1. 27 Apr, 2007 1 commit
  2. 26 Apr, 2007 5 commits
    • Edward Hervey's avatar
      docs/plugins/: Add documentation for osxaudio plugin. · a9a843b3
      Edward Hervey authored
      Original commit message from CVS:
      * docs/plugins/gst-plugins-good-plugins-docs.sgml:
      * docs/plugins/gst-plugins-good-plugins-sections.txt:
      * docs/plugins/gst-plugins-good-plugins.hierarchy:
      * docs/plugins/inspect/plugin-osxaudio.xml:
      Add documentation for osxaudio plugin.
      a9a843b3
    • Edward Hervey's avatar
      docs/plugins/: Add documentation for osxvideo · 4566295e
      Edward Hervey authored
      Original commit message from CVS:
      * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
      * docs/plugins/gst-plugins-bad-plugins-sections.txt:
      * docs/plugins/gst-plugins-bad-plugins.hierarchy:
      * docs/plugins/inspect/plugin-osxvideo.xml:
      Add documentation for osxvideo
      4566295e
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Protect state changes with a lock. · 530f214b
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
      (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_open), (gst_rtspsrc_close),
      (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
      (gst_rtspsrc_pause):
      * gst/rtsp/gstrtspsrc.h:
      Protect state changes with a lock.
      * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
      (parse_line):
      * gst/rtsp/rtspconnection.h:
      Remove some unused stuff.
      530f214b
    • Wim Taymans's avatar
      gst/udp/gstudpsrc.c: Handle the case where there are exactly 0 bytes to read... · 45b77c57
      Wim Taymans authored
      gst/udp/gstudpsrc.c: Handle the case where there are exactly 0 bytes to read and the ioctl did not report an error. F...
      
      Original commit message from CVS:
      * gst/udp/gstudpsrc.c: (gst_udpsrc_create):
      Handle the case where there are exactly 0 bytes to read and the ioctl
      did not report an error. Fixes #433530.
      45b77c57
    • Wim Taymans's avatar
      gst/wavparse/gstwavparse.*: Apply DISCONT to buffers. · 88bf47c9
      Wim Taymans authored
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
      (gst_wavparse_stream_headers), (gst_wavparse_stream_data):
      * gst/wavparse/gstwavparse.h:
      Apply DISCONT to buffers.
      Only apply timestamp to the first sample after a DISCONT, too many VBR
      files cause random jitter in the timestamps. Fixes #433119.
      88bf47c9
  3. 25 Apr, 2007 7 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtpdec.*: Add dummy latency property to be backwards compat with rtpbin. · 6937be1a
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
      (gst_rtp_dec_init), (gst_rtp_dec_set_property),
      (gst_rtp_dec_get_property):
      * gst/rtsp/gstrtpdec.h:
      Add dummy latency property to be backwards compat with rtpbin.
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
      (gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
      (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_parse_rtpinfo):
      * gst/rtsp/gstrtspsrc.h:
      Add latency property and configure in the session manager.
      Don't set invalid clock-base and seqnum-base on caps, some servers
      sometimes don't send them.
      6937be1a
    • Tim-Philipp Müller's avatar
      gst/alpha/gstalphacolor.c: Double-check that RGB input caps are really RGBA... · e53a2451
      Tim-Philipp Müller authored
      gst/alpha/gstalphacolor.c: Double-check that RGB input caps are really RGBA caps (apparently the core doesn't always ...
      
      Original commit message from CVS:
      * gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
      (gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
      Double-check that RGB input caps are really RGBA caps (apparently
      the core doesn't always catch it if those caps aren't a subset of
      our template caps, also see #421543). Fixes #429319 in a way.
      Also, don't leak the pad template in the transform_caps function.
      * tests/check/Makefile.am:
      * tests/check/elements/.cvsignore:
      * tests/check/elements/alphacolor.c: (setup_alphacolor),
      (cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
      (create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
      (GST_START_TEST), (alphacolor_suite):
      Add some basic unit tests for alphacolor.
      e53a2451
    • Tim-Philipp Müller's avatar
      ext/libpng/gstpngdec.c: If we get a fatal flow return in the loop function,... · 3f55b6e9
      Tim-Philipp Müller authored
      ext/libpng/gstpngdec.c: If we get a fatal flow return in the loop function, first post the error message and only the...
      
      Original commit message from CVS:
      * ext/libpng/gstpngdec.c: (gst_pngdec_task):
      If we get a fatal flow return in the loop function, first post the
      error message and only then send the EOS event downstream, otherwise
      applications might get an eos message before the error message and
      think everything was ok (related to #429319).
      3f55b6e9
    • Wim Taymans's avatar
      gst/rtsp/rtspconnection.c: Read the channel byte as an unsigned byte. · a7531984
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
      Read the channel byte as an unsigned byte.
      a7531984
    • Wim Taymans's avatar
      gst/rtp/: Make sure we configure the clock_rate in the baseclass in the... · 24c5812d
      Wim Taymans authored
      gst/rtp/: Make sure we configure the clock_rate in the baseclass in the setcaps function. Fixes #431282.
      
      Original commit message from CVS:
      * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
      * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
      (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
      * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
      (gst_rtp_gsm_depay_setcaps):
      * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
      * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
      * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
      (gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
      (gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
      (gst_ilbc_depay_get_property):
      * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
      * gst/rtp/gstrtpmp4adepay.c:
      * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
      (gst_rtp_pcma_depay_setcaps):
      * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
      (gst_rtp_pcmu_depay_setcaps):
      Make sure we configure the clock_rate in the baseclass in the setcaps
      function. Fixes #431282.
      24c5812d
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Parse server address from SDP. · 1beeda3f
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
      (gst_rtspsrc_stream_free), (request_pt_map),
      (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
      * gst/rtsp/gstrtspsrc.h:
      Parse server address from SDP.
      Hook up a udpsink to send RTCP back to the server.
      * docs/plugins/gst-plugins-good-plugins-sections.txt:
      * gst/rtsp/rtsptransport.h:
      Add some docs.
      1beeda3f
    • Stefan Kost's avatar
      gst/wavparse/gstwavparse.c: Make header field check conditional. Fixes #433135 · fa7454bd
      Stefan Kost authored
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
      Make header field check conditional. Fixes #433135
      fa7454bd
  4. 24 Apr, 2007 1 commit
  5. 20 Apr, 2007 1 commit
  6. 18 Apr, 2007 2 commits
  7. 17 Apr, 2007 3 commits
  8. 16 Apr, 2007 1 commit
    • Brian Cameron's avatar
      sys/sunaudio/: Fix and/or update copyright attributions (#430228). · f520911b
      Brian Cameron authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      Original commit message from CVS:
      Patch by: Brian Cameron  <brian.cameron at sun dot com>
      * sys/sunaudio/gstsunaudio.c:
      * sys/sunaudio/gstsunaudiomixer.c:
      * sys/sunaudio/gstsunaudiomixer.h:
      * sys/sunaudio/gstsunaudiomixerctrl.c:
      * sys/sunaudio/gstsunaudiomixerctrl.h:
      * sys/sunaudio/gstsunaudiomixertrack.h:
      * sys/sunaudio/gstsunaudiosink.c:
      * sys/sunaudio/gstsunaudiosink.h:
      * sys/sunaudio/gstsunaudiosrc.c:
      * sys/sunaudio/gstsunaudiosrc.h:
      Fix and/or update copyright attributions (#430228).
      f520911b
  9. 14 Apr, 2007 1 commit
    • Sebastien Moutte's avatar
      docs/plugins/inspect/: Add xml doc files for Windows sinks · 5100794b
      Sebastien Moutte authored
      Original commit message from CVS:
      * docs/plugins/inspect/plugin-directdraw.xml:
      * docs/plugins/inspect/plugin-directsound.xml:
      * docs/plugins/inspect/plugin-waveform.xml:
      Add xml doc files for Windows sinks
      * win32/vs6/libgstqtdemux.dsp:
      * win32/vs6/libgstmpegvideoparse.dsp:
      * win32/vs6/gst_plugins_bad.dsw:
      Update projects files.
      5100794b
  10. 13 Apr, 2007 3 commits
    • Wim Taymans's avatar
      docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs. · b7524708
      Wim Taymans authored
      Original commit message from CVS:
      * docs/plugins/gst-plugins-good-plugins-sections.txt:
      Fix docs.
      * gst/rtsp/URLS:
      Add some more example urls.
      * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
      (gst_rtp_dec_chain_rtp):
      Better debugging.
      * gst/rtsp/gstrtspsrc.c: (request_pt_map),
      (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_parse_rtpinfo):
      Remove unused code.
      b7524708
    • Stefan Kost's avatar
      gst/wavparse/gstwavparse.c: Relax the audio/mpeg caps again and add FIXME: comment. · 3bf1b5ec
      Stefan Kost authored
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
      (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
      (gst_wavparse_stream_data):
      Relax the audio/mpeg caps again and add FIXME: comment.
      3bf1b5ec
    • Stefan Kost's avatar
      gst/wavparse/gstwavparse.*: More sanity check for the header fields. Fix type... · 0722106b
      Stefan Kost authored
      gst/wavparse/gstwavparse.*: More sanity check for the header fields. Fix type for 'rate' header field.
      
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
      (gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
      (gst_wavparse_stream_data):
      * gst/wavparse/gstwavparse.h:
      More sanity check for the header fields. Fix type for 'rate' header
      field.
      0722106b
  11. 12 Apr, 2007 6 commits
    • Tim-Philipp Müller's avatar
      gst/icydemux/gsticydemux.c: If the metadata strings we get in the stream are... · ef7c1881
      Tim-Philipp Müller authored
      gst/icydemux/gsticydemux.c: If the metadata strings we get in the stream are not UTF-8, try to interpret them accordi...
      
      Original commit message from CVS:
      * gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8),
      (gst_icydemux_unicodify):
      If the metadata strings we get in the stream are not UTF-8, try to
      interpret them according to the character encodings specified in the
      GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and
      only fall back to locale/ISO-8859-1 if those aren't set or don't
      work. Should fix #428901.
      ef7c1881
    • Wim Taymans's avatar
      gst/rtp/gstrtph264depay.c: Use the proper sync word for SPS and PPS. · f5e4a8b0
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtph264depay.c:
      Use the proper sync word for SPS and PPS.
      f5e4a8b0
    • Thomas Vander Stichele's avatar
      gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT,... · 2fc86884
      Thomas Vander Stichele authored
      gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME, fnv1_hash_32_new, fnv1_hash_...
      
      Original commit message from CVS:
      * gst/rtp/Makefile.am:
      * gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
      fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
      * gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
      Add a simple hashing implementation that we can use to generate
      a 24-bit ident value based on the codebooks for vorbis and theora.
      * gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
      gst_rtp_theora_pay_handle_buffer):
      * gst/rtp/gstrtpvorbisdepay.c
      (gst_rtp_vorbis_depay_parse_configuration,
      gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
      * gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
      gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
      gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
      Use the hashing function, ensuring that the same codebooks result
      in the same ident and thus the same SDP description.
      Various log fixes/changes.
      2fc86884
    • jerry tan's avatar
      sys/sunaudio/gstsunaudiosrc.c: it is the application's responsibility to make... · a7efc5ce
      jerry tan authored and Wim Taymans's avatar Wim Taymans committed
      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
  12. 11 Apr, 2007 3 commits
  13. 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 and Wim Taymans's avatar Wim Taymans committed
      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
  14. 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
  15. 05 Apr, 2007 2 commits