| 1 | ffmpeg started on 2017-08-29 at 14:36:10
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| 2 | Report written to "live-rtp-g711-toMP3-REPORT.log"
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| 3 | Command line:
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| 4 | ffmpeg -re -f mulaw -i rtp://10.200.1.14:32760 -sample_rate 44100 -filter:a "asetrate=9000" -acodec libmp3lame -ar 44100 "c:\\test\\live-rtp-g711-toMP3.mp3" -report
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| 5 | ffmpeg version N-86996-g931c0ac95c-Reino Copyright (c) 2000-2017 the FFmpeg developers
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| 6 | built with gcc 7.1.0 (GCC)
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| 7 | configuration: --arch=x86_64 --target-os=mingw32 --cross-prefix=/Users/scuser/Development/ffmpegwinbuildDEBUG/sandbox/cross_compilers/mingw-w64-x86_64/bin/x86_64-w64-mingw32- --pkg-config=pkg-config --pkg-config-flags=--static --extra-version=Reino --enable-gray --enable-version3 --disable-debug --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-w32threads --enable-fontconfig --enable-gmp --enable-gnutls --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libflite --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libzimg --enable-libzvbi --enable-pt libavutil 55. 74.100 / 55. 74.100
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| 8 | libavcodec 57.102.100 / 57.102.100
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| 9 | libavformat 57. 76.100 / 57. 76.100
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| 10 | libavdevice 57. 7.100 / 57. 7.100
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| 11 | libavfilter 6. 99.100 / 6. 99.100
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| 12 | libavresample 3. 6. 0 / 3. 6. 0
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| 13 | libswscale 4. 7.102 / 4. 7.102
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| 14 | libswresample 2. 8.100 / 2. 8.100
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| 15 | libpostproc 54. 6.100 / 54. 6.100
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| 16 | Splitting the commandline.
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| 17 | Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
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| 18 | Reading option '-f' ... matched as option 'f' (force format) with argument 'mulaw'.
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| 19 | Reading option '-i' ... matched as input url with argument 'rtp://10.200.1.14:32760'.
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| 20 | Reading option '-sample_rate' ... matched as AVOption 'sample_rate' with argument '44100'.
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| 21 | Reading option '-filter:a' ... matched as option 'filter' (set stream filtergraph) with argument 'asetrate=9000'.
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| 22 | Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'libmp3lame'.
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| 23 | Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100'.
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| 24 | Reading option 'c:\test\live-rtp-g711-toMP3.mp3' ... matched as output url.
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| 25 | Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
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| 26 | Finished splitting the commandline.
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| 27 | Parsing a group of options: global .
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| 28 | Applying option report (generate a report) with argument 1.
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| 29 | Successfully parsed a group of options.
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| 30 | Parsing a group of options: input url rtp://10.200.1.14:32760.
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| 31 | Applying option re (read input at native frame rate) with argument 1.
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| 32 | Applying option f (force format) with argument mulaw.
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| 33 | Successfully parsed a group of options.
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| 34 | Opening an input file: rtp://10.200.1.14:32760.
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| 35 | [mulaw @ 0000002c730d7540] Opening 'rtp://10.200.1.14:32760' for reading
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| 36 | [rtp @ 0000002c730d7c00] No default whitelist set
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| 37 | [udp @ 0000002c730d7e80] No default whitelist set
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| 38 | [udp @ 0000002c730d7e80] end receive buffer size reported is 65536
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| 39 | [udp @ 0000002c748241e0] No default whitelist set
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| 40 | [udp @ 0000002c748241e0] end receive buffer size reported is 65536
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| 41 |
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| 42 | Received windows signal 0
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| 43 | [mulaw @ 0000002c730d7540] Before avformat_find_stream_info() pos: 0 bytes read:0 seeks:0 nb_streams:1
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| 44 | [mulaw @ 0000002c730d7540] interrupted
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| 45 | [mulaw @ 0000002c730d7540] After avformat_find_stream_info() pos: 0 bytes read:0 seeks:0 frames:0
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| 46 | Guessed Channel Layout for Input Stream #0.0 : mono
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| 47 | Input #0, mulaw, from 'rtp://10.200.1.14:32760':
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| 48 | Duration: N/A, bitrate: 352 kb/s
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| 49 | Stream #0:0, 0, 1/44100: Audio: pcm_mulaw, 44100 Hz, mono, s16, 352 kb/s
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| 50 | Successfully opened the file.
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| 51 | Parsing a group of options: output url c:\test\live-rtp-g711-toMP3.mp3.
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| 52 | Applying option filter:a (set stream filtergraph) with argument asetrate=9000.
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| 53 | Applying option acodec (force audio codec ('copy' to copy stream)) with argument libmp3lame.
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| 54 | Applying option ar (set audio sampling rate (in Hz)) with argument 44100.
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| 55 | Successfully parsed a group of options.
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| 56 | Opening an output file: c:\test\live-rtp-g711-toMP3.mp3.
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| 57 | [file @ 0000002c730d9f40] Setting default whitelist 'file,crypto'
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| 58 | Successfully opened the file.
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| 59 | Stream mapping:
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| 60 | Stream #0:0 -> #0:0 (pcm_mulaw (native) -> mp3 (libmp3lame))
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| 61 | Press [q] to stop, [?] for help
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| 62 | Finishing stream 0:0 without any data written to it.
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| 63 | detected 2 logical cores
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| 64 | [Parsed_asetrate_0 @ 0000002c730da4a0] Setting 'sample_rate' to value '9000'
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| 65 | [graph_0_in_0_0 @ 0000002c730da8e0] Setting 'time_base' to value '1/44100'
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| 66 | [graph_0_in_0_0 @ 0000002c730da8e0] Setting 'sample_rate' to value '44100'
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| 67 | [graph_0_in_0_0 @ 0000002c730da8e0] Setting 'sample_fmt' to value 's16'
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| 68 | [graph_0_in_0_0 @ 0000002c730da8e0] Setting 'channel_layout' to value '0x4'
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| 69 | [graph_0_in_0_0 @ 0000002c730da8e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x4
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| 70 | [format_out_0_0 @ 0000002c730db1e0] Setting 'sample_fmts' to value 's32p|fltp|s16p'
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| 71 | [format_out_0_0 @ 0000002c730db1e0] Setting 'sample_rates' to value '44100'
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| 72 | [format_out_0_0 @ 0000002c730db1e0] Setting 'channel_layouts' to value '0x4|0x3'
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| 73 | [format_out_0_0 @ 0000002c730db1e0] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_asetrate_0' and the filter 'format_out_0_0'
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| 74 | [AVFilterGraph @ 0000002c74898be0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
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| 75 | [auto_resampler_0 @ 0000002c730dba80] picking s16p out of 3 ref:s16
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| 76 | [auto_resampler_0 @ 0000002c730dba80] [SWR @ 0000002c74898fa0] Using s16p internally between filters
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| 77 | [auto_resampler_0 @ 0000002c730dba80] ch:1 chl:mono fmt:s16 r:9000Hz -> ch:1 chl:mono fmt:s16p r:44100Hz
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| 78 | Output #0, mp3, to 'c:\test\live-rtp-g711-toMP3.mp3':
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| 79 | Metadata:
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| 80 | TSSE : Lavf57.76.100
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| 81 | Stream #0:0, 0, 1/44100: Audio: mp3 (libmp3lame), 44100 Hz, mono, s16p
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| 82 | Metadata:
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| 83 | encoder : Lavc57.102.100 libmp3lame
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| 84 | size= 0kB time=00:00:00.00 bitrate=N/A speed= 0x
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| 85 | video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
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| 86 | Input file #0 (rtp://10.200.1.14:32760):
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| 87 | Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
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| 88 | Total: 0 packets (0 bytes) demuxed
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| 89 | Output file #0 (c:\test\live-rtp-g711-toMP3.mp3):
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| 90 | Output stream #0:0 (audio): 0 frames encoded (0 samples); 0 packets muxed (0 bytes);
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| 91 | Total: 0 packets (0 bytes) muxed
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| 92 | Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
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| 93 | 0 frames successfully decoded, 0 decoding errors
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| 94 | [AVIOContext @ 0000002c74898020] Statistics: 0 seeks, 1 writeouts
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| 95 | [AVIOContext @ 0000002c748444c0] Statistics: 0 bytes read, 0 seeks
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| 96 | Exiting normally, received signal 2.
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