1. 12 Apr, 2007 3 commits
    • jerry tan's avatar
      sys/sunaudio/gstsunaudiosrc.c: it is the application's responsibility to make... · a7efc5ce
      jerry tan authored
      sys/sunaudio/gstsunaudiosrc.c: it is the application's responsibility to make sure it open the device once.
      
      Original commit message from CVS:
      Patch by: jerry tan <jerry dot tan at sun dot com>
      * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
      remove the call of  ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the
      application's responsibility to make sure it open the device once.
      Remove a careless error if AUDIODEV is set. Fixes #392620.
      a7efc5ce
    • Wim Taymans's avatar
      gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the... · eae68a64
      Wim Taymans authored
      gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the pts_offset calculations.
      
      Original commit message from CVS:
      * gst/qtdemux/qtdemux.c:
      Make timescale 32 bits again so we don't screw up the pts_offset
      calculations.
      eae68a64
    • Wim Taymans's avatar
      gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals. · 86a4c1c6
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
      (gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
      * gst/rtsp/gstrtpdec.h:
      Make backward compat with rtpbin by adding the request-pt-map signals.
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
      (new_session_pad), (request_pt_map),
      (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_stream_configure_caps),
      (gst_rtspsrc_activate_streams):
      * gst/rtsp/gstrtspsrc.h:
      Implement request-pt-map signals instead of setting caps on the buffers
      for the session manager.
      86a4c1c6
  2. 11 Apr, 2007 3 commits
  3. 10 Apr, 2007 3 commits
    • Wim Taymans's avatar
      gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups. · acddbd83
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
      (gst_rtp_amr_depay_process):
      Fix depayloader clock_rate and some cleanups.
      * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
      (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
      * gst/rtp/gstrtph264depay.h:
      Don't push codec_data in the adapter because it might get flushed when
      we get a discont.
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
      Handle multiple AU per packet.
      * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
      (gst_rtp_sv3v_depay_plugin_init):
      Disable rank, this one does not work.
      Remove timestamping, base class does that.
      acddbd83
    • Stefan Kost's avatar
      gst/auparse/gstauparse.c: limit caps to the formats we announce in the template · 497d589d
      Stefan Kost authored
      Original commit message from CVS:
      * gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
      limit caps to the formats we announce in the template
      * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
      (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
      (gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
      fix some crashers/asserts when dealing with broken files
      497d589d
    • Peter Kjellerstedt's avatar
      gst/: Fix some compiler warnings. Fixes #428182. · 50f88db3
      Peter Kjellerstedt authored
      Original commit message from CVS:
      Patch by: Peter Kjellerstedt  <pkj at axis com>
      * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
      * gst/rtp/gstrtpL16depay.c:
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
      * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
      (gst_rtp_speex_depay_setcaps):
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
      (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
      * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
      Fix some compiler warnings. Fixes #428182.
      50f88db3
  4. 06 Apr, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback. · f80444aa
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/Makefile.am:
      * gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
      (free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
      (gst_rtp_dec_init), (gst_rtp_dec_finalize),
      (gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
      (gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
      (gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
      (gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
      (create_rtcp), (gst_rtp_dec_request_new_pad),
      (gst_rtp_dec_release_pad):
      * gst/rtsp/gstrtpdec.h:
      * gst/rtsp/gstrtsp.c: (plugin_init):
      Morph RTPDec into something compatible with RTPBin as a fallback.
      Various other style fixes.
      * gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
      (find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
      (gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
      (new_session_pad), (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
      (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
      * gst/rtsp/gstrtspsrc.h:
      Implement RTPBin session manager handling.
      Don't try to add empty properties to caps.
      Implement fallback session manager, handling.
      Don't combine errors from RTCP streams, just ignore them.
      * gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
      * gst/rtsp/rtsptransport.h:
      Implement fallback session manager.
      Make RTPBin the default one when available.
      f80444aa
  5. 05 Apr, 2007 3 commits
  6. 04 Apr, 2007 1 commit
    • Stefan Kost's avatar
      gst/avi/: Don't abort on out-of-memory. Use stream-nr as unsigned integer only. · 30df72cc
      Stefan Kost authored
      Original commit message from CVS:
      * gst/avi/README:
      * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
      (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
      (gst_avi_demux_stream_index), (gst_avi_demux_sync),
      (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
      (gst_avi_demux_calculate_durations_from_index),
      (gst_avi_demux_stream_header_push),
      (gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
      (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
      Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
      30df72cc
  7. 03 Apr, 2007 1 commit
    • Wim Taymans's avatar
      gst/smpte/barboxwipes.c: · 9598d82c
      Wim Taymans authored
      Original commit message from CVS:
      * gst/smpte/barboxwipes.c:
      Fix error as spotted by Snaik <snaik32 at gmail dot com>
      9598d82c
  8. 30 Mar, 2007 3 commits
    • Sebastian Dröge's avatar
      gst/wavparse/gstwavparse.c: Support audio/x-raw-float in wav files. This only... · c11fefd4
      Sebastian Dröge authored
      gst/wavparse/gstwavparse.c: Support audio/x-raw-float in wav files. This only works with plugins-base CVS, using an o...
      
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c:
      Support audio/x-raw-float in wav files. This only works with
      plugins-base CVS, using an older version doesn't have any
      disadvantages though.
      c11fefd4
    • Sebastian Dröge's avatar
      Revert last change as we don't want plugins-good to depend on plugins-base CVS now. · 6632cdb0
      Sebastian Dröge authored
      Original commit message from CVS:
      * configure.ac:
      * gst/auparse/gstauparse.c: (gst_au_parse_reset),
      (gst_au_parse_parse_header), (gst_au_parse_chain):
      * gst/auparse/gstauparse.h:
      Revert last change as we don't want plugins-good to depend on
      plugins-base CVS now.
      6632cdb0
    • Sebastian Dröge's avatar
      ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples... · 6d8e6c9b
      Sebastian Dröge authored
      ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept th...
      
      Original commit message from CVS:
      * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
      (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps),
      (gst_wavpack_dec_clip_outgoing_buffer),
      (gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain):
      * ext/wavpack/gstwavpackdec.h:
      * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
      (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
      (gst_wavpack_enc_chain):
      * ext/wavpack/gstwavpackenc.h:
      * ext/wavpack/gstwavpackparse.c:
      Don't play audioconvert. As wavpack wants/outputs all samples with
      width==32 and depth=[1,32] accept this and let audioconvert convert
      to accepted formats instead of doing it in the element for n*8 depths.
      This also adds support for non-n*8 depths and prevents some useless
      memory allocations. Fixes #421598
      Also add a workaround for bug #421542 in wavpackenc for now...
      * tests/check/elements/wavpackdec.c: (GST_START_TEST):
      * tests/check/elements/wavpackenc.c: (GST_START_TEST):
      * tests/check/elements/wavpackparse.c: (GST_START_TEST):
      Consider the change above in the unit tests and test if the correct
      caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in
      the wavpackparse unit test.
      * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
      (gst_wavpack_dec_sink_set_caps):
      Set caps on the src pad as soon as possible.
      * ext/wavpack/gstwavpackdec.h:
      * ext/wavpack/gstwavpackcommon.h:
      * ext/wavpack/gstwavpackenc.h:
      * ext/wavpack/gstwavpackparse.h:
      Fix indention. gst-indent is now called by cicl.
      6d8e6c9b
  9. 29 Mar, 2007 6 commits
    • René Stadler's avatar
      configure.ac: Require gst-plugins-base CVS for audioconvert with non-native... · bfd65c42
      René Stadler authored
      configure.ac: Require gst-plugins-base CVS for audioconvert with non-native float support and width/depth fix in libg...
      
      Original commit message from CVS:
      * configure.ac:
      Require gst-plugins-base CVS for audioconvert with non-native
      float support and width/depth fix in libgstriff.
      Patch by: René Stadler <mail at renestadler dot de>
      * gst/auparse/gstauparse.c: (gst_au_parse_reset),
      (gst_au_parse_parse_header), (gst_au_parse_chain):
      * gst/auparse/gstauparse.h:
      Don't swap the floats ourself if they're not in native endianness.
      Instead let audioconvert handle this. Fixes #339838.
      bfd65c42
    • Wim Taymans's avatar
      gst/rtp/: Flush adapter on disconts. · a87260cb
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstasteriskh263.h:
      * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process),
      (gst_rtp_h263p_depay_change_state):
      * gst/rtp/gstrtph263pdepay.h:
      * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
      (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
      (gst_rtp_h264_depay_change_state):
      * gst/rtp/gstrtph264depay.h:
      * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
      (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process):
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
      Flush adapter on disconts.
      a87260cb
    • Wim Taymans's avatar
      gst/rtp/: Use more efficient adapter and rtpbuffer methods when possible. · da3e23d3
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process):
      * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process):
      * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process):
      * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
      * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush):
      * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
      (gst_rtp_mp4v_depay_process):
      * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush):
      * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process):
      * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush):
      * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
      * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
      * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
      * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process):
      Use more efficient adapter and rtpbuffer methods when possible.
      da3e23d3
    • Sebastian Dröge's avatar
      gst/wavenc/gstwavenc.c: Correctly handle width!=depth input. · d26cbc8c
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
      (gst_wavenc_sink_setcaps):
      Correctly handle width!=depth input.
      * gst/wavparse/gstwavparse.c:
      Already export in the caps that width==8 uses unsigned samples and
      everything else uses signed samples.
      d26cbc8c
    • Laurent Glayal's avatar
      gst/udp/: Rework the socket allocation a bit based on the sockfd argument so... · 112216c2
      Laurent Glayal authored
      gst/udp/: Rework the socket allocation a bit based on the sockfd argument so that it becomes usable.
      
      Original commit message from CVS:
      Patch by: Laurent Glayal <spglegle at yahoo dot fr>
      * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
      (gst_dynudpsink_init), (gst_dynudpsink_set_property),
      (gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
      (gst_dynudpsink_close):
      * gst/udp/gstdynudpsink.h:
      * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
      (gst_udpsrc_create), (gst_udpsrc_set_property),
      (gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
      * gst/udp/gstudpsrc.h:
      Rework the socket allocation a bit based on the sockfd argument so that
      it becomes usable.
      Add a closefd property to instruct the udp elements to close the custom
      file descriptors when going to READY. Fixes #423304.
      API:GstUDPSrc::closefd property
      API:GstDynUDPSink::closefd property
      112216c2
    • Laurent Glayal's avatar
      gst/rtp/: Added H264 payloader. Fixes #423782. · d94a696b
      Laurent Glayal authored
      Original commit message from CVS:
      Patch by: Laurent Glayal <spglegle at yahoo dot fr>
      * gst/rtp/Makefile.am:
      * gst/rtp/gstrtp.c: (plugin_init):
      * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
      (gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
      (gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
      (gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
      (gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
      (gst_rtp_h264_pay_plugin_init):
      * gst/rtp/gstrtph264pay.h:
      Added H264 payloader. Fixes #423782.
      * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
      (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
      Small fixes.
      d94a696b
  10. 28 Mar, 2007 4 commits
    • Sebastian Dröge's avatar
      gst/wavparse/gstwavparse.c: Actually support depths from 1 to 32, not only 8 to 32. · c76eea67
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c:
      Actually support depths from 1 to 32, not only 8 to 32.
      c76eea67
    • Sebastian Dröge's avatar
      gst/wavparse/gstwavparse.c: Add support for wav files containing... · 7add372a
      Sebastian Dröge authored
      gst/wavparse/gstwavparse.c: Add support for wav files containing audio/x-raw-int with random depths between 1 and 32 ...
      
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c:
      Add support for wav files containing audio/x-raw-int with random
      depths between 1 and 32 bits.
      7add372a
    • Stefan Kost's avatar
      gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792. · c0cdcae5
      Stefan Kost authored
      Original commit message from CVS:
      Based on patch by: Stefan Kost  <ensonic@users.sf.net>
      * gst/rtp/Makefile.am:
      * gst/rtp/gstrtp.c: (plugin_init):
      * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
      (gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
      (gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
      (gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
      (gst_rtp_mp4a_depay_get_property),
      (gst_rtp_mp4a_depay_change_state),
      (gst_rtp_mp4a_depay_plugin_init):
      * gst/rtp/gstrtpmp4adepay.h:
      Added MP4A-LATM depayloader. Fixes #417792.
      * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
      (gst_rtp_mp4v_depay_process):
      Fixup depayloader, setting codec_data, using more efficient adaptor and
      rtpbuffer handling.
      * gst/rtsp/URLS:
      Add url to test above.
      c0cdcae5
    • Edward Hervey's avatar
      gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov... · ab589bff
      Edward Hervey authored
      gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov files with h264 video).
      
      Original commit message from CVS:
      * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
      (gst_qtdemux_chain), (qtdemux_parse_samples):
      * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_ctts):
      * gst/qtdemux/qtdemux_dump.h:
      * gst/qtdemux/qtdemux_fourcc.h:
      * gst/qtdemux/qtdemux_types.c:
      Process 'ctts' atoms, which are present in AVC ISO files (.mov files
      with h264 video).
      Use the offset present in 'ctts' to calculate the PTS for each packet
      and set the PTS on outgoing buffers.
      Fixes #423283
      ab589bff
  11. 25 Mar, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types,... · 8f5fb88b
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field ...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
      (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
      (get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
      (gst_rtspsrc_media_to_caps),
      (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_stream_configure_caps),
      (gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
      * gst/rtsp/gstrtspsrc.h:
      Handle default clock-rates for static payload types, rearrange stuff so
      that the rtpmap field in the sdp can override the defaults.
      Parse RTP-Info field to get the seqnum and timebase fields that should
      go in the caps.
      Delay configuring caps after we got the RTP-Info from the PLAY reply from
      the server.
      8f5fb88b
  12. 24 Mar, 2007 1 commit
    • Tim-Philipp Müller's avatar
      gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging... · c53ad300
      Tim-Philipp Müller authored
      gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging input caps into 1-channel output caps (I...
      
      Original commit message from CVS:
      * gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps):
      Remove 'channel-positions' field when munging input caps into
      1-channel output caps (I guess technically we should set the
      position for each channel on the output caps if it's non-NONE,
      but I'll save that as a task for another day).
      c53ad300
  13. 22 Mar, 2007 6 commits
    • Tim-Philipp Müller's avatar
      gst/interleave/deinterleave.c: Don't leak input buffer in chain function;... · 56b1a888
      Tim-Philipp Müller authored
      gst/interleave/deinterleave.c: Don't leak input buffer in chain function; maintain our own list of source pads - ther...
      
      Original commit message from CVS:
      * gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
      (gst_deinterleave_remove_pads), (gst_deinterleave_process),
      (gst_deinterleave_chain):
      Don't leak input buffer in chain function; maintain our own list of
      source pads - there are no guarantees about the order of the list
      in the GstElement struct, and we want a very specific order; lastly,
      some more debugging.
      56b1a888
    • Sebastian Dröge's avatar
      ext/wavpack/gstwavpackparse.c: Revert last commit, preventing infinite... · 7edf0661
      Sebastian Dröge authored
      ext/wavpack/gstwavpackparse.c: Revert last commit, preventing infinite plugging loops with ranks is no clean solution...
      
      Original commit message from CVS:
      * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
      Revert last commit, preventing infinite plugging loops with ranks
      is no clean solution and in general there's no reason why one wants
      to parse framed wavpack data again.
      7edf0661
    • Sebastian Dröge's avatar
      ext/wavpack/gstwavpackenc.c: Send the new segment event in time format instead... · 20dd20b2
      Sebastian Dröge authored
      ext/wavpack/gstwavpackenc.c: Send the new segment event in time format instead of bytes. This allows "wavpackenc ! wa...
      
      Original commit message from CVS:
      * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
      Send the new segment event in time format instead of bytes. This
      allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines.
      * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
      Accept framed and non-framed input, wavpackparse doesn't care. To
      prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the
      rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse !
      ..." pipelines.
      20dd20b2
    • Sebastian Dröge's avatar
      ext/wavpack/gstwavpackdec.c: Revert to use gst_pad_alloc_buffer() here. We can and should use it. · bc6a9a97
      Sebastian Dröge authored
      Original commit message from CVS:
      * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
      Revert to use gst_pad_alloc_buffer() here. We can and should use it.
      Thanks to Jan and Mike for noticing my mistake.
      bc6a9a97
    • Christophe Dehais's avatar
      ext/gconf/gconf.c: Accept complex pipeline descriptions as an audio profile... · c410265b
      Christophe Dehais authored
      ext/gconf/gconf.c: Accept complex pipeline descriptions as an audio profile instead of just a single element. Fixes #...
      
      Original commit message from CVS:
      Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>
      * ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
      Accept complex pipeline descriptions as an audio profile instead of just
      a single element. Fixes #420658.
      c410265b
    • Sebastian Dröge's avatar
      ext/wavpack/gstwavpackenc.*: Put the write helpers into the GstWavpackEnc... · a1a03796
      Sebastian Dröge authored
      ext/wavpack/gstwavpackenc.*: Put the write helpers into the GstWavpackEnc struct directly and not as a pointer to sav...
      
      Original commit message from CVS:
      * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
      (gst_wavpack_enc_init), (gst_wavpack_enc_chain),
      (gst_wavpack_enc_rewrite_first_block):
      * ext/wavpack/gstwavpackenc.h:
      Put the write helpers into the GstWavpackEnc struct directly and not
      as a pointer to save two small, but useless mallocs. This also makes
      it possible to drop the finalize method.
      * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_push_buffer):
      For consistency reasons also set GST_BUFFER_OFFSET_END on the outgoing
      buffers the same way wavpackenc does it.
      a1a03796
  14. 21 Mar, 2007 2 commits
  15. 19 Mar, 2007 1 commit
  16. 18 Mar, 2007 1 commit