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* 5XZf D EuCHH HtH H tDH[H1]A\l H H( s.H b.H ǀ HH audio library audio-channels object audio-channels audio-converter object audio-converter output samples %p %lu set samples %p %lu convert in %p, %p, %lu convert out %p, %p %lu pack %p, %p %lu GStreamer-Audio deinterleaving %p, %p %lu GstAudioConverter temp samples %p %lu alloc samples %d %lu %lu GstAudioFormatInfo GstAudioChannelMixerFlags GstAudioChannelPosition GstAudioConverterFlags GstAudioFormat GstAudioFormatFlags GstAudioPackFlags GstAudioLayout GstAudioFlags GstAudioDitherMethod GstAudioNoiseShapingMethod GstAudioQuantizeFlags GstAudioResamplerFilterMode GstAudioResamplerMethod GstAudioResamplerFlags GstAudioBaseSinkSlaveMethod GstAudioBaseSinkDiscontReason GstAudioBaseSrcSlaveMethod GstAudioCdSrcMode GstAudioRingBufferState GstAudioRingBufferFormatType Matrix for %d -> %d: } , { audio-channel-mixer object audio-channel-mixer mix->matrix != NULL mix != NULL mix %p, %p, %lu channels != 0 to != NULL from != NULL reorder_map != NULL MONO INVALID LFE1 FLoC FRoC LF2 TFL TFR TFC TRL TRR TSL TSR TRC BFC BFL BFR WL WR ] [ convert != NULL skipping empty buffer out != NULL res info->width <= 8 * 64 data != NULL dest != NULL copy %p, %p, %lu unpack %p, %p, %lu unpack in-place %p, %lu unpack to tmp %p, %lu get in samples %p convert->in_place len non-interleaved audio/x-raw layout ../gst-libs/gst/audio/audio.c trim < orig_samples GST_IS_BUFFER (buffer) timestamp %u:%02u:%02u.%09u trim %lu size %lu meta->offsets != NULL meta->info.channels <= 64 meta->info.channels > 0 cannot map memory range %u-%u GST_IS_BUFFER (gstbuffer) buffer != NULL free chain %p new rate %d -> %d resample %p %p,%lu %lu quantize %p, %p %lu GstAudioConverter.mix-matrix unpositioned channels chain %p: %d %d depth in %d, out %d convert F64 to S32 convert S32 to F64 unpack format %s to %s unitsizes: %d -> %d Invalid mix matrix out_info != NULL in_info != NULL Unknown audio Encoded audio 16-bit signed PCM audio 16-bit unsigned PCM audio 24-bit signed PCM audio 24-bit unsigned PCM audio 32-bit signed PCM audio 32-bit unsigned PCM audio 20-bit signed PCM audio 20-bit unsigned PCM audio 18-bit signed PCM audio 18-bit unsigned PCM audio 32-bit floating-point audio 64-bit floating-point audio mu-law a-law ima-adpcm gsm 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 alignment 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_FLAG_NONE GST_AUDIO_FLAG_UNPOSITIONED unpositioned GST_AUDIO_LAYOUT_INTERLEAVED GST_AUDIO_PACK_FLAG_NONE truncate-range GST_AUDIO_FORMAT_FLAG_INTEGER integer GST_AUDIO_FORMAT_FLAG_FLOAT float GST_AUDIO_FORMAT_FLAG_SIGNED signed GST_AUDIO_FORMAT_FLAG_COMPLEX complex GST_AUDIO_FORMAT_FLAG_UNPACK unpack GST_AUDIO_FORMAT_UNKNOWN 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 ../gst-libs/gst/audio/audio-converter.c GstAudioResamplerFilterInterpolation ../gst-libs/gst/audio/audio-channel-mixer.c out_channels > 0 && out_channels <= 64 in_channels > 0 && in_channels <= 64 format == GST_AUDIO_FORMAT_S16 || format == GST_AUDIO_FORMAT_S32 || format == GST_AUDIO_FORMAT_F32 || format == GST_AUDIO_FORMAT_F64 ../gst-libs/gst/audio/audio-channels.c no channel-mask property given Only partially valid channel mask 0x%016lx for %d channels converting %d channels for channel mask 0x%016lx check_valid_channel_positions (to, channels, FALSE, NULL) check_valid_channel_positions (from, channels, FALSE, NULL) check_valid_channel_positions (position, channels, FALSE, NULL) { F64LE, F64BE, F32LE, F32BE, S32LE, S32BE, U32LE, U32BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, S16LE, S16BE, U16LE, U16BE, S8, U8 } all->formats[i] != GST_AUDIO_FORMAT_UNKNOWN && all->formats[i] != GST_AUDIO_FORMAT_ENCODED ../gst-libs/gst/audio/audio-format.c format != GST_AUDIO_FORMAT_UNKNOWN (gint) format < G_N_ELEMENTS (formats) size % ((info->width * channels) / 8) == 0 info != NULL && info->width > 0 GST_AUDIO_FORMAT_INFO_IS_VALID_RAW (info) passthrough: %lu / %lu samples convert endian: %lu / %lu samples formats[i] != GST_AUDIO_FORMAT_UNKNOWN && formats[i] != GST_AUDIO_FORMAT_ENCODED (formats && len > 0) || (!formats && len == 0) Truncating %lu to %lu (trim start %lu, end %lu) samples == -1 || trim + samples <= orig_samples GST_BUFFER_OFFSET_IS_VALID (buffer) gst_audio_buffer_truncate failed segment->format == GST_FORMAT_TIME || segment->format == GST_FORMAT_DEFAULT channels == meta->info.channels gst_buffer_is_writable (buffer) ../gst-libs/gst/audio/audio-buffer.c plane %u, no memory at offset %lu meta->samples <= gst_buffer_get_size (gstbuffer) / GST_AUDIO_INFO_BPF (&meta->info) the GstAudioInfo argument is not equal to the GstAudioMeta's attached info (!meta && info->layout == GST_AUDIO_LAYOUT_INTERLEAVED) || (meta && info->layout == meta->info.layout) GST_AUDIO_INFO_FORMAT (info) != GST_AUDIO_FORMAT_UNKNOWN GST_AUDIO_INFO_IS_VALID (info) (in_rate == 0 && out_rate == 0) || convert->flags & GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE GstAudioConverter.resampler-method GstAudioConverter.dither-method GstAudioConverter.dither-threshold GstAudioConverter.noise-shaping-method same formats, same layout, no resampler and passthrough mixing -> passthrough same formats, and passthrough mixing -> only resampling unsupported sample width for endian conversion initializing 16-bit endian conversion initializing 24-bit endian conversion initializing 32-bit endian conversion initializing 64-bit endian conversion no resampler, passthrough mixing -> only endian conversion quantize to %d bits, dither %d, ns %d using no dither and noise shaping using dither %d and noise shaping %d mix format %s, passthrough %d, in_channels %d, out_channels %d Invalid mix matrix size, should be %d Invalid mix matrix row size, should be %d Invalid mix matrix element type, should be float flags ^ GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE 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 hxȜXHP`8(؝ȝxhP` gst_audio_make_raw_caps gst_audio_formats_raw generate_raw_audio_formats gst_audio_format_info_fill_silence gst_audio_format_get_info gst_audio_format_to_string gst_audio_format_from_string gst_audio_converter_convert gst_audio_converter_samples audio_chain_free gst_audio_converter_free check_mix_matrix chain_unpack chain_convert_in chain_mix chain_convert_out chain_quantize chain_pack setup_allocators gst_audio_converter_new converter_generic converter_endian converter_passthrough do_change_layout do_quantize do_convert_out do_resample do_mix do_convert_in audio_chain_set_samples do_unpack get_temp_samples get_output_samples gst_audio_converter_get_config gst_audio_converter_update_config gst_audio_channel_positions_to_string gst_audio_channel_get_fallback_mask 3 7 ? ? ? gst_audio_channel_positions_to_valid_order gst_audio_get_channel_reorder_map gst_audio_channel_positions_from_mask gst_audio_meta_reorder_channels gst_audio_buffer_reorder_channels gst_audio_reorder_channels
gst_audio_channel_mixer_samples gst_audio_channel_mixer_new gst_audio_channel_mixer_new_with_matrix gst_audio_buffer_map gst_audio_buffer_truncate gst_audio_buffer_clip audio_orc_double_to_s32 A audio_orc_s32_to_double A 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 > A ? ? D5?> audio-info object audio-info GstAudioInfo can't find exact taps Quality of Service Messages qos-messages GstAudioAggregatorPad 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 Ignore inactive pads ignore-inactive-pads Force live force-live Updated src caps to %pA ret=%d result %li no rate or bpf configured quant != NULL in != NULL || samples == 0 out != NULL || samples == 0 GstAudioResampler.n-taps GstAudioResampler.cubic-b GstAudioResampler.cubic-c GstAudioResampler.cutoff in_rate > 0 && out_rate > 0 options != NULL resampler != NULL avail %d < need %d out %d < samp_phase %d realloc %d -> %d property GstAudioAggregatorConvertPad Got %s event on src pad frames error: %s Pad is in EOS state pad->priv->buffer Buffer outside output segment skipping GAP buffer pushed outbuf, result = %s We're EOS Starting at offset %li Buffer duration: %u/%u position output-offset Invalid channel positions audio/ channel-mask no format given unknown format given no layout given no rate property given no channels property given unknown layout given parsing caps %pA gst_caps_is_fixed (caps) caps != NULL Rejecting invalid caps: %pA setting caps to %pA src caps negotiated %pA Got caps %pA Got %s event on sink pad info->finfo != NULL returned sink caps : %pA downstream caps %pA sink template caps : %pA Getting caps with filter %pA GstAudioResampler.filter-mode using resample function %d using full filter function using nearest filter function taps %d->%d, %d setting up filter cache Fc %f, A %f, tr_bw %f new phase %d/%d using exact phase divider divide by factor %d, gcd %d old phase %d/%d audio-resampler object audio-resampler sse sse2 sse41 GstAudioResampler.options enable SSE optimisations enable SSE2 optimisations enable SSE41 optimisations target %s, default flags %08x target flag %s in_rate > 0 Failed to update converter "S16"LE Fixated src caps to %pA ../gst-libs/gst/audio/audio-resampler.c gst_audio_aggregator_pad_flush_pad Emit QoS messages when dropping buffers 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 Output block size in nanoseconds, expressed as a fraction Output buffer duration fraction output-buffer-duration-fraction Timestamp alignment threshold in nanoseconds Window of time in nanoseconds to wait before creating a discontinuity Avoid timing out waiting for inactive pads Always operate in live mode and aggregate on timeout regardless of whether any live sources are linked upstream ../gst-libs/gst/audio/gstaudioaggregator.c GstAudioAggregatorSelectedSamplesInfo ../gst-libs/gst/audio/audio-info.c converting value %li from %s (%d) to %s (%d) format == GST_AUDIO_FORMAT_S32 GstAudioResampler.transition-bandwidth GstAudioResampler.stop-attenutation GstAudioResampler.filter-oversample quality <= GST_AUDIO_RESAMPLER_QUALITY_MAX out %d = ((%d * %d - %d) / %d) + 1 need %d = %d + %d + %d, avail %d = %d + %d not enough samples to start: need %lu, avail %lu, out %lu in %lu, avail %lu, consumed %lu %s:%d: invalid %s id %u for "%s" of type '%s' in '%s' seeking failed, unhandled seek type for start: %d seeking failed, unhandled seek type for end: %d seeking failed, unhandled seek format: %s Total duration in format %s: %u:%02u:%02u.%09u Starting to mix %u samples for offset %li with timestamp %u:%02u:%02u.%09u Timeout, missing %li frames (%u:%02u:%02u.%09u) Dropped buffer of %u samples at running time %u:%02u:%02u.%09u because input buffer is entirely before current output offset Dropped %u samples at running time %u:%02u:%02u.%09u because input buffer starts before current output offset Dropped %li samples at running time %u:%02u:%02u.%09u because input buffer is before output offset Buffer was late by %u:%02u:%02u.%09u, dropping %pA pushing outbuf %p, timestamp %u:%02u:%02u.%09u offset %li Unknown data received, not negotiated Failed to calculate the number of samples per buffer Got timeout before receiving any caps, don't output anything Dropping 0-sized buffer missing either a duration or a GAP flag: %pA Queued new buffer at offset %lu Buffer before segment or current position: %lu < %li Buffer resynced: Pad offset %lu, current audio aggregator offset %li Have discont. Expected %lu, got %lu start_time >= aggpad->segment.start Finished mixing buffer, waiting for next Mixing buffer for current offset 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 Samples per buffer: %u (error: %u/%u) Pad is at or after current offset: %lu >= %li GstAudioAggregatorPadNextSampleInfo channels <= 64 || position == NULL Invalid channel mask 0x%016lx for %d channels wrong name, expected audio/x-raw Sample rate %d can't be configured (downstream supported: %d, configured rate: %d) 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 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 info->finfo->format != GST_AUDIO_FORMAT_UNKNOWN first configured pad has sample rate %d GstAudioResampler.max-phase-error GstAudioResampler.filter-mode-threshold GstAudioResampler.filter-interpolation using interpolated filter function using filter interpolate function %d using linear interpolation for filter coefficients using cubic interpolation for filter coefficients allocate bps %d n_taps %d n_phases %d using linear interpolation to build filter using cubic interpolation to build filter automatically selected full filter, %d <= %d automatically selected full filter, memory %d <= %d automatically selected interpolated filter using n_taps %d cutoff %f oversample %d using Beta %f n_taps %d cutoff %f have new options, reconfigure filter phase %d out_rate %d, in_rate %d, gcd %d ../gst-libs/gst/audio/audio-resampler-x86.h method %d, bps %d, channels %d method >= GST_AUDIO_RESAMPLER_METHOD_NEAREST && method <= GST_AUDIO_RESAMPLER_METHOD_KAISER UNREACHABLE CODE: Unknown input format Creating output buffer with size %d ~('''(P(p')<)l)|))) ) gst_audio_aggregator_update_samples_per_buffer gst_audio_aggregator_fill_buffer gst_audio_aggregator_mix_buffer gst_audio_aggregator_aggregate gst_audio_aggregator_create_output_buffer gst_audio_aggregator_set_sink_caps gst_audio_aggregator_query_duration gst_audio_aggregator_sink_getcaps gst_audio_aggregator_convert_sink_getcaps gst_audio_aggregator_sink_event gst_audio_aggregator_src_event gst_audio_aggregator_negotiated_src_caps gst_audio_aggregator_fixate_src_caps gst_audio_aggregator_update_src_caps gst_audio_aggregator_sink_setcaps gst_audio_aggregator_convert_pad_update_converter 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_free calculate_kaiser_params alloc_taps_mem resampler_calculate_taps get_sample_bufs setup_functions gst_audio_resampler_update gst_audio_resampler_reset audio_resampler_check_x86 audio_resampler_init gst_audio_resampler_new Q ?\].? 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ףpMbX?R Q$C @ gst_audio_quantize_new gst_audio_info_convert gst_audio_info_to_caps gst_audio_info_from_caps gst_audio_info_set_format gst_audio_info_init ? ? @ A @ @ "@ (@ 2@ @ >@ ( H@ @ 8@-DT! @ @; %N?zD?1|?C ˅? !@ ? )@ N@ @ @qU? `GUտ?333333?Gz?qx? @U@ʡE?:0yE> I@ffffff!@K46? 5@?W2ı?Uj@+0?-DT!@HzG@*>;../gst-libs/gst/audio/gstaudioclock.c ../gst-libs/gst/audio/gstaudiobasesink.c Unexpected discontinuity in audio timestamps of %s%u:%02u:%02u.%09u, resyncing align with prev sample, ABS (%li) < %li ../gst-libs/gst/audio/gstaudiobasesrc.c binding text domain %s to locale dir %s ../gst-libs/gst/audio/gstaudiocdsrc.c cannot do this conversion, device not open source value not within allowed range converting value %li from %s into %s duration updated to %u:%02u:%02u.%09u seek segment %u:%02u:%02u.%09u-%u:%02u:%02u.%09u segment seek in track format is only supported in CONTINUOUS mode, not in mode %d cannot convert track %d to time start seek type is NONE, nothing to do is current track, just seeking back to start ignoring stop seek type (expected NONE) GST_CLOCK_TIME_IS_VALID (alignment_threshold) GST_CLOCK_TIME_IS_VALID (discont_wait) ../gst-libs/gst/audio/gstaudioringbuffer.c Could not mark ringbuffer as errored. It must have been stopped or already errored (was state %d) ringbuffer was not started, checking paused report latency min %u:%02u:%02u.%09u max %u:%02u:%02u.%09u Track %03u %u:%02u:%02u.%09u - %u:%02u:%02u.%09u track %02u: lsn %6u (%02u:%02u), length: %u seconds (%02u:%02u) i < sizeof (src->priv->mb_discid) + 1 total length = %u secs (%02u:%02u), added title lengths = %u seconds (%02u:%02u) last_sector = %u = %u secs (%02u:%02u) first_sector = %u = %u secs (%02u:%02u) unhandled query, chaining up to parent class position: sector %u, %li in format %s duration: %u sectors, %li in format %s seek in sector format not supported let base class handle seek in %s format segment %u:%02u:%02u.%09u-%u:%02u:%02u.%09u src->priv->cur_track >= 0 && src->priv->cur_track < src->priv->num_tracks pushing sector %d with timestamp %u:%02u:%02u.%09u EOS at sector %d, cur_track=%d, mode=%d adding track %2u (%2u) [%6u-%6u] [%5s], tags: %pA track %2u overlaps with previous tracks 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 last sample time %u:%02u:%02u.%09u Sink not negotiated before %s event. Got flow '%s' but can't return it: %d got different size than requested from sink pad: %u != %lu pulling %u bytes offset %lu to fill audio buffer 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 we are not negotiated, can't report latency yet peer or we are not live, don't care about latency Ringbuffer caps haven't changed, skipping reconfiguration processed samples: raw %lu, delay %u, real %lu, time %u:%02u:%02u.%09u 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, start %u:%02u:%02u.%09u, samples %u 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 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 custom slaving requested skew %c%u:%02u:%02u.%09u final timestamps: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u apply time offset %c%u:%02u:%02u.%09u dropping sample out of segment time %u:%02u:%02u.%09u, start %u:%02u:%02u.%09u sink received buffer of wrong size. Received GAP or ignoring audio for trickplay. Dropping contents start playback because we are at the end of segment preroll got interrupted: %d (%s) Buffer of size %lu has no time. Using render_start=%lu no sync needed. Using render_start=%lu no align possible: no previous sample position known no stop time align needed: we are slaved last_align %li driftsamples %u, next %lu correct clock skew %c%u:%02u:%02u.%09u > %c%u:%02u:%02u.%09u no custom slaving callback set - clock drift will not be compensated running: start %u:%02u:%02u.%09u - stop %u:%02u:%02u.%09u 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 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 create DISCONT of %lu samples at sample %lu ringbuffer was in error state, bailing out reading from sample %lu length %u reading from %d, we are at %d, sample %lu first sample, align to current %d audio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) 44100, channels = (int) 2 + - width depth endianness audioclock GstAudioBaseSink audiobasesink element audiobasesink GstAudioBaseSrc audiobasesrc element audiobasesrc /usr/share/locale gst-plugins-base-1.0 UTF-8 conversion failed: %s track number %d out of bounds returning %li unsupported format sample source value negative source time value invalid GstAudioCdSrc invalid track %li starting at sector %d seeking to track %li GstAudioClock GST_IS_AUDIO_BASE_SINK (sink) ringbuffer is errored subclass posted error GST_IS_AUDIO_BASE_SRC (src) Invalid track %u /dev/cdrom klass->close != NULL %02X %08X 00000000 musicbrainz-discid-full musicbrainz-discid track-count track-number audiocd-track-%03u This CD has no audio tracks failed to open device seek to track %d src->tags = %pA musicbrainz-discid-full = %s musicbrainz-discid = %s cddb discid = %s opening device %s klass->open != NULL position: track %d src->priv->tracks != NULL duration: %d tracks handling %s query let base class handle event seek failed: device not open handling %s event conversion failed seek'd to sector %d failed to read sector %u asking for sector %u announcing tags: %pA klass->read_sector != NULL cdda://%s#%d cdda://1 # Could not handle CDDA URI cannot handle URI '%s' successfully handled uri '%s' AUDIO DATA track->num > 0 track != NULL GST_IS_AUDIO_CD_SRC (src) 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 create failed open failed ringbuffer may start now READY->NULL PAUSED->READY NULL->READY READY->PAUSED PAUSED->PLAYING PLAYING->PAUSED ringbuffer not acquired clock provide disabled sink not negotiated. ringbuffer in wrong state drained audio we are flushing error %s latency query query convert time was %u:%02u:%02u.%09u cannot parse audio format. could not parse caps could not acquire ringbuffer activate ringbuffer release old ringbuffer actual-buffer-time actual-latency-time acquire new ringbuffer deactivating pull GstAudioSinkClock GstAudioSrcClock failed to payload. unknown slaving method %d ringbuffer not negotiated failed waiting for latency wrong size next sample expected at %lu wrote %u of %u rendering at %lu %d/%d dropping late sample resync after discont/resync Clipped start: %u/%u samples we have no clock we are not slaved Setting clock as master late preroll, adding jitter latency synced checking preroll Actual Buffer 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 ringbuffer is flushing forward event %p flush-start flush-stop refuse to seek alloc failed: %s warning: %s ringbuffer stopped continue playing wait playing read %u of %u dropped, align to segment %d at segment %d and sample %lu 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 cdda ;PpЅ 0 ,Ll܈ؓ`\|,,T,,,,,,,,|,,,,,, ,<,,,,,,,,,,,,,,,,,,,,,,,,,,, gst_audio_clock_init gst_audio_cd_src_create gst_audio_cd_src_stop gst_audio_cd_src_calculate_cddb_id gst_audio_cd_src_calculate_musicbrainz_discid gst_audio_cd_src_add_tags gst_audio_cd_src_make_toc gst_audio_cd_src_start gst_audio_cd_src_update_duration gst_audio_cd_src_add_track gst_audio_cd_src_uri_set_uri gst_audio_cd_src_handle_event gst_audio_cd_src_handle_track_seek gst_audio_cd_src_do_seek gst_audio_cd_src_query gst_audio_cd_src_convert __gst_audio_ring_buffer_set_errored gst_audio_base_src_post_message gst_audio_base_src_change_state gst_audio_base_src_get_offset gst_audio_base_src_create gst_audio_base_src_event gst_audio_base_src_query gst_audio_base_src_setcaps 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 gst_audio_base_src_get_time gst_audio_base_src_provide_clock _do_init gst_audio_base_sink_change_state gst_audio_base_sink_activate_pull gst_audio_base_sink_callback gst_audio_base_sink_sync_latency gst_audio_base_sink_resample_slaving gst_audio_base_sink_skew_slaving gst_audio_base_sink_custom_slaving gst_audio_base_sink_render gst_audio_base_sink_get_alignment gst_audio_base_sink_preroll gst_audio_base_sink_wait_event gst_audio_base_sink_drain gst_audio_base_sink_setcaps 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_get_time gst_audio_base_sink_query gst_audio_base_sink_query_pad gst_audio_base_sink_provide_clock result %u:%02u:%02u.%09u, last_time %u:%02u:%02u.%09u 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 ../gst-libs/gst/audio/gstaudiodecoder.c not copying memory specific metadata %s transformed metadata %s: copy: %d ../gst-libs/gst/audio/gstaudioencoder.c seeking to %u:%02u:%02u.%09u at byte offset %li unsupported seek: not flushing cannot convert start/stop for seek reset clock to %u:%02u:%02u.%09u, last %u:%02u:%02u.%09u, offset %c%u:%02u:%02u.%09u 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) Max consecutive decoder errors before returning flow error 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 Subclass failed to decide allocation klass->decide_allocation != NULL didn't get downstream ALLOCATION hints caps != NULL && gst_caps_is_fixed (caps) GST_IS_CAPS (dec->priv->ctx.caps) GST_AUDIO_INFO_IS_VALID (&dec->priv->ctx.info) discarding non-TIME SEEKING query min_latency:%u:%02u:%02u.%09u max_latency:%u:%02u:%02u.%09u GST_CLOCK_TIME_IS_VALID (tolerance) tags == NULL || GST_IS_TAG_LIST (tags) Failed to negotiate, fallback allocation couldn't allocate output buffer 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 pushing buffer of size %lu with ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u Can't copy metadata because input buffer disappeared received more encoded samples %d than provided %d as inputs new upstream ts %u:%02u:%02u.%09u at distance %lu new upstream ts only introduces jitter buf == NULL || gst_buffer_get_size (buf) > 0 no more data drained from subclass bypassing subclass with leftover providing subclass with %d bytes at offset %d non-drainable and no more data available: %d, needed: %d, force: %d query %p: our time: %u:%02u:%02u.%09u Peer latency: live %d, min %u:%02u:%02u.%09u max %u:%02u:%02u.%09u query %p: we return %li (format %u) gst_audio_encoder_sink_activate_mode Requested output format %pA do not match template %pA 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) output buffer of size %lu with ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u buffer %d ms apart from current fragment buffer size %lu not a multiple of %d leading frame ts %u:%02u:%02u.%09u Received decoded subframe, but no pending frame buffer is %lu samples past base_ts %u:%02u:%02u.%09u, expected ts %u:%02u:%02u.%09u GstAudioMeta samples (%lu) are inconsistent with the buffer size and layout (size/bpf = %lu) Can't copy metadata because input buffers disappeared GST_CLOCK_TIME_IS_VALID (priv->base_ts) received more decoded frames %d than provided %d buf == NULL || GST_AUDIO_INFO_IS_VALID (&ctx->info) accepting %lu bytes == %lu samples for %d frames Chose default caps %pA for initial gap No default channel-mask for %d channels Decoder output not negotiated before GAP event. gap event: plc %d, do_plc %d, position %u:%02u:%02u.%09u duration %u:%02u:%02u.%09u Failed to negotiate with downstream converted to TIME start %u:%02u:%02u.%09u No valid frames decoded before end of stream received STREAM_START. Clearing taglist Caps did not change, not setting again tracking frame size %lu, ts %u:%02u:%02u.%09u Skipping decode in trickmode and sending gap providing subclass with NULL frame frame at offset %d of length %d decoding buffer %p, ts %u:%02u:%02u.%09u 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 buffer did not decode, keeping GST_CLOCK_TIME_IS_VALID (duration) received buffer of size %lu with ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u ... but restoring previous ts tracking new audio format identical to configured format subclass did not accept 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 audio encoder base class audioencoder gst_audio_encoder_finalize Perfect Timestamps perfect-timestamp Granule Marking mark-granule Hard Resync hard-resync merged : %pA mode : %d decoder : %pA upstream : %pA copy metadata %s unsupported seek: rate unsupported seek: start time unsupported seek: end time new base gp %li no time, reuse last GstAudioDecoder audio decoder base class audiodecoder Minimum Latency min-latency Packet Loss Concealment plc Max errors max-errors GST_IS_AUDIO_DECODER (object) received event %d, %s received query %pA starting segment %pB ALLOCATION (%d) params: %pA GST_IS_PAD (pad) Pushing pending events setting src caps %pA GST_IS_AUDIO_DECODER (dec) Returning caps %pA handling query: %pA min <= max GST_CLOCK_TIME_IS_VALID (min) enabled: %d GST_CLOCK_TIME_IS_VALID (num) setting decoder tags to %pA alloc src buffer size > 0 GstAudioEncoder Failed to open codec Failed to close codec encoder : %pA no caps set pushing header returned %s marking discont subclass returned %s, buf %p Pushing early pending events buffer pushed: %s out samples %d taking %lu bytes for output Sending headers marked discont ts diff %d ms distance % ctx->info.bpf == 0 priv->offset <= av klass->handle_frame != NULL clock-type name returning peer response Ignoring BYTES position query upstream size %li gst_audio_decoder_reset init ok srcpad created pad_template != NULL gst_audio_decoder_src_event gst_audio_decoder_src_query sinkpad created gst_audio_decoder_sink_event gst_audio_decoder_chain gst_audio_decoder_sink_query gst_audio_decoder_init Failed to start codec Failed to stop codec gst_audio_decoder_stop gst_audio_decoder_start reset full %d src created gst_audio_encoder_src_event gst_audio_encoder_src_query gst_audio_encoder_sink_event gst_audio_encoder_sink_query gst_audio_encoder_chain gst_audio_encoder_init invalid output format Setting srcpad caps %pA ctx->info.bpf != 0 buffer queued assembling fragment forcing fragment flush starting new fragment adding to fragment base_ts now %u:%02u:%02u.%09u base_ts resync !is_subframe || buf != NULL frames != 0 peer caps %pA no valid frames found Enqueuing event %d, %s received TIME SEGMENT %pB unsupported format; ignoring received SEGMENT %pB upstream stream tags: %pA failed to parse stream done pushing to subclass no frame yet av >= 0 ts == prev_ts; discarding offset + len <= av skipped %d; setting DISCONT parsing available: %d chain-done discarding empty buffer decoded buffer to %p tracking ts %u:%02u:%02u.%09u flushing buffers to decoder received discont handling discont decoder not initialized rejected caps %pA klass->set_format != NULL new base ts %u:%02u:%02u.%09u encoder not initialized chain leaving gst_audio_encoder_sink_setcaps gst_audio_encoder_chain gst_audio_encoder_set_base_gp gst_audio_encoder_push_buffers gst_audio_encoder_push_early_pending_events gst_audio_encoder_push_pending_events gst_audio_encoder_finish_frame foreach_metadata gst_audio_encoder_create_merged_tags_event gst_audio_encoder_push_event gst_audio_encoder_change_state gst_audio_encoder_reset gst_audio_encoder_init gst_audio_decoder_get_allocator gst_audio_decoder_allocate_output_buffer 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_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_start gst_audio_decoder_stop gst_audio_decoder_change_state gst_audio_decoder_src_query gst_audio_decoder_src_query_default gst_audio_decoder_sink_query gst_audio_decoder_sink_getcaps gst_audio_decoder_sink_query_default gst_audio_decoder_src_event gst_audio_decoder_do_seek gst_audio_decoder_src_eventfunc gst_audio_decoder_sink_event gst_audio_decoder_sink_setcaps gst_audio_decoder_sink_eventfunc gst_audio_decoder_handle_gap gst_audio_decoder_negotiate_default_caps gst_audio_decoder_chain gst_audio_decoder_flush_decode gst_audio_decoder_chain_reverse gst_audio_decoder_chain_forward gst_audio_decoder_push_buffers gst_audio_decoder_handle_frame gst_audio_decoder_finish_frame_or_subframe gst_audio_decoder_finish_frame gst_audio_decoder_finish_subframe foreach_metadata send_pending_events gst_audio_decoder_output gst_audio_decoder_push_forward gst_audio_decoder_set_output_caps gst_audio_decoder_set_output_format gst_audio_decoder_negotiate gst_pad_set_caps gst_audio_decoder_negotiate_default gst_audio_decoder_push_event gst_audio_decoder_create_merged_tags_event gst_audio_decoder_finalize gst_audio_decoder_reset gst_audio_decoder_init gst_audio_clock_get_time gst_audio_clock_get_internal_time gst_audio_clock_reset gst_audio_filter_get_unit_size 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_resume gst_audio_sink_ring_buffer_stop gst_audio_sink_ring_buffer_delay gst_audio_sink_ring_buffer_activate gst_audio_sink_ring_buffer_clear_all 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-libs/gst/audio/gstaudiosrc.c GST_IS_AUDIO_FILTER_CLASS (klass) ../gst-libs/gst/audio/gstaudioiec61937.c dst_n >= gst_audio_iec61937_frame_size (spec) format != GST_FORMAT_UNDEFINED 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 info->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED || !offsets ../gst-libs/gst/audio/gstaudiosink.c gst_audio_src_create_ringbuffer parsed caps: type: %d, '%s' acquire ringbuffer: segment size: %d bytes = %d samples acquire ringbuffer: buffer size: %d bytes = %d samples acquire ringbuffer: latency segments: %d acquire ringbuffer: total segments: %d acquire ringbuffer: latency time: %li usec acquire ringbuffer: buffer time: %li usec GST_IS_AUDIO_RING_BUFFER (buf) Device for ring buffer %p already open, fix your code Device for ring buffer already open Device for ring buffer %p already closed, fix your code Resources for ring buffer %p still acquired Device for ring buffer already closed Resources for ring buffer still acquired was not paused, must have been started write @%p seg %d, sps %d, off %d, avail %d pointer at %d, write to %d-%d, diff %d, segtotal %d, segsize %d, base %d was not paused or errored, must have been stopped Retrieved timestamp %u:%02u:%02u.%09u @ %d read @%p seg %d, off %d, sampleslen %d pointer at %d, sample %lu, read from %d-%d, to_read %d, diff %d, segtotal %d, segsize %d prepare read from segment %d (real %d) @%p Storing timestamp %u:%02u:%02u.%09u @ %d no write function, exit thread error reading data %d (reason: %s), skipping segment transferred %d bytes of %d to segment %d perfect-timestamp can not be set FALSE while granule handling is enabled !enc->priv->granule || enc->priv->perfect_ts tags == NULL || mode != GST_TAG_MERGE_UNDEFINED GST_IS_CAPS (enc->priv->ctx.caps) Freeing timestamp buffer, %d entries ../gst-libs/gst/audio/gstaudiofilter.c Got buffer, but not negotiated yet! invalid bytes_per_frame from acquire ringbuffer %p, fix the element Allocating an array for %d timestamps single channel, no need to reorder position-less channels, no need to reorder may have to reorder channels: %s -> %s could not create thread for unknown reason error writing data in %s (reason: %s), skipping segment (left: %d, written: %d) transferred %d bytes of %d from segment %d gst_audio_sink_create_ringbuffer gst_audio_filter_change_state gst_audio_filter_set_caps ringbuffer class default_clear_all default_commit GstAudioFilter audiofilter GstAudioRingBuffer was not started (state %d) failed to pause signal waiter pausing ringbuffer start, sending signal returning caps %pA new headers %p GST_IS_CAPS (allowed_caps) frame-size block-size mpegaudioversion layer iec61937 version > 0 && layer > 0 src_n >= 6 src_n != 0 src_n >= 7 src != dst dst != NULL src != NULL GstAudioDownmixMetaAPI GstAudioDownmixMeta to_channels > 0 to_position != NULL from_channels > 0 from_position != NULL GstAudioClippingMetaAPI GstAudioClippingMeta GstAudioMetaAPI GstAudioMeta GstAudioLevelMetaAPI GstAudioLevelMeta GstAudioSinkRingBuffer created ringbuffer @%p creating ringbuffer GstAudioSrcRingBuffer spec caps: %p %pA failed opening device opened device !buf->acquired error closing device closed device Device for %p not acquired device not acquired device was active in mode %d failed to activate device we are not acquired we may not start failed to start resuming was not stopped, try paused starting ringbuffer not allowed to start stopped processing waiting.. start! copy %u bytes fwd_up end %d/%d fwd_down end %d/%d rev_up end %d/%d rev_down end %d/%d write %d : %d buf->memory != NULL failed to stop was not paused, try errored was not started, try paused stopping set sample to %lu, segbase %d len != NULL readptr != NULL segment != NULL clear segment %d @%p clear all segments may start: %d reset done reset... posting LEAVE stream status stop running, exit thread continue running got signal wait for action signal wait posting ENTER stream status enter thread audio-codec video-codec subtitle-codec container-format nominal-bitrate maximum-bitrate minimum-bitrate encoder-version event result %d received query %d, %s sink activate push return: %d activate return: %d activate %d sink activate push %d GST_IS_AUDIO_ENCODER (enc) set to %d set to %u:%02u:%02u.%09u setting encoder tags to %pA refused caps %pA device was released failed to release device released device buf->open releasing device couldn't parse %pA configured caps: %pA info: %d audio/x-alaw audio/x-mulaw audio/x-iec958 audio/x-ac3 audio/x-eac3 audio/x-dts audio/mpeg mpegversion stream-format adts audio/x-flac spec->latency_time != 0 Device for %p not opened device not opened device was acquired failed to acquire device acquired device acquiring device %p audiosrc-ringbuffer GstAudioSink audiosink element audiosink could not open device could not close device could not prepare device could not unprepare device unprepared pause done pause... resume done resume... stop done stop... audiosink-ringbuffer could not create thread %s thread is started waiting for thread starting thread clear all mu law a law ima adpcm aac mpeg2 aac mpeg4 aac mpeg2 raw aac mpeg4 raw X@ ``gst_audio_src_ring_buffer_stop gst_audio_src_ring_buffer_start audioringbuffer_thread_func gst_audio_sink_create_ringbuffer gst_audio_sink_ring_buffer_clear_all gst_audio_sink_ring_buffer_stop gst_audio_sink_ring_buffer_resume gst_audio_sink_ring_buffer_pause gst_audio_sink_ring_buffer_start gst_audio_sink_ring_buffer_release gst_audio_sink_ring_buffer_activate gst_audio_sink_ring_buffer_acquire gst_audio_sink_ring_buffer_close_device gst_audio_sink_ring_buffer_open_device audioringbuffer_thread_func gst_audio_ring_buffer_set_timestamp gst_audio_ring_buffer_set_channel_positions gst_audio_ring_buffer_may_start gst_audio_ring_buffer_clear gst_audio_ring_buffer_advance gst_audio_ring_buffer_prepare_read gst_audio_ring_buffer_read gst_audio_ring_buffer_commit default_commit wait_segment gst_audio_ring_buffer_clear_all default_clear_all gst_audio_ring_buffer_set_sample gst_audio_ring_buffer_samples_done gst_audio_ring_buffer_delay gst_audio_ring_buffer_stop gst_audio_ring_buffer_pause gst_audio_ring_buffer_pause_unlocked gst_audio_ring_buffer_start gst_audio_ring_buffer_is_flushing gst_audio_ring_buffer_set_flushing gst_audio_ring_buffer_is_active gst_audio_ring_buffer_activate gst_audio_ring_buffer_is_acquired gst_audio_ring_buffer_release gst_audio_ring_buffer_acquire gst_audio_ring_buffer_device_is_open gst_audio_ring_buffer_close_device gst_audio_ring_buffer_open_device gst_audio_ring_buffer_set_callback_full gst_audio_ring_buffer_parse_caps gst_audio_ring_buffer_debug_spec_buff gst_audio_ring_buffer_debug_spec_caps gst_buffer_add_audio_level_meta gst_buffer_add_audio_meta gst_buffer_add_audio_clipping_meta gst_buffer_add_audio_downmix_meta gst_audio_iec61937_payload gst_audio_filter_class_add_pad_templates gst_audio_filter_submit_input_buffer gst_audio_filter_set_caps gst_audio_encoder_get_allocator gst_audio_encoder_allocate_output_buffer gst_audio_encoder_set_output_format gst_audio_encoder_negotiate gst_pad_set_caps gst_audio_encoder_negotiate_default 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_get_hard_resync gst_audio_encoder_set_hard_resync gst_audio_encoder_get_perfect_timestamp gst_audio_encoder_set_perfect_timestamp gst_audio_encoder_get_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_get_lookahead gst_audio_encoder_set_lookahead gst_audio_encoder_get_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_get_frame_samples_min gst_audio_encoder_set_frame_samples_min gst_audio_encoder_get_audio_info gst_audio_encoder_activate gst_audio_encoder_sink_activate_mode gst_audio_encoder_set_property gst_audio_encoder_src_query gst_audio_encoder_src_query_default gst_audio_encoder_src_event gst_audio_encoder_sink_query gst_audio_encoder_sink_event gst_audio_encoder_sink_event_default gst_audio_encoder_getcaps_default Linear volume factor, 1.0=100% Mute the audio channel without changing the volume ../gst-libs/gst/audio/gstaudioutilsprivate.c not enough metadata yet to convert ../gst-libs/gst/audio/streamvolume.c ../gst-libs/gst/audio/gstaudiostreamalign.c still %d frames left after draining audio decoder push buffers failed volume Mute mute GstAudioStreamAlign align != NULL channels-mask proxy caps %pA intersecting with %pA allowed caps %pA template caps %pA GstStreamVolume GST_IS_STREAM_VOLUME (volume) val >= 0.0 rate != 0 ... really draining GstAudioSrc audiosrc element audiosrc klass->convert_buffer audio decoder output failed Applying pending segments flush hard %d *hH *`@ gst_audio_decoder_flush apply_pending_events gst_audio_decoder_drain gst_audio_encoder_drain gst_audio_aggregator_convert_buffer 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 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 __gst_audio_encoded_audio_convert 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