_info_init format == GST_AUDIO_FORMAT_S32 quant != NULL out != NULL || samples == 0 in != NULL || samples == 0 gst_audio_quantize_samples gst_audio_quantize_free gst_audio_quantize_new oտfd?(r*-9@+j0
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ףpMbX?R Q$C @ ? @ ? 0C ? audio-resampler.c can't find exact taps options != NULL in_rate > 0 && out_rate > 0 GstAudioResampler.n-taps GstAudioResampler.cubic-b GstAudioResampler.cubic-c GstAudioResampler.cutoff resampler != NULL old phase %d/%d using exact phase divider divide by factor %d, gcd %d new phase %d/%d Fc %f, A %f, tr_bw %f GstAudioResampler.filter-mode setting up filter cache realloc %d -> %d taps %d->%d, %d using nearest filter function using full filter function using resample function %d in_rate > 0 audio-resampler object audio-resampler target %s, default flags %08x target flag %s sse audio-resampler-x86.h enable SSE optimisations sse2 enable SSE2 optimisations sse41 enable SSE41 optimisations GstAudioResampler.options quality <= GST_AUDIO_RESAMPLER_QUALITY_MAX GstAudioResampler.transition-bandwidth GstAudioResampler.stop-attenutation GstAudioResampler.filter-oversample GstAudioResampler.max-phase-error phase %d out_rate %d, in_rate %d, gcd %d have new options, reconfigure filter using Beta %f n_taps %d cutoff %f GstAudioResampler.filter-mode-threshold GstAudioResampler.filter-interpolation using n_taps %d cutoff %f oversample %d automatically selected full filter, %d <= %d automatically selected full filter, memory %d <= %d automatically selected interpolated filter using linear interpolation to build filter using cubic interpolation to build filter allocate bps %d n_taps %d n_phases %d using linear interpolation for filter coefficients using cubic interpolation for filter coefficients using filter interpolate function %d using interpolated filter function method >= GST_AUDIO_RESAMPLER_METHOD_NEAREST && method <= GST_AUDIO_RESAMPLER_METHOD_KAISER method %d, bps %d, channels %d need %d = %d + %d + %d, avail %d = %d + %d out %d = ((%d * %d - %d) / %d) + 1 in %lu, avail %lu, consumed %lu wvvur gst_audio_resampler_resample gst_audio_resampler_get_max_latency gst_audio_resampler_get_in_frames gst_audio_resampler_get_out_frames gst_audio_resampler_get_out_frames gst_audio_resampler_free setup_functions get_sample_bufs alloc_taps_mem calculate_kaiser_params resampler_calculate_taps gst_audio_resampler_update gst_audio_resampler_update gst_audio_resampler_reset audio_resampler_check_x86 audio_resampler_init convert_taps_gint16_c convert_taps_gint32_c gst_audio_resampler_new gst_audio_resampler_new gst_audio_resampler_options_set_quality Bcil`g> _K ?+^?.0<`7 =tui=xVL=U@=[[R>lg>/
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?`"? ? \@+?333333? ? ^@ Zd;? @ @ "@ (@ 2@ @ >@ ( H@ @ 8@-DT! @ @; %N?zD?1|?C ˅? !@ ? )@ N@ A @ @qU? `GUտ?333333?Gz?qx? @U@ʡE?:0yE> I@ffffff!@K46? 5@?W2ı?Uj@+0?-DT!@HzG@*>;gst_audio_aggregator_pad_flush_pad A GstStructure describing the configuration that should be used when converting this pad's audio buffers gst_audio_aggregator_src_event gst_audio_aggregator_sink_event gst_audio_aggregator_src_query gst_audio_aggregator_aggregate gst_audio_aggregator_update_src_caps Output block size in nanoseconds Timestamp alignment threshold in nanoseconds Window of time in nanoseconds to wait before creating a discontinuity Creating output buffer with size %d Total duration in format %s: %u:%02u:%02u.%09u Segment of type %s are not supported, only TIME segments are supported Got segment event with wrong rate %lf, expected %lf Negative rates not supported yet seeking failed, unhandled seek type for start: %d seeking failed, unhandled seek type for end: %d seeking failed, unhandled seek format: %s %s:%d: invalid %s id %u for "%s" of type '%s' in '%s' Got timeout before receiving any caps, don't output anything Starting to mix %u samples for offset %li with timestamp %u:%02u:%02u.%09u Timeout, missing %li frames (%u:%02u:%02u.%09u) Dropping 0-sized buffer missing either a duration or a GAP flag: %pA start_time >= aggpad->segment.start Have discont. Expected %lu, got %lu Buffer before segment or current position: %lu < %li Buffer resynced: Pad offset %lu, current audio aggregator offset %li Queued new buffer at offset %lu Buffer was late by %u:%02u:%02u.%09u, dropping %pA Mixing buffer for current offset Finished mixing buffer, waiting for next A pad dropped a buffer, wait for the next one We're not done yet for the current offset, waiting for more data Last buffer is incomplete: %lu <= %li pushing outbuf %p, timestamp %u:%02u:%02u.%09u offset %li Unknown data received, not negotiated Pad is at or after current offset: %lu >= %li first configured pad has sample rate %d Converter configuration converter-config gst_audio_aggregator_do_clip GstAudioAggregator audioaggregator Output Buffer Duration output-buffer-duration Alignment Threshold alignment-threshold Discont Wait discont-wait gstaudioaggregator.c "S16"LE Fixated src caps to %pA Updated src caps to %pA Rejecting invalid caps: %pA Got %s event on sink pad Got caps %pA Got %s event on src pad property klass->convert_buffer src caps negotiated %pA setting caps to %pA Starting at offset %li Buffer outside output segment Pad is in EOS state pad->priv->buffer skipping GAP buffer We're EOS pushed outbuf, result = %s error: %s GstAudioAggregatorPad GstAudioAggregatorConvertPad Getting caps with filter %pA sink template caps : %pA downstream caps %pA returned sink caps : %pA gst_audio_aggregator_set_sink_caps gst_audio_aggregator_src_event gst_audio_aggregator_sink_setcaps gst_audio_aggregator_sink_event gst_audio_aggregator_query_duration gst_audio_aggregator_sink_getcaps gst_audio_aggregator_convert_sink_getcaps gst_audio_aggregator_mix_buffer gst_audio_aggregator_fill_buffer gst_audio_aggregator_fill_buffer gst_audio_aggregator_aggregate gst_audio_aggregator_aggregate gst_audio_aggregator_update_src_caps gst_audio_aggregator_fixate_src_caps gst_audio_aggregator_convert_buffer gst_audio_aggregator_negotiated_src_caps gst_audio_aggregator_create_output_buffer ringbuffer class default_clear_all default_commit gstaudioringbuffer.c pausing ringbuffer signal waiter failed to pause was not started GstAudioRingBuffer spec caps: %p %pA audio/x-alaw audio/x-mulaw audio/x-iec958 audio/x-ac3 audio/x-eac3 audio/x-dts audio/mpeg mpegaudioversion mpegversion stream-format adts audio/x-flac spec->latency_time != 0 could not parse caps !buf->acquired opened device failed opening device closed device error closing device acquiring device %p acquired device device not opened Device for %p not opened device was acquired failed to acquire device device not acquired Device for %p not acquired device was active in mode %d failed to activate device starting ringbuffer was not stopped, try paused resuming failed to start we are flushing we are not acquired we may not start start! waiting.. stopped processing not allowed to start buf->memory != NULL write %d : %d copy %u bytes fwd_up end %d/%d fwd_down end %d/%d rev_up end %d/%d rev_down end %d/%d stopping was not started, try paused failed to stop releasing device buf->open released device device was released failed to release device set sample to %lu, segbase %d segment != NULL readptr != NULL len != NULL clear segment %d @%p clear all segments may start: %d mu law a law ima adpcm gsm aac mpeg2 aac mpeg4 aac mpeg2 raw aac mpeg4 raw parsed caps: type: %d, '%s' 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 GST_IS_AUDIO_RING_BUFFER (buf) Device for ring buffer already open Device for ring buffer %p already open, fix your code 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 Allocating an array for %d timestamps invalid bytes_per_frame from acquire ringbuffer %p, fix the element 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 was not paused, must have been stopped Freeing timestamp buffer, %d entries 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 Retrieved timestamp %u:%02u:%02u.%09u @ %d prepare read from segment %d (real %d) @%p position-less channels, no need to reorder may have to reorder channels: %s -> %s Storing timestamp %u:%02u:%02u.%09u @ %d gst_audio_ring_buffer_set_timestamp gst_audio_ring_buffer_set_timestamp gst_audio_ring_buffer_set_channel_positions gst_audio_ring_buffer_set_channel_positions gst_audio_ring_buffer_may_start gst_audio_ring_buffer_may_start gst_audio_ring_buffer_clear gst_audio_ring_buffer_clear gst_audio_ring_buffer_advance gst_audio_ring_buffer_advance gst_audio_ring_buffer_prepare_read gst_audio_ring_buffer_prepare_read gst_audio_ring_buffer_read gst_audio_ring_buffer_read gst_audio_ring_buffer_commit gst_audio_ring_buffer_clear_all gst_audio_ring_buffer_set_sample gst_audio_ring_buffer_set_sample gst_audio_ring_buffer_samples_done gst_audio_ring_buffer_delay gst_audio_ring_buffer_delay gst_audio_ring_buffer_stop gst_audio_ring_buffer_stop gst_audio_ring_buffer_pause gst_audio_ring_buffer_pause gst_audio_ring_buffer_start gst_audio_ring_buffer_start gst_audio_ring_buffer_is_flushing gst_audio_ring_buffer_pause_unlocked gst_audio_ring_buffer_set_flushing gst_audio_ring_buffer_is_active gst_audio_ring_buffer_activate gst_audio_ring_buffer_activate gst_audio_ring_buffer_is_acquired gst_audio_ring_buffer_release gst_audio_ring_buffer_release gst_audio_ring_buffer_acquire gst_audio_ring_buffer_acquire gst_audio_ring_buffer_device_is_open gst_audio_ring_buffer_close_device gst_audio_ring_buffer_close_device gst_audio_ring_buffer_open_device gst_audio_ring_buffer_open_device gst_audio_ring_buffer_set_callback_full gst_audio_ring_buffer_parse_caps gst_audio_ring_buffer_parse_caps gst_audio_ring_buffer_debug_spec_buff gst_audio_ring_buffer_debug_spec_caps default_clear_all wait_segment default_commit default_commit gstaudioclock.c audioclock GstAudioClock clock-type name no time, reuse last result %u:%02u:%02u.%09u, last_time %u:%02u:%02u.%09u reset clock to %u:%02u:%02u.%09u, last %u:%02u:%02u.%09u, offset %c%u:%02u:%02u.%09u gst_audio_clock_get_time gst_audio_clock_reset gst_audio_clock_get_internal_time gst_audio_clock_init cdda://%s#%d cdda://1 Audio CD source base class audiocdsrc CD track CD sector CD device location Device Track gst_audio_cd_src_start gst_audio_cd_src_stop gst_audio_cd_src_query gst_audio_cd_src_handle_event gst_audio_cd_src_do_seek gst_audio_cd_src_is_seekable gst_audio_cd_src_create gstaudiocdsrc.c klass->read_sector != NULL announcing tags: %pA asking for sector %u failed to read sector %u track number %d out of bounds sample source value negative source time value invalid returning %li unsupported format conversion failed: %s conversion failed seek'd to sector %d invalid track %li seeking to track %li starting at sector %d handling %s event seek failed: device not open audiocd-track-%03u let base class handle event handling %s query src->priv->tracks != NULL duration: %d tracks position: track %d /dev/cdrom # successfully handled uri '%s' cannot handle URI '%s' Could not handle CDDA URI Invalid track %u klass->close != NULL klass->open != NULL opening device %s cddb discid = %s %02X %08X 00000000 musicbrainz-discid = %s musicbrainz-discid-full = %s musicbrainz-discid-full musicbrainz-discid track-count track-number src->tags = %pA seek to track %d failed to open device This CD has no audio tracks gst-plugins-base-1.0 GstAudioCdSrc AUDIO DATA GST_IS_AUDIO_CD_SRC (src) track != NULL track->num > 0 cdda duration updated to %u:%02u:%02u.%09u EOS at sector %d, cur_track=%d, mode=%d pushing sector %d with timestamp %u:%02u:%02u.%09u converting value %li from %s into %s source value not within allowed range cannot do this conversion, device not open segment %u:%02u:%02u.%09u-%u:%02u:%02u.%09u src->priv->cur_track >= 0 && src->priv->cur_track < src->priv->num_tracks segment seek in track format is only supported in CONTINUOUS mode, not in mode %d cannot convert track %d to time seek segment %u:%02u:%02u.%09u-%u:%02u:%02u.%09u start seek type is NONE, nothing to do ignoring stop seek type (expected NONE) is current track, just seeking back to start seek in sector format not supported let base class handle seek in %s format duration: %u sectors, %li in format %s position: sector %u, %li in format %s unhandled query, chaining up to parent class track %02u: lsn %6u (%02u:%02u), length: %u seconds (%02u:%02u) first_sector = %u = %u secs (%02u:%02u) last_sector = %u = %u secs (%02u:%02u) total length = %u secs (%02u:%02u), added title lengths = %u seconds (%02u:%02u) i < sizeof (src->priv->mb_discid) + 1 Track %03u %u:%02u:%02u.%09u - %u:%02u:%02u.%09u adding track %2u (%2u) [%6u-%6u] [%5s], tags: %pA track %2u overlaps with previous tracks audio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) 44100, channels = (int) 2 Px(pgst_audio_cd_src_add_track gst_audio_cd_src_add_track gst_audio_cd_src_make_toc gst_audio_cd_src_add_tags gst_audio_cd_src_calculate_musicbrainz_discid gst_audio_cd_src_calculate_musicbrainz_discid gst_audio_cd_src_calculate_cddb_id gst_audio_cd_src_start gst_audio_cd_src_start gst_audio_cd_src_stop gst_audio_cd_src_query gst_audio_cd_src_query gst_audio_cd_src_handle_track_seek gst_audio_cd_src_handle_track_seek gst_audio_cd_src_handle_event gst_audio_cd_src_convert gst_audio_cd_src_do_seek gst_audio_cd_src_do_seek gst_audio_cd_src_update_duration gst_audio_cd_src_create gst_audio_cd_src_create gst_audio_cd_src_uri_set_uri gstaudiodecoder.c received query %pA received event %d, %s peer caps %pA starting segment %pB Pushing pending events ctx->info.bpf != 0 marking discont subclass returned %s, buf %p forcing fragment flush starting new fragment adding to fragment assembling fragment buffer pushed: %s buffer queued gst_audio_decoder_reset gst_audio_decoder_init pad_template != NULL gst_audio_decoder_sink_event gst_audio_decoder_chain gst_audio_decoder_sink_query sinkpad created gst_audio_decoder_src_event gst_audio_decoder_src_query srcpad created init ok audio decoder base class audiodecoder Minimum Latency min-latency Packet Loss Concealment plc handling query: %pA upstream size %li returning peer response Ignoring BYTES position query upstream : %pA decoder : %pA mode : %d merged : %pA Failed to open codec gst_audio_decoder_start gst_audio_decoder_stop Failed to close codec Failed to stop codec Failed to start codec copy metadata %s !is_subframe || buf != NULL warning: %s base_ts now %u:%02u:%02u.%09u ts diff %d ms base_ts resync klass->handle_frame != NULL parsing available: %d skipped %d; setting DISCONT failed to parse stream no frame yet len offset + len <= av ts == prev_ts; discarding av >= 0 done pushing to subclass discarding empty buffer chain-done Applying pending segments audio decoder output failed flush hard %d received discont flushing buffers to decoder decoded buffer to %p tracking ts %u:%02u:%02u.%09u decoder not initialized handling discont received TIME SEGMENT %pB received SEGMENT %pB unsupported format; ignoring no valid frames found upstream stream tags: %pA Enqueuing event %d, %s unsupported seek: rate unsupported seek: start time unsupported seek: end time GstAudioDecoder GST_IS_AUDIO_DECODER (dec) setting src caps %pA GST_IS_PAD (pad) ALLOCATION (%d) params: %pA GST_IS_AUDIO_DECODER (object) Setting srcpad caps %pA invalid output format frames != 0 Returning caps %pA GST_CLOCK_TIME_IS_VALID (min) min <= max enabled: %d setting decoder tags to %pA size > 0 alloc src buffer No default channel-mask for %d channels Chose default caps %pA for initial gap clipping buffer of size %lu with ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u no data after clipping to segment GST_BUFFER_DURATION_IS_VALID (buf) pushing buffer of size %lu with ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u output buffer of size %lu with ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u buffer %d ms apart from current fragment gst_audio_decoder_change_state Aggregate output data to a minimum of latency time (ns) Perfect ts while timestamp jitter/imperfection within tolerance (ns) Perform packet loss concealment (if supported) gst_audio_decoder_sink_eventfunc gst_audio_decoder_src_eventfunc gst_audio_decoder_propose_allocation_default gst_audio_decoder_decide_allocation_default gst_audio_decoder_negotiate_default gst_audio_decoder_sink_query_default gst_audio_decoder_src_query_default gst_audio_decoder_transform_meta_default query %p: our time: %u:%02u:%02u.%09u query %p: we return %li (format %u) Peer latency: live %d, min %u:%02u:%02u.%09u max %u:%02u:%02u.%09u not copying memory specific metadata %s transformed metadata %s: copy: %d buf == NULL || gst_buffer_get_size (buf) > 0 buf == NULL || GST_AUDIO_INFO_IS_VALID (&ctx->info) buffer size %lu not a multiple of %d received more decoded frames %d than provided %d leading frame ts %u:%02u:%02u.%09u Received decoded subframe, but no pending frame GST_CLOCK_TIME_IS_VALID (priv->base_ts) buffer is %lu samples past base_ts %u:%02u:%02u.%09u, expected ts %u:%02u:%02u.%09u Can't copy metadata because input buffers disappeared GstAudioMeta samples (%lu) are inconsistent with the buffer size and layout (size/bpf = %lu) accepting %lu bytes == %lu samples for %d frames Skipping decode in trickmode and sending gap tracking frame size %lu, ts %u:%02u:%02u.%09u providing subclass with NULL frame Decoder output not negotiated before GAP event. Failed to negotiate with downstream gap event: plc %d, do_plc %d, position %u:%02u:%02u.%09u duration %u:%02u:%02u.%09u frame at offset %d of length %d audio decoder push buffers failed still %d frames left after draining decoding buffer %p, ts %u:%02u:%02u.%09u buffer did not decode, keeping GST_CLOCK_TIME_IS_VALID (duration) applying reverse interpolated ts %u:%02u:%02u.%09u pushing buffer %p of size %lu, time %u:%02u:%02u.%09u, dur %u:%02u:%02u.%09u gathering buffer %p of size %lu, time %u:%02u:%02u.%09u, dur %u:%02u:%02u.%09u received buffer of size %lu with ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u ... but restoring previous ts tracking received STREAM_START. Clearing taglist converted to TIME start %u:%02u:%02u.%09u No valid frames decoded before end of stream Caps did not change, not setting again unsupported seek: not flushing seeking to %u:%02u:%02u.%09u at byte offset %li cannot convert start/stop for seek GST_AUDIO_INFO_IS_VALID (&dec->priv->ctx.info) GST_IS_CAPS (dec->priv->ctx.caps) caps != NULL && gst_caps_is_fixed (caps) didn't get downstream ALLOCATION hints klass->decide_allocation != NULL Subclass failed to decide allocation Requested output format %pA do not match template %pA discarding non-TIME SEEKING query tags == NULL || GST_IS_TAG_LIST (tags) Failed to negotiate, fallback allocation couldn't allocate output buffer gst_audio_decoder_get_allocator gst_audio_decoder_allocate_output_buffer gst_audio_decoder_allocate_output_buffer gst_audio_decoder_merge_tags gst_audio_decoder_merge_tags gst_audio_decoder_get_needs_format gst_audio_decoder_set_needs_format gst_audio_decoder_get_drainable gst_audio_decoder_set_drainable gst_audio_decoder_get_tolerance gst_audio_decoder_set_tolerance gst_audio_decoder_get_min_latency gst_audio_decoder_set_min_latency gst_audio_decoder_get_plc gst_audio_decoder_set_plc gst_audio_decoder_set_plc gst_audio_decoder_set_allocation_caps gst_audio_decoder_get_parse_state gst_audio_decoder_get_latency gst_audio_decoder_set_latency gst_audio_decoder_get_max_errors gst_audio_decoder_set_max_errors gst_audio_decoder_get_delay gst_audio_decoder_get_estimate_rate gst_audio_decoder_set_estimate_rate gst_audio_decoder_get_plc_aware gst_audio_decoder_set_plc_aware gst_audio_decoder_get_audio_info _gst_audio_decoder_error gst_audio_decoder_finish_frame foreach_metadata gst_audio_decoder_finish_frame_or_subframe gst_audio_decoder_finish_frame_or_subframe gst_audio_decoder_finish_subframe gst_audio_decoder_set_output_caps gst_audio_decoder_set_output_caps gst_audio_decoder_set_output_format gst_audio_decoder_set_output_format gst_audio_decoder_negotiate gst_audio_decoder_finalize gst_audio_decoder_stop gst_audio_decoder_start gst_audio_decoder_change_state gst_audio_decoder_create_merged_tags_event gst_audio_decoder_sink_setcaps gst_audio_decoder_sink_eventfunc gst_audio_decoder_do_seek gst_audio_decoder_src_eventfunc gst_pad_set_caps gst_audio_decoder_negotiate_default gst_audio_decoder_negotiate_default gst_audio_decoder_sink_getcaps gst_audio_decoder_sink_query_default gst_audio_decoder_src_query_default gst_audio_decoder_reset gst_audio_decoder_sink_event gst_audio_decoder_flush_decode gst_audio_decoder_flush_decode gst_audio_decoder_chain_reverse gst_audio_decoder_chain_forward gst_audio_decoder_push_forward gst_audio_decoder_push_forward gst_audio_decoder_output gst_audio_decoder_push_event send_pending_events gst_audio_decoder_negotiate_default_caps gst_audio_decoder_handle_gap gst_audio_decoder_handle_frame gst_audio_decoder_push_buffers gst_audio_decoder_push_buffers apply_pending_events gst_audio_decoder_drain gst_audio_decoder_flush gst_audio_decoder_chain gst_audio_decoder_sink_query gst_audio_decoder_src_event gst_audio_decoder_src_query gst_audio_decoder_init gst_audio_decoder_init gstaudioencoder.c received query %d, %s event result %d reset full %d gst_audio_encoder_init gst_audio_encoder_sink_event gst_audio_encoder_sink_query gst_audio_encoder_chain gst_audio_encoder_src_event gst_audio_encoder_src_query src created sink activate push %d activate %d activate return: %d sink activate push return: %d audio encoder base class audioencoder gst_audio_encoder_finalize Perfect Timestamps perfect-timestamp Granule Marking mark-granule Hard Resync hard-resync encoder : %pA new base gp %li GstAudioEncoder GST_IS_AUDIO_ENCODER (enc) no caps set distance % ctx->info.bpf == 0 marked discont Sending headers pushing header returned %s taking %lu bytes for output out samples %d priv->offset <= av draining ... really klass->set_format != NULL rejected caps %pA audio-codec video-codec subtitle-codec container-format nominal-bitrate maximum-bitrate minimum-bitrate encoder-version new base ts %u:%02u:%02u.%09u chain leaving encoder not initialized returning caps %pA set to %d new headers %p set to %u:%02u:%02u.%09u setting encoder tags to %pA refused caps %pA gst_audio_encoder_sink_activate_mode !enc->priv->granule || enc->priv->perfect_ts Favour perfect timestamps over tracking upstream timestamps Apply granule semantics to buffer metadata (implies perfect-timestamp) Perform clipping and sample flushing upon discontinuity Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns) gst_audio_encoder_change_state perfect-timestamp can not be set FALSE while granule handling is enabled GST_IS_CAPS (enc->priv->ctx.caps) accepting %lu bytes encoded data as %d samples %lu samples past prev_ts %u:%02u:%02u.%09u %lu samples past base_ts %u:%02u:%02u.%09u new upstream ts %u:%02u:%02u.%09u at distance %lu new upstream ts only introduces jitter received more encoded samples %d than provided %d as inputs Can't copy metadata because input buffer disappeared available: %d, needed: %d, force: %d non-drainable and no more data providing subclass with %d bytes at offset %d bypassing subclass with leftover no more data drained from subclass subclass did not accept format new audio format identical to configured format incoming buffer had incorrect duration %u:%02u:%02u.%09u, expected duration %u:%02u:%02u.%09u buffer after segment clipping has size %lu with ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u Buffer is older than expected ts %u:%02u:%02u.%09u. Clipping buffer buffer going too far back in time set to %u:%02u:%02u.%09u-%u:%02u:%02u.%09u tags == NULL || mode != GST_TAG_MERGE_UNDEFINED gst_audio_encoder_get_allocator gst_audio_encoder_allocate_output_buffer gst_audio_encoder_allocate_output_buffer gst_audio_encoder_set_output_format gst_audio_encoder_negotiate gst_audio_encoder_merge_tags gst_audio_encoder_merge_tags gst_audio_encoder_get_drainable gst_audio_encoder_set_drainable gst_audio_encoder_get_hard_min gst_audio_encoder_set_hard_min gst_audio_encoder_get_tolerance gst_audio_encoder_set_tolerance gst_audio_encoder_set_tolerance gst_audio_encoder_get_hard_resync gst_audio_encoder_set_hard_resync gst_audio_encoder_set_hard_resync gst_audio_encoder_get_perfect_timestamp gst_audio_encoder_set_perfect_timestamp gst_audio_encoder_set_perfect_timestamp gst_audio_encoder_get_mark_granule gst_audio_encoder_set_mark_granule gst_audio_encoder_set_mark_granule gst_audio_encoder_set_allocation_caps gst_audio_encoder_set_headers gst_audio_encoder_get_latency gst_audio_encoder_set_latency gst_audio_encoder_set_latency gst_audio_encoder_get_lookahead gst_audio_encoder_set_lookahead gst_audio_encoder_set_lookahead gst_audio_encoder_get_frame_max gst_audio_encoder_set_frame_max gst_audio_encoder_set_frame_max gst_audio_encoder_get_frame_samples_max gst_audio_encoder_set_frame_samples_max gst_audio_encoder_set_frame_samples_max gst_audio_encoder_get_frame_samples_min gst_audio_encoder_set_frame_samples_min gst_audio_encoder_set_frame_samples_min gst_audio_encoder_get_audio_info foreach_metadata gst_audio_encoder_finish_frame gst_audio_encoder_finish_frame gst_audio_encoder_set_property gst_audio_encoder_change_state gst_audio_encoder_getcaps_default gst_audio_encoder_create_merged_tags_event gst_audio_encoder_push_pending_events gst_audio_encoder_sink_event_default gst_audio_encoder_src_query_default gst_pad_set_caps gst_audio_encoder_push_event gst_audio_encoder_negotiate_default gst_audio_encoder_negotiate_default gst_audio_encoder_reset gst_audio_encoder_sink_event gst_audio_encoder_sink_query gst_audio_encoder_push_buffers gst_audio_encoder_push_buffers gst_audio_encoder_set_base_gp gst_audio_encoder_drain gst_audio_encoder_sink_setcaps gst_audio_encoder_sink_setcaps gst_audio_encoder_chain gst_audio_encoder_activate gst_audio_encoder_activate gst_audio_encoder_sink_activate_mode gst_audio_encoder_src_event gst_audio_encoder_src_query gst_audio_encoder_init gst_audio_encoder_init GstAudioSinkClock gstaudiobasesink.c Provide Clock provide-clock Slave Method slave-method Allow pull-based scheduling Allow Pull Scheduling can-activate-pull Drift Tolerance drift-tolerance gst_audio_base_sink_query gst_audio_base_sink_fixate gst_audio_base_sink_setcaps gst_audio_base_sink_event gst_audio_base_sink_get_times gst_audio_base_sink_preroll gst_audio_base_sink_render gst_audio_base_sink_query_pad width depth signed endianness latency query query convert deactivating pull drained audio error %s 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 failed to payload. unknown slaving method %d Clipped start: %u/%u samples resync after discont/resync rendering at %lu %d/%d wrote %u of %u next sample expected at %lu dropping late sample ringbuffer not negotiated wrong size failed waiting for latency release old ringbuffer time was %u:%02u:%02u.%09u activate ringbuffer cannot parse audio format. could not acquire ringbuffer clock provide disabled ringbuffer not acquired GstAudioBaseSink audiobasesink element audiobasesink GST_IS_AUDIO_BASE_SINK (sink) ringbuffer may start now create failed open failed processed samples: raw %lu, delay %u, real %lu, time %u:%02u:%02u.%09u Size of audio buffer in microseconds, this is the minimum latency that the sink reports The minimum amount of data to write in each iteration in microseconds Provide a clock to be used as the global pipeline clock Algorithm used to match the rate of the masterclock Tolerance for clock drift in microseconds gst_audio_base_sink_change_state gst_audio_base_sink_provide_clock gst_audio_base_sink_wait_event gst_audio_base_sink_activate_pull we are not negotiated, can't report latency yet peer min %u:%02u:%02u.%09u, our min latency: %u:%02u:%02u.%09u peer max %u:%02u:%02u.%09u, our max latency: %u:%02u:%02u.%09u peer or we are not live, don't care about latency last sample time %u:%02u:%02u.%09u pulling %u bytes offset %lu to fill audio buffer got different size than requested from sink pad: %u != %lu Got flow '%s' but can't return it: %d Sink not negotiated before %s event. 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 Received GAP or ignoring audio for trickplay. Dropping contents time %u:%02u:%02u.%09u, start %u:%02u:%02u.%09u, samples %u Buffer of size %lu has no time. Using render_start=%lu sync-offset %c%u:%02u:%02u.%09u, render-delay %u:%02u:%02u.%09u, ts-offset %c%u:%02u:%02u.%09u 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 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 %c%u:%02u:%02u.%09u avg %c%u:%02u:%02u.%09u correct clock skew %c%u:%02u:%02u.%09u > %c%u:%02u:%02u.%09u last_align %li driftsamples %u, next %lu correct clock skew %c%u:%02u:%02u.%09u < -%c%u:%02u:%02u.%09u custom slaving requested skew %c%u:%02u:%02u.%09u no custom slaving callback set - clock drift will not be compensated final timestamps: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u apply time offset %c%u:%02u:%02u.%09u no align possible: no previous sample position known align with prev sample, ABS (%li) < %li Unexpected discontinuity in audio timestamps of %s%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. preroll got interrupted: %d (%s) Ringbuffer caps haven't changed, skipping reconfiguration _ _8_P_p____ _ cdb|bbbbbbLb8ecccccccceccccccdc8fcccccchd gst_audio_base_sink_get_discont_wait gst_audio_base_sink_report_device_failure gst_audio_base_sink_set_custom_slaving_callback gst_audio_base_sink_set_discont_wait gst_audio_base_sink_get_alignment_threshold gst_audio_base_sink_set_alignment_threshold gst_audio_base_sink_get_drift_tolerance gst_audio_base_sink_set_drift_tolerance gst_audio_base_sink_get_slave_method gst_audio_base_sink_set_slave_method gst_audio_base_sink_get_provide_clock gst_audio_base_sink_set_provide_clock gst_audio_base_sink_change_state gst_audio_base_sink_provide_clock gst_audio_base_sink_query gst_audio_base_sink_setcaps gst_audio_base_sink_wait_event gst_audio_base_sink_preroll gst_audio_base_sink_get_alignment gst_audio_base_sink_custom_slaving gst_audio_base_sink_skew_slaving gst_audio_base_sink_resample_slaving gst_audio_base_sink_sync_latency gst_audio_base_sink_render gst_audio_base_sink_query_pad gst_audio_base_sink_drain gst_audio_base_sink_callback gst_audio_base_sink_activate_pull gst_audio_base_sink_get_time GstAudioSrcClock gstaudiobasesrc.c Actual Buffer Time actual-buffer-time Actual Latency Time actual-latency-time gst_audio_base_src_setcaps gst_audio_base_src_event gst_audio_base_src_query gst_audio_base_src_get_times gst_audio_base_src_create gst_audio_base_src_fixate at segment %d and sample %lu dropped, align to segment %d read %u of %u wait playing continue playing alloc failed: %s ringbuffer stopped flush-start flush-stop refuse to seek forward event %p acquire new ringbuffer subclass posted error ringbuffer is flushing GstAudioBaseSrc audiobasesrc element audiobasesrc /usr/share/locale UTF-8 GST_IS_AUDIO_BASE_SRC (src) NULL->READY READY->PAUSED PAUSED->PLAYING PLAYING->PAUSED PAUSED->READY READY->NULL Size of audio buffer in microseconds. This is the maximum amount of data that is buffered in the device and the maximum latency that the source reports. This value might be ignored by the element if necessary; see "actual-buffer-time" The minimum amount of data to read in each iteration in microseconds. This is the minimum latency that the source reports. This value might be ignored by the element if necessary; see "actual-latency-time" Actual configured size of audio buffer in microseconds Actual configured audio latency in microseconds gst_audio_base_src_change_state gst_audio_base_src_provide_clock gst_audio_base_src_post_message resource can only be operated on sequentially but offset %lu was given first sample, align to current %d reading from %d, we are at %d, sample %lu reading from sample %lu length %u 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 Pushed buffer timestamp %u:%02u:%02u.%09u ringbuffer was in error state, bailing out report latency min %u:%02u:%02u.%09u max %u:%02u:%02u.%09u binding text domain %s to locale dir %s `~~~@h~DDDDDDDDDDDDDDdD$DDDDDD gst_audio_base_src_get_slave_method gst_audio_base_src_set_slave_method gst_audio_base_src_get_provide_clock gst_audio_base_src_set_provide_clock _do_init gst_audio_base_src_change_state gst_audio_base_src_provide_clock gst_audio_base_src_post_message gst_audio_base_src_setcaps gst_audio_base_src_event gst_audio_base_src_query gst_audio_base_src_get_offset gst_audio_base_src_create gst_audio_base_src_get_time gst_audio_filter_change_state gst_audio_filter_set_caps gstaudiofilter.c info: %d configured caps: %pA couldn't parse %pA GstAudioFilter audiofilter GST_IS_CAPS (allowed_caps) gst_audio_filter_get_unit_size GST_IS_AUDIO_FILTER_CLASS (klass) gst_audio_filter_class_add_pad_templates gst_audio_filter_set_caps GstAudioDownmixMetaAPI GstAudioDownmixMeta from_position != NULL from_channels > 0 to_position != NULL to_channels > 0 GstAudioClippingMetaAPI GstAudioClippingMeta GstAudioMetaAPI GstAudioMeta format != GST_FORMAT_UNDEFINED info->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED || !offsets GstAudioMeta properties would cause channel memory areas to overlap! offsets: %lu (%d), %lu (%d) with plane size %lu GstAudioMeta properties would cause out-of-bounds memory access on the buffer: max_offset %lu, samples %lu, bps %u, buffer size %lu gst_buffer_add_audio_meta gst_buffer_add_audio_clipping_meta gst_buffer_add_audio_downmix_meta gst_audio_sink_ring_buffer_open_device gst_audio_sink_ring_buffer_close_device gst_audio_sink_ring_buffer_acquire gst_audio_sink_ring_buffer_release gst_audio_sink_ring_buffer_start gst_audio_sink_ring_buffer_pause gst_audio_sink_ring_buffer_stop gst_audio_sink_ring_buffer_delay gst_audio_sink_ring_buffer_activate 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 gstaudiosink.c reset... reset done could not close device could not open device GstAudioSinkRingBuffer creating ringbuffer created ringbuffer @%p start, sending signal starting thread audiosink-ringbuffer waiting for thread thread is started signal wait could not create thread %s unknown enter thread failed to set thread priority 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 GstAudioSink audiosink element audiosink gst_audio_sink_ring_buffer_open_device gst_audio_sink_ring_buffer_close_device gst_audio_sink_ring_buffer_acquire gst_audio_sink_ring_buffer_release gst_audio_sink_ring_buffer_start gst_audio_sink_ring_buffer_pause gst_audio_sink_ring_buffer_stop audioringbuffer_thread_func gst_audio_sink_ring_buffer_activate gst_audio_sink_create_ringbuffer gst_audio_src_ring_buffer_open_device gst_audio_src_ring_buffer_close_device gst_audio_src_ring_buffer_acquire gst_audio_src_ring_buffer_release gst_audio_src_ring_buffer_start gst_audio_src_ring_buffer_stop gst_audio_src_ring_buffer_delay gst_audio_src_create_ringbuffer transfered %d bytes of %d to segment %d error reading data %d (reason: %s), skipping segment GstAudioSrcRingBuffer gstaudiosrc.c audiosrc-ringbuffer GstAudioSrc audiosrc element audiosrc audioringbuffer_thread_func gst_audio_src_ring_buffer_start gst_audio_src_ring_buffer_stop gst_audio_src_create_ringbuffer channels-mask gstaudioutilsprivate.c template caps %pA allowed caps %pA intersecting with %pA proxy caps %pA dest_format != NULL dest_value != NULL not enough metadata yet to convert __gst_audio_encoded_audio_convert __gst_audio_encoded_audio_convert __gst_audio_element_proxy_getcaps Linear volume factor, 1.0=100% Mute the audio channel without changing the volume volume Mute mute GstStreamVolume GST_IS_STREAM_VOLUME (volume) val >= 0.0 streamvolume.c gst_stream_volume_convert_volume gst_stream_volume_set_mute gst_stream_volume_get_mute gst_stream_volume_set_volume gst_stream_volume_get_volume UUUUUU? 4@ $@alignment frame-size block-size layer src != NULL dst != NULL src != dst src_n >= 6 iec61937 src_n != 0 version > 0 && layer > 0 gstaudioiec61937.c src_n >= 7 pxxX ȰЯgst_audio_iec61937_payload dst_n >= gst_audio_iec61937_frame_size (spec) align != NULL GstAudioStreamAlign rate != 0 gstaudiostreamalign.c gst_audio_stream_align_process gst_audio_stream_align_process gst_audio_stream_align_get_samples_since_discont gst_audio_stream_align_get_timestamp_at_discont gst_audio_stream_align_mark_discont gst_audio_stream_align_get_discont_wait gst_audio_stream_align_set_discont_wait gst_audio_stream_align_get_alignment_threshold gst_audio_stream_align_set_alignment_threshold gst_audio_stream_align_get_rate gst_audio_stream_align_set_rate gst_audio_stream_align_free gst_audio_stream_align_copy gst_audio_stream_align_new audio_orc_double_to_s32 A audio_orc_s32_to_double A audio_orc_update_rand mNA90 x g audio_orc_int_dither h j audio_orc_int_bias h j audio_orc_splat_u64 audio_orc_splat_u32 p audio_orc_splat_u16 O audio_orc_pack_f64_swap audio_orc_pack_f64 audio_orc_pack_f32_swap audio_orc_pack_f32 audio_orc_pack_s32_swap audio_orc_pack_u32_swap audio_orc_pack_s32 p audio_orc_pack_u32 audio_orc_pack_s24_32_swap } audio_orc_pack_u24_32_swap ~ audio_orc_pack_s24_32 } audio_orc_pack_u24_32 ~ audio_orc_pack_s16_swap audio_orc_pack_u16_swap ! ! audio_orc_pack_s16 audio_orc_pack_u16 audio_orc_pack_s8 audio_orc_pack_u8 ! ! audio_orc_unpack_f64_swap audio_orc_unpack_f64 audio_orc_unpack_f32_swap audio_orc_unpack_f32 audio_orc_unpack_s32_swap audio_orc_unpack_s32 p audio_orc_unpack_u32_swap audio_orc_unpack_u32 audio_orc_unpack_s24_32_swap | audio_orc_unpack_u24_32_swap | audio_orc_unpack_s24_32 | audio_orc_unpack_u24_32 | audio_orc_unpack_s16_swap_trunc ! | ! audio_orc_unpack_s16_swap audio_orc_unpack_u16_swap_trunc ! |!! ! audio_orc_unpack_u16_swap ! ! audio_orc_unpack_s16_trunc | audio_orc_unpack_s16 audio_orc_unpack_u16_trunc | audio_orc_unpack_u16 audio_orc_unpack_s8_trunc | audio_orc_unpack_s8 ! ! audio_orc_unpack_u8_trunc | audio_orc_unpack_u8 > AGstAudioChannelMixerFlags GstAudioChannelPosition GstAudioConverterFlags GstAudioFormat GstAudioFormatFlags GstAudioPackFlags GstAudioFlags GstAudioLayout GstAudioDitherMethod GstAudioNoiseShapingMethod GstAudioQuantizeFlags GstAudioResamplerFilterMode GstAudioResamplerMethod GstAudioResamplerFlags GstAudioBaseSinkSlaveMethod GstAudioBaseSinkDiscontReason GstAudioBaseSrcSlaveMethod GstAudioCdSrcMode GstAudioRingBufferState GstAudioRingBufferFormatType mu-law a-law ima-adpcm mpeg2-aac mpeg4-aac mpeg2-aac-raw mpeg4-aac-raw GST_AUDIO_CD_SRC_MODE_NORMAL normal continuous resample re-timestamp GST_AUDIO_BASE_SRC_SLAVE_SKEW skew GST_AUDIO_BASE_SRC_SLAVE_NONE none no-discont new-caps sync-latency device-failure custom GST_AUDIO_RESAMPLER_FLAG_NONE non-interleaved-in non-interleaved-out variable-rate nearest linear cubic blackman-nuttall kaiser interpolated auto GST_AUDIO_QUANTIZE_FLAG_NONE GST_AUDIO_NOISE_SHAPING_NONE error-feedback simple medium GST_AUDIO_NOISE_SHAPING_HIGH high GST_AUDIO_DITHER_NONE GST_AUDIO_DITHER_RPDF rpdf GST_AUDIO_DITHER_TPDF tpdf GST_AUDIO_DITHER_TPDF_HF tpdf-hf GST_AUDIO_LAYOUT_INTERLEAVED GST_AUDIO_FLAG_NONE GST_AUDIO_FLAG_UNPOSITIONED unpositioned GST_AUDIO_PACK_FLAG_NONE truncate-range GST_AUDIO_FORMAT_FLAG_INTEGER integer GST_AUDIO_FORMAT_FLAG_FLOAT float GST_AUDIO_FORMAT_FLAG_SIGNED GST_AUDIO_FORMAT_FLAG_COMPLEX complex GST_AUDIO_FORMAT_FLAG_UNPACK unpack GST_AUDIO_FORMAT_UNKNOWN GST_AUDIO_FORMAT_ENCODED encoded GST_AUDIO_FORMAT_S8 s8 GST_AUDIO_FORMAT_U8 u8 GST_AUDIO_FORMAT_S16LE s16le GST_AUDIO_FORMAT_S16BE s16be GST_AUDIO_FORMAT_U16LE u16le GST_AUDIO_FORMAT_U16BE u16be GST_AUDIO_FORMAT_S24_32LE s24-32le GST_AUDIO_FORMAT_S24_32BE s24-32be GST_AUDIO_FORMAT_U24_32LE u24-32le GST_AUDIO_FORMAT_U24_32BE u24-32be GST_AUDIO_FORMAT_S32LE s32le GST_AUDIO_FORMAT_S32BE s32be GST_AUDIO_FORMAT_U32LE u32le GST_AUDIO_FORMAT_U32BE u32be GST_AUDIO_FORMAT_S24LE s24le GST_AUDIO_FORMAT_S24BE s24be GST_AUDIO_FORMAT_U24LE u24le GST_AUDIO_FORMAT_U24BE u24be GST_AUDIO_FORMAT_S20LE s20le GST_AUDIO_FORMAT_S20BE s20be GST_AUDIO_FORMAT_U20LE u20le GST_AUDIO_FORMAT_U20BE u20be GST_AUDIO_FORMAT_S18LE s18le GST_AUDIO_FORMAT_S18BE s18be GST_AUDIO_FORMAT_U18LE u18le GST_AUDIO_FORMAT_U18BE u18be GST_AUDIO_FORMAT_F32LE f32le GST_AUDIO_FORMAT_F32BE f32be GST_AUDIO_FORMAT_F64LE f64le GST_AUDIO_FORMAT_F64BE f64be GST_AUDIO_FORMAT_S16 s16 GST_AUDIO_FORMAT_U16 u16 GST_AUDIO_FORMAT_S24_32 s24-32 GST_AUDIO_FORMAT_U24_32 u24-32 GST_AUDIO_FORMAT_S32 s32 GST_AUDIO_FORMAT_U32 u32 GST_AUDIO_FORMAT_S24 s24 GST_AUDIO_FORMAT_U24 u24 GST_AUDIO_FORMAT_S20 s20 GST_AUDIO_FORMAT_U20 u20 GST_AUDIO_FORMAT_S18 s18 GST_AUDIO_FORMAT_U18 u18 GST_AUDIO_FORMAT_F32 f32 GST_AUDIO_FORMAT_F64 f64 GST_AUDIO_CONVERTER_FLAG_NONE in-writable mono lfe1 front-left-of-center front-right-of-center lfe2 top-front-left top-front-right top-front-center top-center top-rear-left top-rear-right top-side-left top-side-right top-rear-center bottom-front-center bottom-front-left bottom-front-right wide-left wide-right surround-left surround-right unpositioned-in unpositioned-out GstAudioResamplerFilterInterpolation GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3 GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC GST_AUDIO_RING_BUFFER_STATE_STOPPED GST_AUDIO_RING_BUFFER_STATE_PAUSED GST_AUDIO_RING_BUFFER_STATE_STARTED GST_AUDIO_RING_BUFFER_STATE_ERROR GST_AUDIO_CD_SRC_MODE_CONTINUOUS GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE GST_AUDIO_BASE_SINK_SLAVE_SKEW GST_AUDIO_BASE_SINK_SLAVE_NONE GST_AUDIO_BASE_SINK_SLAVE_CUSTOM GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE GST_AUDIO_RESAMPLER_METHOD_NEAREST GST_AUDIO_RESAMPLER_METHOD_LINEAR GST_AUDIO_RESAMPLER_METHOD_CUBIC GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL GST_AUDIO_RESAMPLER_METHOD_KAISER GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED GST_AUDIO_RESAMPLER_FILTER_MODE_FULL GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK GST_AUDIO_NOISE_SHAPING_SIMPLE GST_AUDIO_NOISE_SHAPING_MEDIUM GST_AUDIO_LAYOUT_NON_INTERLEAVED GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE GST_AUDIO_CHANNEL_POSITION_NONE GST_AUDIO_CHANNEL_POSITION_MONO GST_AUDIO_CHANNEL_POSITION_INVALID GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER GST_AUDIO_CHANNEL_POSITION_LFE1 GST_AUDIO_CHANNEL_POSITION_REAR_LEFT GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER GST_AUDIO_CHANNEL_POSITION_REAR_CENTER GST_AUDIO_CHANNEL_POSITION_LFE2 GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER GST_AUDIO_CHANNEL_POSITION_TOP_CENTER GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT GST_AUDIO_CHANNEL_MIXER_FLAGS_NONE GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT @ @ @ @ ; j ` 8 `X @ X 4 PP d @ 0 Я $ p@ T p ` P @ 0 D 0` Pt p д 0 P p( < P еd x 0 P p ж , 0@ PT h | з @ p 0 P d P ` `4 нT h | 8! ! ! pD" @t" " p" 8# pL# # # $ \$ $ $ 0(% l% 0% % 8& t& & & @' @t' ' ' |( ( ( ( P( ( ) d) ) ) ) * 8* `* * @ ,+ t+ + $, , (-
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