1. 07 Oct, 2008 1 commit
  2. 03 Oct, 2008 4 commits
  3. 01 Oct, 2008 1 commit
    • Michael Smith's avatar
      configure.ac: Fix libs for linking directsound. · e2dbf108
      Michael Smith authored
      Original commit message from CVS:
      * configure.ac:
      Fix libs for linking directsound.
      * sys/directsound/gstdirectsoundsink.c:
      Fix buffer sizing to prevent racing the ringbuffer at startup.
      Add volume property.
      e2dbf108
  4. 27 Sep, 2008 1 commit
  5. 26 Sep, 2008 2 commits
    • Wim Taymans's avatar
      gst/rtp/gstrtpamrdepay.c: Mark DISCONT on output buffers when the marker bit... · b17599a2
      Wim Taymans authored
      gst/rtp/gstrtpamrdepay.c: Mark DISCONT on output buffers when the marker bit signals a new talk spurt.
      
      Original commit message from CVS:
      * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init),
      (gst_rtp_amr_depay_process):
      Mark DISCONT on output buffers when the marker bit signals a new talk
      spurt.
      * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
      Set the marker bit for buffers with a DISCONT flag to signal a talk
      spurt.
      b17599a2
    • Wim Taymans's avatar
      gst/rtp/: Added MP4A-LATM payloader to match the depayloader. · c77bfaac
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/Makefile.am:
      * gst/rtp/gstrtp.c: (plugin_init):
      * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type),
      (gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init),
      (gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize),
      (gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps),
      (gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer),
      (gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init):
      * gst/rtp/gstrtpmp4apay.h:
      Added MP4A-LATM payloader to match the depayloader.
      c77bfaac
  6. 25 Sep, 2008 4 commits
  7. 23 Sep, 2008 2 commits
  8. 17 Sep, 2008 2 commits
    • Edward Hervey's avatar
      gst/qtdemux/qtdemux.c: Some 'broken' files out there have atom lengths of... · 53a576bb
      Edward Hervey authored
      gst/qtdemux/qtdemux.c: Some 'broken' files out there have atom lengths of zero... which basically results in qtdemux ...
      
      Original commit message from CVS:
      * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header),
      (gst_qtdemux_chain):
      Some 'broken' files out there have atom lengths of zero...
      which basically results in qtdemux consuming that atom again and again
      until the *end of night* !
      Detect that and emits an adequate element error message.
      53a576bb
    • Jan Schmidt's avatar
      gst/: Fix build flags order. · a236a2df
      Jan Schmidt authored
      Original commit message from CVS:
      * gst/interleave/Makefile.am:
      * gst/matroska/Makefile.am:
      Fix build flags order.
      * tests/check/elements/audioamplify.c: (GST_START_TEST):
      * tests/check/elements/audiodynamic.c: (GST_START_TEST):
      * tests/check/elements/audioinvert.c: (GST_START_TEST):
      * tests/check/elements/audiopanorama.c: (GST_START_TEST):
      Format fixes.
      * tests/check/elements/multifile.c:
      Pull in unistd.h
      a236a2df
  9. 15 Sep, 2008 2 commits
    • Wim Taymans's avatar
      gst/rtp/gstrtpmp4gdepay.*: Handle interleaved streams by reordering AU in a queue. · 1c6a371d
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_init),
      (gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_setcaps),
      (gst_rtp_mp4g_depay_clear_queue), (gst_rtp_mp4g_depay_flush_queue),
      (gst_rtp_mp4g_depay_queue), (gst_rtp_mp4g_depay_process),
      (gst_rtp_mp4g_depay_change_state):
      * gst/rtp/gstrtpmp4gdepay.h:
      Handle interleaved streams by reordering AU in a queue.
      1c6a371d
    • Wim Taymans's avatar
      gst/rtp/gstrtpmp4gdepay.c: Change some of the ranges in the caps, mostly for... · fe9b4496
      Wim Taymans authored
      gst/rtp/gstrtpmp4gdepay.c: Change some of the ranges in the caps, mostly for the amount of bits we can use.
      
      Original commit message from CVS:
      * gst/rtp/gstrtpmp4gdepay.c: (gst_bs_parse_init),
      (gst_bs_parse_read), (gst_rtp_mp4g_depay_process):
      Change some of the ranges in the caps, mostly for the amount of bits we
      can use.
      Added a little bitstream parse and use it to parse the AU header fields.
      Check for malformed and wrongly sized packets better.
      Implement more header field parsing.
      Handle the size of fragmented packets correctly.
      fe9b4496
  10. 14 Sep, 2008 1 commit
  11. 11 Sep, 2008 1 commit
  12. 04 Sep, 2008 1 commit
    • Tim-Philipp Müller's avatar
      ext/flac/gstflacenc.c: Make sure the desired default values are actually set,... · 9a120212
      Tim-Philipp Müller authored
      ext/flac/gstflacenc.c: Make sure the desired default values are actually set, not only registered as defaults (actual...
      
      Original commit message from CVS:
      * ext/flac/gstflacenc.c: (gst_flac_enc_class_init):
      Make sure the desired default values are actually set, not only
      registered as defaults (actual problem is that the stereo-specific
      values are only updated if channels==2, which is not the case yet
      when the object is created, so the default values for the
      mid-side-stereo and loose-mid-side-stereo settings are never
      set in _update_quality()). Makes flacenc create smaller files by
      default (for stereo input), and fixes #550791.
      9a120212
  13. 03 Sep, 2008 2 commits
  14. 02 Sep, 2008 3 commits
    • Wim Taymans's avatar
      gst/qtdemux/qtdemux.c: Add mapping for IMA Loki SDL MJPEG ADPCM codec. · 105e0023
      Wim Taymans authored
      Original commit message from CVS:
      * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
      Add mapping for IMA Loki SDL MJPEG ADPCM codec.
      Add some alternative byteswapped mappings that seem to pop up sometimes.
      Fixes #550288.
      105e0023
    • Tim-Philipp Müller's avatar
      po/: Add 'ca' to LINGUAS; add some more files with translations and some files... · cb434b2b
      Tim-Philipp Müller authored
      po/: Add 'ca' to LINGUAS; add some more files with translations and some files which should be ignored by translation...
      
      Original commit message from CVS:
      * po/LINGUAS:
      * po/POTFILES.in:
      * po/POTFILES.skip:
      Add 'ca' to LINGUAS; add some more files with translations and some
      files which should be ignored by translation tools.
      cb434b2b
    • Sebastian Dröge's avatar
      ext/speex/: Use integer encoding and decoding functions instead of converting... · 3fa17e67
      Sebastian Dröge authored
      ext/speex/: Use integer encoding and decoding functions instead of converting the integer input to float in the eleme...
      
      Original commit message from CVS:
      * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
      * ext/speex/gstspeexdec.h:
      * ext/speex/gstspeexenc.c: (gst_speex_enc_encode):
      * ext/speex/gstspeexenc.h:
      Use integer encoding and decoding functions instead of converting
      the integer input to float in the element. The libspeex integer
      functions are doing this for us already or, if libspeex was compiled
      in integer mode, they're doing everything using integer arithmetics.
      Also saves some copying around.
      3fa17e67
  15. 01 Sep, 2008 1 commit
  16. 31 Aug, 2008 4 commits
    • Sebastian Dröge's avatar
      ext/wavpack/gstwavpackenc.*: Handle non-zero start timestamps and stream... · 912cb980
      Sebastian Dröge authored
      ext/wavpack/gstwavpackenc.*: Handle non-zero start timestamps and stream discontinuities correctly. This only has an ...
      
      Original commit message from CVS:
      * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
      (gst_wavpack_enc_push_block), (gst_wavpack_enc_chain):
      * ext/wavpack/gstwavpackenc.h:
      Handle non-zero start timestamps and stream discontinuities
      correctly. This only has an effect if we're muxing into
      a container format as the raw WavPack stream must contain
      continous sample numbers.
      912cb980
    • Sebastian Dröge's avatar
      ext/speex/gstspeexenc.c: Correct the timestamp and granulepos calculation by one Speex frame. · a414f86f
      Sebastian Dröge authored
      Original commit message from CVS:
      * ext/speex/gstspeexenc.c: (gst_speex_enc_encode):
      Correct the timestamp and granulepos calculation by one Speex
      frame.
      a414f86f
    • Sebastian Dröge's avatar
      ext/speex/gstspeexdec.c: Correctly take the granulepos from upstream if... · 25896b3a
      Sebastian Dröge authored
      ext/speex/gstspeexdec.c: Correctly take the granulepos from upstream if possible and correctly handle the granulepos ...
      
      Original commit message from CVS:
      * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
      Correctly take the granulepos from upstream if possible and
      correctly handle the granulepos in various calculations: the
      granulepos is the sample number of the _last_ sample in a frame, not
      the first.
      * ext/speex/gstspeexenc.c: (gst_speex_enc_sinkevent),
      (gst_speex_enc_encode), (gst_speex_enc_chain),
      (gst_speex_enc_change_state):
      * ext/speex/gstspeexenc.h:
      Handle non-zero start timestamps in the encoder and detect/handle
      stream discontinuities. Fixes bug #547075.
      25896b3a
    • Craig Keogh's avatar
      ext/annodex/gstcmmlparser.c: Fix compiler warnings caused by passing a string... · 467b9f28
      Craig Keogh authored
      ext/annodex/gstcmmlparser.c: Fix compiler warnings caused by passing a string as format string instead of "%s" and th...
      
      Original commit message from CVS:
      Patch by: Craig Keogh <cskeogh at adam dot com dot au>
      * ext/annodex/gstcmmlparser.c: (gst_cmml_parser_parse_chunk):
      Fix compiler warnings caused by passing a string as format string
      instead of "%s" and then the string. This is only exposed by -Wformat=2
      as used by default on Ubuntu. Fixes bug #550015.
      467b9f28
  17. 30 Aug, 2008 1 commit
    • Tim-Philipp Müller's avatar
      Make stuff compile with GST_DISABLE_GST_DEBUG. · 5c4b6ce0
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * ext/raw1394/gsthdv1394src.c: (gst_hdv1394src_create):
      * gst/alpha/gstalpha.c: (gst_alpha_get_unit_size):
      * gst/audiofx/audiocheblimit.c: (generate_coefficients):
      * gst/avi/gstavidemux.c: (gst_avi_demux_src_convert):
      * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
      (gst_ebml_read_element_length):
      * gst/matroska/matroska-demux.c:
      (gst_matroska_demux_check_subtitle_buffer):
      Make stuff compile with GST_DISABLE_GST_DEBUG.
      5c4b6ce0
  18. 29 Aug, 2008 1 commit
  19. 28 Aug, 2008 1 commit
    • Mersad Jelacic's avatar
      gst/multipart/: Convert audio/x-adpcm to and from the audio/G726-X in the... · 9b08b530
      Mersad Jelacic authored
      gst/multipart/: Convert audio/x-adpcm to and from the audio/G726-X in the muxer and demuxer. Fixes #549551.
      
      Original commit message from CVS:
      Patch by: Mersad Jelacic <mersad at axis dot com>
      * gst/multipart/multipartdemux.c:
      * gst/multipart/multipartmux.c: (gst_multipart_mux_get_mime):
      Convert audio/x-adpcm to and from the audio/G726-X in the muxer and
      demuxer. Fixes #549551.
      9b08b530
  20. 27 Aug, 2008 3 commits
  21. 26 Aug, 2008 2 commits