mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
Rename files and types from speexresample to audioresample
Rename files and types from speexresample to audioresample to finish the move and to prevent any confusion.
This commit is contained in:
Sebastian Dröge
parent
e327bbcd7d
commit
5dfcb63252
73 changed files with 2774 additions and 6090 deletions
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@ -305,7 +305,7 @@ AG_GST_CHECK_PLUGIN(audiotestsrc)
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AG_GST_CHECK_PLUGIN(ffmpegcolorspace)
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AG_GST_CHECK_PLUGIN(gdp)
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AG_GST_CHECK_PLUGIN(playback)
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AG_GST_CHECK_PLUGIN(speexresample)
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AG_GST_CHECK_PLUGIN(audioresample)
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AG_GST_CHECK_PLUGIN(subparse)
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AG_GST_CHECK_PLUGIN(tcp)
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AG_GST_CHECK_PLUGIN(typefind)
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@ -690,7 +690,7 @@ gst/audiotestsrc/Makefile
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gst/ffmpegcolorspace/Makefile
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gst/gdp/Makefile
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gst/playback/Makefile
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gst/speexresample/Makefile
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gst/audioresample/Makefile
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gst/subparse/Makefile
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gst/tcp/Makefile
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gst/typefind/Makefile
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@ -106,7 +106,7 @@ EXTRA_HFILES = \
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$(top_srcdir)/gst/gdp/gstgdpdepay.h \
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$(top_srcdir)/gst/gdp/gstgdppay.h \
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$(top_srcdir)/gst/playback/gstplay-enum.h \
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$(top_srcdir)/gst/speexresample/gstspeexresample.h \
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$(top_srcdir)/gst/audioresample/gstaudioresample.h \
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$(top_srcdir)/gst/tcp/gstmultifdsink.h \
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$(top_srcdir)/gst/tcp/gsttcpclientsrc.h \
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$(top_srcdir)/gst/tcp/gsttcpclientsink.h \
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@ -107,15 +107,15 @@ audio_convert_prepare_context
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<SECTION>
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<FILE>element-audioresample</FILE>
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<TITLE>audioresample</TITLE>
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GstSpeexResample
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GstAudioResample
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<SUBSECTION Standard>
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GST_SPEEX_RESAMPLE
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GST_IS_SPEEX_RESAMPLE
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GST_TYPE_SPEEX_RESAMPLE
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gst_speex_resample_get_type
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GST_SPEEX_RESAMPLE_CLASS
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GST_IS_SPEEX_RESAMPLE_CLASS
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GstSpeexResampleClass
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GST_AUDIO_RESAMPLE
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GST_IS_AUDIO_RESAMPLE
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GST_TYPE_AUDIO_RESAMPLE
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gst_audio_resample_get_type
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GST_AUDIO_RESAMPLE_CLASS
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GST_IS_AUDIO_RESAMPLE_CLASS
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GstAudioResampleClass
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</SECTION>
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<SECTION>
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@ -641,7 +641,7 @@
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<ARG>
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<NAME>GstMultiFdSink::buffers-max</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Buffers max</NICK>
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<BLURB>max number of buffers to queue for a client (-1 = no limit).</BLURB>
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@ -661,7 +661,7 @@
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<ARG>
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<NAME>GstMultiFdSink::buffers-soft-max</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Buffers soft max</NICK>
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<BLURB>Recover client when going over this limit (-1 = no limit).</BLURB>
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@ -751,7 +751,7 @@
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<ARG>
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<NAME>GstMultiFdSink::buffers-min</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Buffers min</NICK>
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<BLURB>min number of buffers to queue (-1 = as few as possible).</BLURB>
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@ -781,7 +781,7 @@
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<ARG>
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<NAME>GstMultiFdSink::bytes-min</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Bytes min</NICK>
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<BLURB>min number of bytes to queue (-1 = as little as possible).</BLURB>
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@ -791,7 +791,7 @@
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<ARG>
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<NAME>GstMultiFdSink::time-min</NAME>
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<TYPE>gint64</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Time min</NICK>
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<BLURB>min number of time to queue (-1 = as little as possible).</BLURB>
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@ -811,7 +811,7 @@
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<ARG>
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<NAME>GstMultiFdSink::units-max</NAME>
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<TYPE>gint64</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Units max</NICK>
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<BLURB>max number of units to queue (-1 = no limit).</BLURB>
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@ -821,7 +821,7 @@
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<ARG>
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<NAME>GstMultiFdSink::units-soft-max</NAME>
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<TYPE>gint64</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Units soft max</NICK>
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<BLURB>Recover client when going over this limit (-1 = no limit).</BLURB>
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@ -831,7 +831,7 @@
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<ARG>
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<NAME>GstMultiFdSink::qos-dscp</NAME>
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<TYPE>gint</TYPE>
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<RANGE>[-1,63]</RANGE>
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<RANGE>[G_MAXULONG,63]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>QoS diff srv code point</NICK>
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<BLURB>Quality of Service, differentiated services code point (-1 default).</BLURB>
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@ -981,7 +981,7 @@
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<ARG>
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<NAME>GstVorbisEnc::bitrate</NAME>
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<TYPE>gint</TYPE>
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<RANGE>[-1,250001]</RANGE>
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<RANGE>[G_MAXULONG,250001]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Target Bitrate</NICK>
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<BLURB>Attempt to encode at a bitrate averaging this (in bps). This uses the bitrate management engine, and is not recommended for most users. Quality is a better alternative. (-1 == disabled).</BLURB>
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@ -1011,7 +1011,7 @@
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<ARG>
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<NAME>GstVorbisEnc::max-bitrate</NAME>
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<TYPE>gint</TYPE>
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<RANGE>[-1,250001]</RANGE>
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<RANGE>[G_MAXULONG,250001]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Maximum Bitrate</NICK>
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<BLURB>Specify a maximum bitrate (in bps). Useful for streaming applications. (-1 == disabled).</BLURB>
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@ -1021,7 +1021,7 @@
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<ARG>
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<NAME>GstVorbisEnc::min-bitrate</NAME>
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<TYPE>gint</TYPE>
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<RANGE>[-1,250001]</RANGE>
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<RANGE>[G_MAXULONG,250001]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Minimum Bitrate</NICK>
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<BLURB>Specify a minimum bitrate (in bps). Useful for encoding for a fixed-size channel. (-1 == disabled).</BLURB>
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@ -1861,7 +1861,7 @@
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<ARG>
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<NAME>GstCdParanoiaSrc::read-speed</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Read speed</NICK>
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<BLURB>Read from device at specified speed.</BLURB>
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@ -1871,7 +1871,7 @@
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<ARG>
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<NAME>GstCdParanoiaSrc::search-overlap</NAME>
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<TYPE>gint</TYPE>
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<RANGE>[-1,75]</RANGE>
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<RANGE>[G_MAXULONG,75]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Search overlap</NICK>
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<BLURB>Force minimum overlap search during verification to n sectors.</BLURB>
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@ -2011,7 +2011,7 @@
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<ARG>
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<NAME>GstURIDecodeBin::buffer-duration</NAME>
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<TYPE>gint64</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Buffer duration (ns)</NICK>
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<BLURB>Buffer duration when buffering network streams.</BLURB>
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@ -2021,7 +2021,7 @@
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<ARG>
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<NAME>GstURIDecodeBin::buffer-size</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Buffer size (bytes)</NICK>
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<BLURB>Buffer size when buffering network streams.</BLURB>
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@ -2181,7 +2181,7 @@
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<ARG>
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<NAME>GstPlayBin2::current-audio</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Current audio</NICK>
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<BLURB>Currently playing audio stream (-1 = auto).</BLURB>
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@ -2191,7 +2191,7 @@
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<ARG>
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<NAME>GstPlayBin2::current-text</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Current Text</NICK>
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<BLURB>Currently playing text stream (-1 = auto).</BLURB>
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@ -2201,7 +2201,7 @@
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<ARG>
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<NAME>GstPlayBin2::current-video</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Current Video</NICK>
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<BLURB>Currently playing video stream (-1 = auto).</BLURB>
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@ -2361,7 +2361,7 @@
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<ARG>
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<NAME>GstPlayBin2::buffer-duration</NAME>
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<TYPE>gint64</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Buffer duration (ns)</NICK>
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<BLURB>Buffer duration when buffering network streams.</BLURB>
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@ -2371,7 +2371,7 @@
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<ARG>
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<NAME>GstPlayBin2::buffer-size</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Buffer size (bytes)</NICK>
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<BLURB>Buffer size when buffering network streams.</BLURB>
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@ -2711,7 +2711,7 @@
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<ARG>
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<NAME>GstAppSrc::max-latency</NAME>
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<TYPE>gint64</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Max Latency</NICK>
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<BLURB>The maximum latency (-1 = unlimited).</BLURB>
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@ -2721,7 +2721,7 @@
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<ARG>
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<NAME>GstAppSrc::min-latency</NAME>
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<TYPE>gint64</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Min Latency</NICK>
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<BLURB>The minimum latency (-1 = default).</BLURB>
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@ -2731,7 +2731,7 @@
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<ARG>
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<NAME>GstAppSrc::size</NAME>
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<TYPE>gint64</TYPE>
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<RANGE>>= -1</RANGE>
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<RANGE>>= G_MAXULONG</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Size</NICK>
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<BLURB>The size of the data stream in bytes (-1 if unknown).</BLURB>
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@ -2748,3 +2748,23 @@
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<DEFAULT>Stream</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstAudioResample::filter-length</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= 0</RANGE>
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<FLAGS>rwx</FLAGS>
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<NICK>Filter length</NICK>
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<BLURB>DEPRECATED, DON'T USE THIS! Length of the resample filter.</BLURB>
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<DEFAULT>64</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstAudioResample::quality</NAME>
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<TYPE>gint</TYPE>
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<RANGE>[0,10]</RANGE>
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<FLAGS>rwx</FLAGS>
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<NICK>Quality</NICK>
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<BLURB>Resample quality with 0 being the lowest and 10 being the best.</BLURB>
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<DEFAULT>4</DEFAULT>
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</ARG>
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@ -12,28 +12,9 @@ GObject
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GstPlayBaseBin
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GstPlayBin
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GstPlayBin2
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GstDecodeBin
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GstDecodeBin2
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GstURIDecodeBin
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GstDecodeBin
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GstBaseSrc
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GstPushSrc
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GstCddaBaseSrc
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GstCdParanoiaSrc
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GstBaseAudioSrc
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GstAudioSrc
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GstAlsaSrc
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GstV4lElement
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GstV4lSrc
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GstTCPClientSrc
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GstTCPServerSrc
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GstVideoTestSrc
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GstGnomeVFSSrc
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GstAppSrc
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GstAudioTestSrc
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GstVorbisEnc
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GstVorbisDec
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GstVorbisParse
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GstVorbisTag
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GstOggDemux
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GstOggMux
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GstOgmParse
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@ -43,49 +24,74 @@ GObject
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GstOggParse
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GstOggAviParse
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GstBaseSink
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GstGnomeVFSSink
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GstGioBaseSink
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GstGioSink
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GstGioStreamSink
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GstBaseAudioSink
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GstAudioSink
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GstAlsaSink
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GstGnomeVFSSink
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GstVideoSink
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GstXvImageSink
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GstXImageSink
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GstAppSink
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GstTCPClientSink
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GstMultiFdSink
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GstTCPServerSink
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GstAppSink
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GstVisual
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GstVisualbumpscope
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GstVisualcorona
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GstVisualinfinite
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GstVisualjakdaw
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GstVisualjess
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GstVisuallv_analyzer
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GstVisuallv_scope
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GstVisualoinksie
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GstTheoraDec
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GstTheoraEnc
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GstTheoraParse
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GstBaseSrc
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GstGioBaseSrc
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GstGioSrc
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GstGioStreamSrc
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GstPushSrc
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GstBaseAudioSrc
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GstAudioSrc
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GstAlsaSrc
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GstCddaBaseSrc
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GstCdParanoiaSrc
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GstV4lElement
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GstV4lSrc
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GstTCPClientSrc
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GstTCPServerSrc
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GstVideoTestSrc
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GstGnomeVFSSrc
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GstAudioTestSrc
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GstAppSrc
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GstVorbisEnc
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GstVorbisDec
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GstVorbisParse
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GstVorbisTag
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GstTextOverlay
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GstTimeOverlay
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GstClockOverlay
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GstTextRender
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GstTheoraDec
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GstTheoraEnc
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GstTheoraParse
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GstAlsaMixerElement
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GstGDPDepay
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GstGDPPay
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GstBaseTransform
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GstFFMpegCsp
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GstVideoScale
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GstAudioFilter
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GstVolume
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GstSpeexResample
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GstAudioConvert
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GstVisual
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GstVisualjess
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GstVisualbumpscope
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GstVisualcorona
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GstVisualinfinite
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GstVisualjakdaw
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GstVisuallv_analyzer
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GstVisuallv_scope
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GstVisualoinksie
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GstSubParse
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GstSsaParse
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GstAudioRate
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GstBaseTransform
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GstAudioConvert
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GstFFMpegCsp
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GstAudioFilter
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GstVolume
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GstAudioResample
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GstVideoScale
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GstAdder
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GstGDPDepay
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GstGDPPay
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GstStreamSelector
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GstQueue2
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GstAudioRate
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GstVideoRate
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GstBus
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GstTask
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@ -96,26 +102,62 @@ GObject
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GstRegistry
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GstRingBuffer
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GstSignalObject
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GConfClient
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GFileMonitor
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GLocalDirectoryMonitor
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GFamDirectoryMonitor
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GInotifyDirectoryMonitor
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GLocalFileMonitor
|
||||
GFamFileMonitor
|
||||
GInotifyFileMonitor
|
||||
GVolumeMonitor
|
||||
GNativeVolumeMonitor
|
||||
GProxyVolumeMonitor
|
||||
GProxyVolumeMonitorHal
|
||||
GProxyVolumeMonitorGPhoto2
|
||||
GUnixVolumeMonitor
|
||||
GDaemonVolumeMonitor
|
||||
GVfs
|
||||
GDaemonVfs
|
||||
GLocalVfs
|
||||
GTypeModule
|
||||
GIOModule
|
||||
GVfsUriMapper
|
||||
GVfsUriMapperSmb
|
||||
GVfsUriMapperHttp
|
||||
GAppLookupGConf
|
||||
GProxyDrive
|
||||
GProxyMount
|
||||
GProxyVolume
|
||||
GOutputStream
|
||||
GInputStream
|
||||
PangoFontMap
|
||||
PangoFcFontMap
|
||||
PangoFT2FontMap
|
||||
PangoContext
|
||||
GstMixerTrack
|
||||
GstMixerOptions
|
||||
LinkConnection
|
||||
GIOPConnection
|
||||
LinkServer
|
||||
GIOPServer
|
||||
GConfClient
|
||||
GstColorBalanceChannel
|
||||
GstTunerNorm
|
||||
GstTunerChannel
|
||||
GstColorBalanceChannel
|
||||
GstMixerTrack
|
||||
GstStreamInfo
|
||||
GInterface
|
||||
GTypePlugin
|
||||
GstChildProxy
|
||||
GstURIHandler
|
||||
GFile
|
||||
GDesktopAppInfoLookup
|
||||
GDrive
|
||||
GMount
|
||||
GVolume
|
||||
GstTagSetter
|
||||
GstImplementsInterface
|
||||
GstMixer
|
||||
GstPropertyProbe
|
||||
GstTuner
|
||||
GstNavigation
|
||||
GstXOverlay
|
||||
GstColorBalance
|
||||
GstNavigation
|
||||
GstTuner
|
||||
|
|
|
|||
|
|
@ -3,21 +3,29 @@ GstPipeline GstChildProxy
|
|||
GstPlayBaseBin GstChildProxy
|
||||
GstPlayBin GstChildProxy
|
||||
GstPlayBin2 GstChildProxy
|
||||
GstDecodeBin GstChildProxy
|
||||
GstDecodeBin2 GstChildProxy
|
||||
GstURIDecodeBin GstChildProxy
|
||||
GstDecodeBin GstChildProxy
|
||||
GstGioSink GstURIHandler
|
||||
GstAlsaSink GstPropertyProbe
|
||||
GstGnomeVFSSink GstURIHandler
|
||||
GstXvImageSink GstImplementsInterface GstPropertyProbe GstNavigation GstXOverlay GstColorBalance
|
||||
GstXImageSink GstImplementsInterface GstNavigation GstXOverlay
|
||||
GstGioSrc GstURIHandler
|
||||
GstAlsaSrc GstImplementsInterface GstMixer GstPropertyProbe
|
||||
GstCddaBaseSrc GstURIHandler
|
||||
GstCdParanoiaSrc GstURIHandler
|
||||
GstAlsaSrc GstImplementsInterface GstMixer GstPropertyProbe
|
||||
GstV4lElement GstImplementsInterface GstPropertyProbe GstTuner GstXOverlay GstColorBalance
|
||||
GstV4lSrc GstImplementsInterface GstPropertyProbe GstTuner GstXOverlay GstColorBalance
|
||||
GstV4lElement GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstTuner
|
||||
GstV4lSrc GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstTuner
|
||||
GstGnomeVFSSrc GstURIHandler
|
||||
GstAppSrc GstURIHandler
|
||||
GstVorbisEnc GstTagSetter
|
||||
GstVorbisTag GstTagSetter
|
||||
GstGnomeVFSSink GstURIHandler
|
||||
GstAlsaSink GstPropertyProbe
|
||||
GstXvImageSink GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstNavigation
|
||||
GstXImageSink GstImplementsInterface GstXOverlay GstNavigation
|
||||
GstAlsaMixerElement GstImplementsInterface GstMixer GstPropertyProbe
|
||||
GstVolume GstImplementsInterface GstMixer
|
||||
GTypeModule GTypePlugin
|
||||
GIOModule GTypePlugin
|
||||
GAppLookupGConf GDesktopAppInfoLookup
|
||||
GProxyDrive GDrive
|
||||
GProxyMount GMount
|
||||
GProxyVolume GVolume
|
||||
|
|
|
|||
|
|
@ -1,7 +1,12 @@
|
|||
GstChildProxy GstObject
|
||||
GFile GObject
|
||||
GDesktopAppInfoLookup GObject
|
||||
GDrive GObject
|
||||
GMount GObject
|
||||
GVolume GObject
|
||||
GstTagSetter GstObject GstElement
|
||||
GstImplementsInterface GstObject GstElement
|
||||
GstMixer GstObject GstImplementsInterface GstElement
|
||||
GstTuner GstObject GstImplementsInterface GstElement
|
||||
GstXOverlay GstObject GstImplementsInterface GstElement
|
||||
GstColorBalance GstObject GstImplementsInterface GstElement
|
||||
GstTuner GstObject GstImplementsInterface GstElement
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Adds multiple streams</description>
|
||||
<filename>../../gst/adder/.libs/libgstadder.so</filename>
|
||||
<basename>libgstadder.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>ALSA plugin library</description>
|
||||
<filename>../../ext/alsa/.libs/libgstalsa.so</filename>
|
||||
<basename>libgstalsa.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Elements used to communicate with applications</description>
|
||||
<filename>../../gst/app/.libs/libgstapp.so</filename>
|
||||
<basename>libgstapp.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Convert audio to different formats</description>
|
||||
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
|
||||
<basename>libgstaudioconvert.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Adjusts audio frames</description>
|
||||
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
|
||||
<basename>libgstaudiorate.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -1,12 +1,12 @@
|
|||
<plugin>
|
||||
<name>audioresample</name>
|
||||
<description>Resamples audio</description>
|
||||
<filename>../../gst/speexresample/.libs/libgstaudioresample.so</filename>
|
||||
<filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
|
||||
<basename>libgstaudioresample.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Creates audio test signals of given frequency and volume</description>
|
||||
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
|
||||
<basename>libgstaudiotestsrc.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Read audio from CD in paranoid mode</description>
|
||||
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
|
||||
<basename>libgstcdparanoia.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>GPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>decoder bin</description>
|
||||
<filename>../../gst/playback/.libs/libgstdecodebin.so</filename>
|
||||
<basename>libgstdecodebin.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>colorspace conversion copied from FFMpeg 0.4.9-pre1</description>
|
||||
<filename>../../gst/ffmpegcolorspace/.libs/libgstffmpegcolorspace.so</filename>
|
||||
<basename>libgstffmpegcolorspace.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>FFMpeg</package>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Payload/depayload GDP packets</description>
|
||||
<filename>../../gst/gdp/.libs/libgstgdp.so</filename>
|
||||
<basename>libgstgdp.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>GIO elements</description>
|
||||
<filename>../../ext/gio/.libs/libgstgio.so</filename>
|
||||
<basename>libgstgio.so</basename>
|
||||
<version>0.10.21.1</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>elements to read from and write to Gnome-VFS uri's</description>
|
||||
<filename>../../ext/gnomevfs/.libs/libgstgnomevfs.so</filename>
|
||||
<basename>libgstgnomevfs.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>libvisual visualization plugins</description>
|
||||
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
|
||||
<basename>libgstlibvisual.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
|
||||
<filename>../../ext/ogg/.libs/libgstogg.so</filename>
|
||||
<basename>libgstogg.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Pango-based text rendering and overlay</description>
|
||||
<filename>../../ext/pango/.libs/libgstpango.so</filename>
|
||||
<basename>libgstpango.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>various playback elements</description>
|
||||
<filename>../../gst/playback/.libs/libgstplaybin.so</filename>
|
||||
<basename>libgstplaybin.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Queue newer version</description>
|
||||
<filename>../../gst/playback/.libs/libgstqueue2.so</filename>
|
||||
<basename>libgstqueue2.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Subtitle parsing</description>
|
||||
<filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
|
||||
<basename>libgstsubparse.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>transfer data over the network via TCP</description>
|
||||
<filename>../../gst/tcp/.libs/libgsttcp.so</filename>
|
||||
<basename>libgsttcp.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Theora plugin library</description>
|
||||
<filename>../../ext/theora/.libs/libgsttheora.so</filename>
|
||||
<basename>libgsttheora.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>default typefind functions</description>
|
||||
<filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename>
|
||||
<basename>libgsttypefindfunctions.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>URI Decoder bin</description>
|
||||
<filename>../../gst/playback/.libs/libgstdecodebin2.so</filename>
|
||||
<basename>libgstdecodebin2.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>elements for Video 4 Linux</description>
|
||||
<filename>../../sys/v4l/.libs/libgstvideo4linux.so</filename>
|
||||
<basename>libgstvideo4linux.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Adjusts video frames</description>
|
||||
<filename>../../gst/videorate/.libs/libgstvideorate.so</filename>
|
||||
<basename>libgstvideorate.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Resizes video</description>
|
||||
<filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename>
|
||||
<basename>libgstvideoscale.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Creates a test video stream</description>
|
||||
<filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename>
|
||||
<basename>libgstvideotestsrc.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>plugin for controlling audio volume</description>
|
||||
<filename>../../gst/volume/.libs/libgstvolume.so</filename>
|
||||
<basename>libgstvolume.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>Vorbis plugin library</description>
|
||||
<filename>../../ext/vorbis/.libs/libgstvorbis.so</filename>
|
||||
<basename>libgstvorbis.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>X11 video output element based on standard Xlib calls</description>
|
||||
<filename>../../sys/ximage/.libs/libgstximagesink.so</filename>
|
||||
<basename>libgstximagesink.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -3,10 +3,10 @@
|
|||
<description>XFree86 video output plugin using Xv extension</description>
|
||||
<filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename>
|
||||
<basename>libgstxvimagesink.so</basename>
|
||||
<version>0.10.22</version>
|
||||
<version>0.10.22.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
<package>GStreamer Base Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
|
|
|
|||
|
|
@ -1,21 +1,37 @@
|
|||
plugin_LTLIBRARIES = libgstaudioresample.la
|
||||
|
||||
resample_SOURCES = \
|
||||
functable.c \
|
||||
resample.c \
|
||||
resample_functable.c \
|
||||
resample_ref.c \
|
||||
resample_chunk.c \
|
||||
resample.h \
|
||||
buffer.c
|
||||
libgstaudioresample_la_SOURCES = \
|
||||
gstaudioresample.c \
|
||||
speex_resampler_int.c \
|
||||
speex_resampler_float.c \
|
||||
speex_resampler_double.c
|
||||
|
||||
libgstaudioresample_la_CFLAGS = \
|
||||
$(GST_PLUGINS_BASE_CFLAGS) \
|
||||
$(GST_BASE_CFLAGS) \
|
||||
$(GST_CFLAGS) \
|
||||
$(LIBOIL_CFLAGS)
|
||||
|
||||
libgstaudioresample_la_LIBADD = \
|
||||
$(GST_PLUGINS_BASE_LIBS) \
|
||||
$(GST_BASE_LIBS) \
|
||||
$(GST_LIBS) \
|
||||
$(LIBOIL_LIBS) \
|
||||
$(LIBM)
|
||||
|
||||
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
|
||||
libgstaudioresample_la_LIBTOOLFLAGS = --tag=disable-static
|
||||
|
||||
noinst_HEADERS = \
|
||||
arch.h \
|
||||
fixed_arm4.h \
|
||||
fixed_arm5e.h \
|
||||
fixed_bfin.h \
|
||||
fixed_debug.h \
|
||||
fixed_generic.h \
|
||||
gstaudioresample.h \
|
||||
functable.h \
|
||||
debug.h \
|
||||
buffer.h
|
||||
resample.c \
|
||||
resample_sse.h \
|
||||
speex_resampler.h \
|
||||
speex_resampler_wrapper.h
|
||||
|
||||
libgstaudioresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
|
||||
libgstaudioresample_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBOIL_CFLAGS)
|
||||
libgstaudioresample_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(LIBOIL_LIBS)
|
||||
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
|
||||
|
|
|
|||
|
|
@ -1,253 +0,0 @@
|
|||
|
||||
#ifndef HAVE_CONFIG_H
|
||||
#include "config.h"
|
||||
#endif
|
||||
|
||||
#include <glib.h>
|
||||
#include <string.h>
|
||||
|
||||
#include "buffer.h"
|
||||
#include "debug.h"
|
||||
|
||||
static void audioresample_buffer_free_mem (AudioresampleBuffer * buffer,
|
||||
void *);
|
||||
static void audioresample_buffer_free_subbuffer (AudioresampleBuffer * buffer,
|
||||
void *priv);
|
||||
|
||||
|
||||
AudioresampleBuffer *
|
||||
audioresample_buffer_new (void)
|
||||
{
|
||||
AudioresampleBuffer *buffer;
|
||||
|
||||
buffer = g_new0 (AudioresampleBuffer, 1);
|
||||
buffer->ref_count = 1;
|
||||
return buffer;
|
||||
}
|
||||
|
||||
AudioresampleBuffer *
|
||||
audioresample_buffer_new_and_alloc (int size)
|
||||
{
|
||||
AudioresampleBuffer *buffer = audioresample_buffer_new ();
|
||||
|
||||
buffer->data = g_malloc (size);
|
||||
buffer->length = size;
|
||||
buffer->free = audioresample_buffer_free_mem;
|
||||
|
||||
return buffer;
|
||||
}
|
||||
|
||||
AudioresampleBuffer *
|
||||
audioresample_buffer_new_with_data (void *data, int size)
|
||||
{
|
||||
AudioresampleBuffer *buffer = audioresample_buffer_new ();
|
||||
|
||||
buffer->data = data;
|
||||
buffer->length = size;
|
||||
buffer->free = audioresample_buffer_free_mem;
|
||||
|
||||
return buffer;
|
||||
}
|
||||
|
||||
AudioresampleBuffer *
|
||||
audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer, int offset,
|
||||
int length)
|
||||
{
|
||||
AudioresampleBuffer *subbuffer = audioresample_buffer_new ();
|
||||
|
||||
if (buffer->parent) {
|
||||
audioresample_buffer_ref (buffer->parent);
|
||||
subbuffer->parent = buffer->parent;
|
||||
} else {
|
||||
audioresample_buffer_ref (buffer);
|
||||
subbuffer->parent = buffer;
|
||||
}
|
||||
subbuffer->data = buffer->data + offset;
|
||||
subbuffer->length = length;
|
||||
subbuffer->free = audioresample_buffer_free_subbuffer;
|
||||
|
||||
return subbuffer;
|
||||
}
|
||||
|
||||
void
|
||||
audioresample_buffer_ref (AudioresampleBuffer * buffer)
|
||||
{
|
||||
buffer->ref_count++;
|
||||
}
|
||||
|
||||
void
|
||||
audioresample_buffer_unref (AudioresampleBuffer * buffer)
|
||||
{
|
||||
buffer->ref_count--;
|
||||
if (buffer->ref_count == 0) {
|
||||
if (buffer->free)
|
||||
buffer->free (buffer, buffer->priv);
|
||||
g_free (buffer);
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
audioresample_buffer_free_mem (AudioresampleBuffer * buffer, void *priv)
|
||||
{
|
||||
g_free (buffer->data);
|
||||
}
|
||||
|
||||
static void
|
||||
audioresample_buffer_free_subbuffer (AudioresampleBuffer * buffer, void *priv)
|
||||
{
|
||||
audioresample_buffer_unref (buffer->parent);
|
||||
}
|
||||
|
||||
|
||||
AudioresampleBufferQueue *
|
||||
audioresample_buffer_queue_new (void)
|
||||
{
|
||||
return g_new0 (AudioresampleBufferQueue, 1);
|
||||
}
|
||||
|
||||
int
|
||||
audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue)
|
||||
{
|
||||
return queue->depth;
|
||||
}
|
||||
|
||||
int
|
||||
audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue)
|
||||
{
|
||||
return queue->offset;
|
||||
}
|
||||
|
||||
void
|
||||
audioresample_buffer_queue_free (AudioresampleBufferQueue * queue)
|
||||
{
|
||||
GList *g;
|
||||
|
||||
for (g = g_list_first (queue->buffers); g; g = g_list_next (g)) {
|
||||
audioresample_buffer_unref ((AudioresampleBuffer *) g->data);
|
||||
}
|
||||
g_list_free (queue->buffers);
|
||||
g_free (queue);
|
||||
}
|
||||
|
||||
void
|
||||
audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
|
||||
AudioresampleBuffer * buffer)
|
||||
{
|
||||
queue->buffers = g_list_append (queue->buffers, buffer);
|
||||
queue->depth += buffer->length;
|
||||
}
|
||||
|
||||
AudioresampleBuffer *
|
||||
audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int length)
|
||||
{
|
||||
GList *g;
|
||||
AudioresampleBuffer *newbuffer;
|
||||
AudioresampleBuffer *buffer;
|
||||
AudioresampleBuffer *subbuffer;
|
||||
|
||||
g_return_val_if_fail (length > 0, NULL);
|
||||
|
||||
if (queue->depth < length) {
|
||||
return NULL;
|
||||
}
|
||||
|
||||
RESAMPLE_LOG ("pulling %d, %d available", length, queue->depth);
|
||||
|
||||
g = g_list_first (queue->buffers);
|
||||
buffer = g->data;
|
||||
|
||||
if (buffer->length > length) {
|
||||
newbuffer = audioresample_buffer_new_subbuffer (buffer, 0, length);
|
||||
|
||||
subbuffer = audioresample_buffer_new_subbuffer (buffer, length,
|
||||
buffer->length - length);
|
||||
g->data = subbuffer;
|
||||
audioresample_buffer_unref (buffer);
|
||||
} else {
|
||||
int offset = 0;
|
||||
|
||||
newbuffer = audioresample_buffer_new_and_alloc (length);
|
||||
|
||||
while (offset < length) {
|
||||
g = g_list_first (queue->buffers);
|
||||
buffer = g->data;
|
||||
|
||||
if (buffer->length > length - offset) {
|
||||
int n = length - offset;
|
||||
|
||||
memcpy (newbuffer->data + offset, buffer->data, n);
|
||||
subbuffer =
|
||||
audioresample_buffer_new_subbuffer (buffer, n, buffer->length - n);
|
||||
g->data = subbuffer;
|
||||
audioresample_buffer_unref (buffer);
|
||||
offset += n;
|
||||
} else {
|
||||
memcpy (newbuffer->data + offset, buffer->data, buffer->length);
|
||||
|
||||
queue->buffers = g_list_delete_link (queue->buffers, g);
|
||||
offset += buffer->length;
|
||||
audioresample_buffer_unref (buffer);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
queue->depth -= length;
|
||||
queue->offset += length;
|
||||
|
||||
return newbuffer;
|
||||
}
|
||||
|
||||
AudioresampleBuffer *
|
||||
audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int length)
|
||||
{
|
||||
GList *g;
|
||||
AudioresampleBuffer *newbuffer;
|
||||
AudioresampleBuffer *buffer;
|
||||
int offset = 0;
|
||||
|
||||
g_return_val_if_fail (length > 0, NULL);
|
||||
|
||||
if (queue->depth < length) {
|
||||
return NULL;
|
||||
}
|
||||
|
||||
RESAMPLE_LOG ("peeking %d, %d available", length, queue->depth);
|
||||
|
||||
g = g_list_first (queue->buffers);
|
||||
buffer = g->data;
|
||||
if (buffer->length > length) {
|
||||
newbuffer = audioresample_buffer_new_subbuffer (buffer, 0, length);
|
||||
} else {
|
||||
newbuffer = audioresample_buffer_new_and_alloc (length);
|
||||
while (offset < length) {
|
||||
buffer = g->data;
|
||||
|
||||
if (buffer->length > length - offset) {
|
||||
int n = length - offset;
|
||||
|
||||
memcpy (newbuffer->data + offset, buffer->data, n);
|
||||
offset += n;
|
||||
} else {
|
||||
memcpy (newbuffer->data + offset, buffer->data, buffer->length);
|
||||
offset += buffer->length;
|
||||
}
|
||||
g = g_list_next (g);
|
||||
}
|
||||
}
|
||||
|
||||
return newbuffer;
|
||||
}
|
||||
|
||||
void
|
||||
audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue)
|
||||
{
|
||||
GList *g;
|
||||
|
||||
for (g = g_list_first (queue->buffers); g; g = g_list_next (g)) {
|
||||
audioresample_buffer_unref ((AudioresampleBuffer *) g->data);
|
||||
}
|
||||
g_list_free (queue->buffers);
|
||||
queue->buffers = NULL;
|
||||
queue->depth = 0;
|
||||
queue->offset = 0;
|
||||
}
|
||||
|
|
@ -1,51 +0,0 @@
|
|||
|
||||
#ifndef __AUDIORESAMPLE_BUFFER_H__
|
||||
#define __AUDIORESAMPLE_BUFFER_H__
|
||||
|
||||
#include <glib.h>
|
||||
|
||||
typedef struct _AudioresampleBuffer AudioresampleBuffer;
|
||||
typedef struct _AudioresampleBufferQueue AudioresampleBufferQueue;
|
||||
|
||||
struct _AudioresampleBuffer
|
||||
{
|
||||
unsigned char *data;
|
||||
int length;
|
||||
|
||||
int ref_count;
|
||||
|
||||
AudioresampleBuffer *parent;
|
||||
|
||||
void (*free) (AudioresampleBuffer *, void *);
|
||||
void *priv;
|
||||
void *priv2;
|
||||
};
|
||||
|
||||
struct _AudioresampleBufferQueue
|
||||
{
|
||||
GList *buffers;
|
||||
int depth;
|
||||
int offset;
|
||||
};
|
||||
|
||||
AudioresampleBuffer * audioresample_buffer_new (void);
|
||||
AudioresampleBuffer * audioresample_buffer_new_and_alloc (int size);
|
||||
AudioresampleBuffer * audioresample_buffer_new_with_data (void *data, int size);
|
||||
AudioresampleBuffer * audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer,
|
||||
int offset,
|
||||
int length);
|
||||
void audioresample_buffer_ref (AudioresampleBuffer * buffer);
|
||||
void audioresample_buffer_unref (AudioresampleBuffer * buffer);
|
||||
|
||||
AudioresampleBufferQueue *
|
||||
audioresample_buffer_queue_new (void);
|
||||
void audioresample_buffer_queue_free (AudioresampleBufferQueue * queue);
|
||||
int audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue);
|
||||
int audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue);
|
||||
void audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
|
||||
AudioresampleBuffer * buffer);
|
||||
AudioresampleBuffer * audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int len);
|
||||
AudioresampleBuffer * audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int len);
|
||||
void audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue);
|
||||
|
||||
#endif
|
||||
|
|
@ -1,65 +0,0 @@
|
|||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include "config.h"
|
||||
#endif
|
||||
|
||||
#include <glib.h>
|
||||
#include <stdio.h>
|
||||
#include <debug.h>
|
||||
|
||||
static const char *resample_debug_level_names[] = {
|
||||
"NONE",
|
||||
"ERROR",
|
||||
"WARNING",
|
||||
"INFO",
|
||||
"DEBUG",
|
||||
"LOG"
|
||||
};
|
||||
|
||||
static int resample_debug_level = RESAMPLE_LEVEL_ERROR;
|
||||
|
||||
void
|
||||
resample_debug_log (int level, const char *file, const char *function,
|
||||
int line, const char *format, ...)
|
||||
{
|
||||
#ifndef GLIB_COMPAT
|
||||
va_list varargs;
|
||||
char *s;
|
||||
|
||||
if (level > resample_debug_level)
|
||||
return;
|
||||
|
||||
va_start (varargs, format);
|
||||
s = g_strdup_vprintf (format, varargs);
|
||||
va_end (varargs);
|
||||
|
||||
fprintf (stderr, "RESAMPLE: %s: %s(%d): %s: %s\n",
|
||||
resample_debug_level_names[level], file, line, function, s);
|
||||
g_free (s);
|
||||
#else
|
||||
va_list varargs;
|
||||
char s[1000];
|
||||
|
||||
if (level > resample_debug_level)
|
||||
return;
|
||||
|
||||
va_start (varargs, format);
|
||||
vsnprintf (s, 999, format, varargs);
|
||||
va_end (varargs);
|
||||
|
||||
fprintf (stderr, "RESAMPLE: %s: %s(%d): %s: %s\n",
|
||||
resample_debug_level_names[level], file, line, function, s);
|
||||
#endif
|
||||
}
|
||||
|
||||
void
|
||||
resample_debug_set_level (int level)
|
||||
{
|
||||
resample_debug_level = level;
|
||||
}
|
||||
|
||||
int
|
||||
resample_debug_get_level (void)
|
||||
{
|
||||
return resample_debug_level;
|
||||
}
|
||||
|
|
@ -1,51 +0,0 @@
|
|||
|
||||
#ifndef __RESAMPLE_DEBUG_H__
|
||||
#define __RESAMPLE_DEBUG_H__
|
||||
|
||||
#if 0
|
||||
enum
|
||||
{
|
||||
RESAMPLE_LEVEL_NONE = 0,
|
||||
RESAMPLE_LEVEL_ERROR,
|
||||
RESAMPLE_LEVEL_WARNING,
|
||||
RESAMPLE_LEVEL_INFO,
|
||||
RESAMPLE_LEVEL_DEBUG,
|
||||
RESAMPLE_LEVEL_LOG
|
||||
};
|
||||
|
||||
#define RESAMPLE_ERROR(...) \
|
||||
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_ERROR, __VA_ARGS__)
|
||||
#define RESAMPLE_WARNING(...) \
|
||||
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_WARNING, __VA_ARGS__)
|
||||
#define RESAMPLE_INFO(...) \
|
||||
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_INFO, __VA_ARGS__)
|
||||
#define RESAMPLE_DEBUG(...) \
|
||||
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_DEBUG, __VA_ARGS__)
|
||||
#define RESAMPLE_LOG(...) \
|
||||
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_LOG, __VA_ARGS__)
|
||||
|
||||
#define RESAMPLE_DEBUG_LEVEL(level,...) \
|
||||
resample_debug_log ((level), __FILE__, __FUNCTION__, __LINE__, __VA_ARGS__)
|
||||
|
||||
void resample_debug_log (int level, const char *file, const char *function,
|
||||
int line, const char *format, ...);
|
||||
void resample_debug_set_level (int level);
|
||||
int resample_debug_get_level (void);
|
||||
#else
|
||||
|
||||
#include <gst/gst.h>
|
||||
|
||||
GST_DEBUG_CATEGORY_EXTERN (libaudioresample_debug);
|
||||
#define GST_CAT_DEFAULT libaudioresample_debug
|
||||
|
||||
#define RESAMPLE_ERROR GST_ERROR
|
||||
#define RESAMPLE_WARNING GST_WARNING
|
||||
#define RESAMPLE_INFO GST_INFO
|
||||
#define RESAMPLE_DEBUG GST_DEBUG
|
||||
#define RESAMPLE_LOG GST_LOG
|
||||
|
||||
#define resample_debug_set_level(x) do { } while (0)
|
||||
|
||||
#endif
|
||||
|
||||
#endif
|
||||
|
|
@ -1,254 +0,0 @@
|
|||
/* Resampling library
|
||||
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include <config.h>
|
||||
#endif
|
||||
|
||||
#include <string.h>
|
||||
#include <math.h>
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
#include "functable.h"
|
||||
#include "debug.h"
|
||||
|
||||
|
||||
|
||||
void
|
||||
functable_func_sinc (double *fx, double *dfx, double x, void *closure)
|
||||
{
|
||||
if (x == 0) {
|
||||
*fx = 1;
|
||||
*dfx = 0;
|
||||
return;
|
||||
}
|
||||
|
||||
*fx = sin (x) / x;
|
||||
*dfx = (cos (x) - sin (x) / x) / x;
|
||||
}
|
||||
|
||||
void
|
||||
functable_func_boxcar (double *fx, double *dfx, double x, void *closure)
|
||||
{
|
||||
double width = *(double *) closure;
|
||||
|
||||
if (x < width && x > -width) {
|
||||
*fx = 1;
|
||||
} else {
|
||||
*fx = 0;
|
||||
}
|
||||
*dfx = 0;
|
||||
}
|
||||
|
||||
void
|
||||
functable_func_hanning (double *fx, double *dfx, double x, void *closure)
|
||||
{
|
||||
double width = *(double *) closure;
|
||||
|
||||
if (x < width && x > -width) {
|
||||
x /= width;
|
||||
*fx = (1 - x * x) * (1 - x * x);
|
||||
*dfx = -2 * 2 * x / width * (1 - x * x);
|
||||
} else {
|
||||
*fx = 0;
|
||||
*dfx = 0;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
Functable *
|
||||
functable_new (void)
|
||||
{
|
||||
Functable *ft;
|
||||
|
||||
ft = malloc (sizeof (Functable));
|
||||
memset (ft, 0, sizeof (Functable));
|
||||
|
||||
return ft;
|
||||
}
|
||||
|
||||
void
|
||||
functable_free (Functable * ft)
|
||||
{
|
||||
free (ft);
|
||||
}
|
||||
|
||||
void
|
||||
functable_set_length (Functable * t, int length)
|
||||
{
|
||||
t->length = length;
|
||||
}
|
||||
|
||||
void
|
||||
functable_set_offset (Functable * t, double offset)
|
||||
{
|
||||
t->offset = offset;
|
||||
}
|
||||
|
||||
void
|
||||
functable_set_multiplier (Functable * t, double multiplier)
|
||||
{
|
||||
t->multiplier = multiplier;
|
||||
}
|
||||
|
||||
void
|
||||
functable_calculate (Functable * t, FunctableFunc func, void *closure)
|
||||
{
|
||||
int i;
|
||||
double x;
|
||||
|
||||
if (t->fx)
|
||||
free (t->fx);
|
||||
if (t->dfx)
|
||||
free (t->dfx);
|
||||
|
||||
t->fx = malloc (sizeof (double) * (t->length + 1));
|
||||
t->dfx = malloc (sizeof (double) * (t->length + 1));
|
||||
|
||||
t->inv_multiplier = 1.0 / t->multiplier;
|
||||
|
||||
for (i = 0; i < t->length + 1; i++) {
|
||||
x = t->offset + t->multiplier * i;
|
||||
|
||||
func (&t->fx[i], &t->dfx[i], x, closure);
|
||||
}
|
||||
}
|
||||
|
||||
void
|
||||
functable_calculate_multiply (Functable * t, FunctableFunc func, void *closure)
|
||||
{
|
||||
int i;
|
||||
double x;
|
||||
|
||||
for (i = 0; i < t->length + 1; i++) {
|
||||
double afx, adfx, bfx, bdfx;
|
||||
|
||||
afx = t->fx[i];
|
||||
adfx = t->dfx[i];
|
||||
x = t->offset + t->multiplier * i;
|
||||
func (&bfx, &bdfx, x, closure);
|
||||
t->fx[i] = afx * bfx;
|
||||
t->dfx[i] = afx * bdfx + adfx * bfx;
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
double
|
||||
functable_evaluate (Functable * t, double x)
|
||||
{
|
||||
int i;
|
||||
double f0, f1, w0, w1;
|
||||
double x2, x3;
|
||||
double w;
|
||||
|
||||
if (x < t->offset || x > (t->offset + t->length * t->multiplier)) {
|
||||
RESAMPLE_DEBUG ("x out of range %g", x);
|
||||
}
|
||||
|
||||
x -= t->offset;
|
||||
x *= t->inv_multiplier;
|
||||
i = floor (x);
|
||||
x -= i;
|
||||
|
||||
x2 = x * x;
|
||||
x3 = x2 * x;
|
||||
|
||||
f1 = 3 * x2 - 2 * x3;
|
||||
f0 = 1 - f1;
|
||||
w0 = (x - 2 * x2 + x3) * t->multiplier;
|
||||
w1 = (-x2 + x3) * t->multiplier;
|
||||
|
||||
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
|
||||
|
||||
/*w = t->fx[i] * (1-x) + t->fx[i+1] * x; */
|
||||
|
||||
return w;
|
||||
}
|
||||
|
||||
|
||||
double
|
||||
functable_fir (Functable * t, double x, int n, double *data, int len)
|
||||
{
|
||||
int i, j;
|
||||
double f0, f1, w0, w1;
|
||||
double x2, x3;
|
||||
double w;
|
||||
double sum;
|
||||
|
||||
x -= t->offset;
|
||||
x /= t->multiplier;
|
||||
i = floor (x);
|
||||
x -= i;
|
||||
|
||||
x2 = x * x;
|
||||
x3 = x2 * x;
|
||||
|
||||
f1 = 3 * x2 - 2 * x3;
|
||||
f0 = 1 - f1;
|
||||
w0 = (x - 2 * x2 + x3) * t->multiplier;
|
||||
w1 = (-x2 + x3) * t->multiplier;
|
||||
|
||||
sum = 0;
|
||||
for (j = 0; j < len; j++) {
|
||||
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
|
||||
sum += data[j * 2] * w;
|
||||
i += n;
|
||||
}
|
||||
|
||||
return sum;
|
||||
}
|
||||
|
||||
void
|
||||
functable_fir2 (Functable * t, double *r0, double *r1, double x,
|
||||
int n, double *data, int len)
|
||||
{
|
||||
int i, j;
|
||||
double f0, f1, w0, w1;
|
||||
double x2, x3;
|
||||
double w;
|
||||
double sum0, sum1;
|
||||
double floor_x;
|
||||
|
||||
x -= t->offset;
|
||||
x *= t->inv_multiplier;
|
||||
floor_x = floor (x);
|
||||
i = floor_x;
|
||||
x -= floor_x;
|
||||
|
||||
x2 = x * x;
|
||||
x3 = x2 * x;
|
||||
|
||||
f1 = 3 * x2 - 2 * x3;
|
||||
f0 = 1 - f1;
|
||||
w0 = (x - 2 * x2 + x3) * t->multiplier;
|
||||
w1 = (-x2 + x3) * t->multiplier;
|
||||
|
||||
sum0 = 0;
|
||||
sum1 = 0;
|
||||
for (j = 0; j < len; j++) {
|
||||
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
|
||||
sum0 += data[j * 2] * w;
|
||||
sum1 += data[j * 2 + 1] * w;
|
||||
i += n;
|
||||
}
|
||||
|
||||
*r0 = sum0;
|
||||
*r1 = sum1;
|
||||
}
|
||||
|
|
@ -1,61 +0,0 @@
|
|||
/* Resampling library
|
||||
* Copyright (C) <2001> David Schleef <ds@schleef.org>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
|
||||
#ifndef __FUNCTABLE_H__
|
||||
#define __FUNCTABLE_H__
|
||||
|
||||
typedef void FunctableFunc (double *fx, double *dfx, double x, void *closure);
|
||||
|
||||
typedef struct _Functable Functable;
|
||||
struct _Functable {
|
||||
int length;
|
||||
|
||||
double offset;
|
||||
double multiplier;
|
||||
|
||||
double inv_multiplier;
|
||||
|
||||
double *fx;
|
||||
double *dfx;
|
||||
};
|
||||
|
||||
Functable *functable_new (void);
|
||||
void functable_setup (Functable *t);
|
||||
void functable_free (Functable *t);
|
||||
|
||||
void functable_set_length (Functable *t, int length);
|
||||
void functable_set_offset (Functable *t, double offset);
|
||||
void functable_set_multiplier (Functable *t, double multiplier);
|
||||
void functable_calculate (Functable *t, FunctableFunc func, void *closure);
|
||||
void functable_calculate_multiply (Functable *t, FunctableFunc func, void *closure);
|
||||
|
||||
|
||||
double functable_evaluate (Functable *t,double x);
|
||||
|
||||
double functable_fir(Functable *t,double x0,int n,double *data,int len);
|
||||
void functable_fir2(Functable *t,double *r0, double *r1, double x0,
|
||||
int n,double *data,int len);
|
||||
|
||||
void functable_func_sinc(double *fx, double *dfx, double x, void *closure);
|
||||
void functable_func_boxcar(double *fx, double *dfx, double x, void *closure);
|
||||
void functable_func_hanning(double *fx, double *dfx, double x, void *closure);
|
||||
|
||||
#endif /* __PRIVATE_H__ */
|
||||
|
||||
File diff suppressed because it is too large
Load diff
|
|
@ -1,5 +1,6 @@
|
|||
/* GStreamer
|
||||
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
|
||||
* Copyright (C) <2007-2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
|
|
@ -18,62 +19,72 @@
|
|||
*/
|
||||
|
||||
|
||||
#ifndef __AUDIORESAMPLE_H__
|
||||
#define __AUDIORESAMPLE_H__
|
||||
#ifndef __AUDIO_RESAMPLE_H__
|
||||
#define __AUDIO_RESAMPLE_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstbasetransform.h>
|
||||
#include <gst/audio/audio.h>
|
||||
|
||||
#include "resample.h"
|
||||
#include "speex_resampler_wrapper.h"
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
#define GST_TYPE_AUDIORESAMPLE \
|
||||
(gst_audioresample_get_type())
|
||||
#define GST_AUDIORESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,GstAudioresample))
|
||||
#define GST_AUDIORESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,GstAudioresampleClass))
|
||||
#define GST_IS_AUDIORESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
|
||||
#define GST_IS_AUDIORESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
|
||||
#define GST_TYPE_AUDIO_RESAMPLE \
|
||||
(gst_audio_resample_get_type())
|
||||
#define GST_AUDIO_RESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RESAMPLE,GstAudioResample))
|
||||
#define GST_AUDIO_RESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RESAMPLE,GstAudioResampleClass))
|
||||
#define GST_IS_AUDIO_RESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RESAMPLE))
|
||||
#define GST_IS_AUDIO_RESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RESAMPLE))
|
||||
|
||||
typedef struct _GstAudioresample GstAudioresample;
|
||||
typedef struct _GstAudioresampleClass GstAudioresampleClass;
|
||||
typedef struct _GstAudioResample GstAudioResample;
|
||||
typedef struct _GstAudioResampleClass GstAudioResampleClass;
|
||||
|
||||
/**
|
||||
* GstAudioresample:
|
||||
* GstAudioResample:
|
||||
*
|
||||
* Opaque data structure.
|
||||
*/
|
||||
struct _GstAudioresample {
|
||||
struct _GstAudioResample {
|
||||
GstBaseTransform element;
|
||||
|
||||
/* <private> */
|
||||
|
||||
GstCaps *srccaps, *sinkcaps;
|
||||
|
||||
gboolean passthru;
|
||||
gboolean need_discont;
|
||||
|
||||
guint64 offset;
|
||||
guint64 ts_offset;
|
||||
guint64 next_offset;
|
||||
GstClockTime next_ts;
|
||||
GstClockTime prev_ts, prev_duration;
|
||||
int channels;
|
||||
GstClockTime next_upstream_ts;
|
||||
|
||||
int i_rate;
|
||||
int o_rate;
|
||||
int filter_length;
|
||||
gint channels;
|
||||
gint inrate;
|
||||
gint outrate;
|
||||
gint quality;
|
||||
gint width;
|
||||
gboolean fp;
|
||||
|
||||
ResampleState * resample;
|
||||
guint8 *tmp_in;
|
||||
guint tmp_in_size;
|
||||
|
||||
guint8 *tmp_out;
|
||||
guint tmp_out_size;
|
||||
|
||||
SpeexResamplerState *state;
|
||||
const SpeexResampleFuncs *funcs;
|
||||
};
|
||||
|
||||
struct _GstAudioresampleClass {
|
||||
struct _GstAudioResampleClass {
|
||||
GstBaseTransformClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_audioresample_get_type(void);
|
||||
GType gst_audio_resample_get_type(void);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
#endif /* __AUDIORESAMPLE_H__ */
|
||||
#endif /* __AUDIO_RESAMPLE_H__ */
|
||||
|
|
|
|||
File diff suppressed because it is too large
Load diff
|
|
@ -1,128 +0,0 @@
|
|||
/* Resampling library
|
||||
* Copyright (C) <2001> David Schleef <ds@schleef.org>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
|
||||
#ifndef __RESAMPLE_H__
|
||||
#define __RESAMPLE_H__
|
||||
|
||||
#include "functable.h"
|
||||
#include "buffer.h"
|
||||
|
||||
#ifndef M_PI
|
||||
#define M_PI 3.14159265358979323846
|
||||
#endif
|
||||
|
||||
#ifdef WIN32
|
||||
#define rint(x) (floor((x)+0.5))
|
||||
#endif
|
||||
|
||||
typedef enum {
|
||||
RESAMPLE_FORMAT_S16 = 0,
|
||||
RESAMPLE_FORMAT_S32,
|
||||
RESAMPLE_FORMAT_F32,
|
||||
RESAMPLE_FORMAT_F64
|
||||
} ResampleFormat;
|
||||
|
||||
typedef void (*ResampleCallback) (void *);
|
||||
|
||||
typedef struct _ResampleState ResampleState;
|
||||
|
||||
struct _ResampleState {
|
||||
/* parameters */
|
||||
|
||||
int n_channels;
|
||||
ResampleFormat format;
|
||||
|
||||
int filter_length;
|
||||
|
||||
double i_rate;
|
||||
double o_rate;
|
||||
|
||||
int method;
|
||||
|
||||
/* internal parameters */
|
||||
|
||||
int need_reinit;
|
||||
|
||||
double halftaps;
|
||||
|
||||
/* filter state */
|
||||
|
||||
unsigned char *o_buf;
|
||||
int o_size;
|
||||
|
||||
AudioresampleBufferQueue *queue;
|
||||
int eos;
|
||||
int started;
|
||||
|
||||
int sample_size;
|
||||
|
||||
unsigned char *buffer;
|
||||
int buffer_len;
|
||||
int buffer_filled;
|
||||
|
||||
double i_start;
|
||||
double o_start;
|
||||
|
||||
double i_inc;
|
||||
double o_inc;
|
||||
|
||||
double sinc_scale;
|
||||
|
||||
double i_end;
|
||||
double o_end;
|
||||
|
||||
int i_samples;
|
||||
int o_samples;
|
||||
|
||||
//void *i_buf;
|
||||
|
||||
Functable *ft;
|
||||
|
||||
double *out_tmp;
|
||||
};
|
||||
|
||||
void resample_init (void);
|
||||
void resample_cleanup (void);
|
||||
|
||||
ResampleState *resample_new (void);
|
||||
void resample_free (ResampleState *state);
|
||||
|
||||
void resample_add_input_data (ResampleState * r, void *data, int size,
|
||||
ResampleCallback free_func, void *closure);
|
||||
void resample_input_eos (ResampleState *r);
|
||||
void resample_input_flush (ResampleState *r);
|
||||
void resample_input_pushthrough (ResampleState *r);
|
||||
|
||||
int resample_get_output_size_for_input (ResampleState * r, int size);
|
||||
int resample_get_input_size_for_output (ResampleState * r, int size);
|
||||
|
||||
int resample_get_output_size (ResampleState *r);
|
||||
int resample_get_output_data (ResampleState *r, void *data, int size);
|
||||
|
||||
void resample_set_filter_length (ResampleState *r, int length);
|
||||
void resample_set_input_rate (ResampleState *r, double rate);
|
||||
void resample_set_output_rate (ResampleState *r, double rate);
|
||||
void resample_set_n_channels (ResampleState *r, int n_channels);
|
||||
void resample_set_format (ResampleState *r, ResampleFormat format);
|
||||
void resample_set_method (ResampleState *r, int method);
|
||||
int resample_format_size (ResampleFormat format);
|
||||
|
||||
#endif /* __RESAMPLE_H__ */
|
||||
|
||||
|
|
@ -1,209 +0,0 @@
|
|||
/* Resampling library
|
||||
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include <config.h>
|
||||
#endif
|
||||
|
||||
|
||||
#include <string.h>
|
||||
#include <math.h>
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <limits.h>
|
||||
#include <liboil/liboil.h>
|
||||
|
||||
#include "resample.h"
|
||||
#include "buffer.h"
|
||||
#include "debug.h"
|
||||
|
||||
|
||||
static double
|
||||
resample_sinc_window (double x, double halfwidth, double scale)
|
||||
{
|
||||
double y;
|
||||
|
||||
if (x == 0)
|
||||
return 1.0;
|
||||
if (x < -halfwidth || x > halfwidth)
|
||||
return 0.0;
|
||||
|
||||
y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
|
||||
|
||||
x /= halfwidth;
|
||||
y *= (1 - x * x) * (1 - x * x);
|
||||
|
||||
return y;
|
||||
}
|
||||
|
||||
void
|
||||
resample_scale_chunk (ResampleState * r)
|
||||
{
|
||||
if (r->need_reinit) {
|
||||
RESAMPLE_DEBUG ("sample size %d", r->sample_size);
|
||||
|
||||
if (r->buffer)
|
||||
free (r->buffer);
|
||||
r->buffer_len = r->sample_size * 1000;
|
||||
r->buffer = malloc (r->buffer_len);
|
||||
memset (r->buffer, 0, r->buffer_len);
|
||||
|
||||
r->i_inc = r->o_rate / r->i_rate;
|
||||
r->o_inc = r->i_rate / r->o_rate;
|
||||
RESAMPLE_DEBUG ("i_inc %g o_inc %g", r->i_inc, r->o_inc);
|
||||
|
||||
r->i_start = -r->i_inc * r->filter_length;
|
||||
|
||||
r->need_reinit = 0;
|
||||
|
||||
#if 0
|
||||
if (r->i_inc < 1.0) {
|
||||
r->sinc_scale = r->i_inc;
|
||||
if (r->sinc_scale == 0.5) {
|
||||
/* strange things happen at integer multiples */
|
||||
r->sinc_scale = 1.0;
|
||||
}
|
||||
} else {
|
||||
r->sinc_scale = 1.0;
|
||||
}
|
||||
#else
|
||||
r->sinc_scale = 1.0;
|
||||
#endif
|
||||
}
|
||||
|
||||
while (r->o_size > 0) {
|
||||
double midpoint;
|
||||
int i;
|
||||
int j;
|
||||
|
||||
RESAMPLE_DEBUG ("i_start %g", r->i_start);
|
||||
midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
|
||||
if (midpoint > 0.5 * r->i_inc) {
|
||||
RESAMPLE_ERROR ("inconsistent state");
|
||||
}
|
||||
while (midpoint < -0.5 * r->i_inc) {
|
||||
AudioresampleBuffer *buffer;
|
||||
|
||||
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
|
||||
if (buffer == NULL) {
|
||||
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
|
||||
return;
|
||||
}
|
||||
|
||||
r->i_start += r->i_inc;
|
||||
RESAMPLE_DEBUG ("pulling (i_start = %g)", r->i_start);
|
||||
|
||||
midpoint += r->i_inc;
|
||||
memmove (r->buffer, r->buffer + r->sample_size,
|
||||
r->buffer_len - r->sample_size);
|
||||
|
||||
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
|
||||
r->sample_size);
|
||||
audioresample_buffer_unref (buffer);
|
||||
}
|
||||
|
||||
switch (r->format) {
|
||||
case RESAMPLE_FORMAT_S16:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(int16_t *) (r->buffer + i * sizeof (int16_t) +
|
||||
j * r->sample_size);
|
||||
acc +=
|
||||
resample_sinc_window (offset, r->filter_length * 0.5,
|
||||
r->sinc_scale) * x;
|
||||
}
|
||||
if (acc < -32768.0)
|
||||
acc = -32768.0;
|
||||
if (acc > 32767.0)
|
||||
acc = 32767.0;
|
||||
|
||||
*(int16_t *) (r->o_buf + i * sizeof (int16_t)) = rint (acc);
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_S32:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(int32_t *) (r->buffer + i * sizeof (int32_t) +
|
||||
j * r->sample_size);
|
||||
acc +=
|
||||
resample_sinc_window (offset, r->filter_length * 0.5,
|
||||
r->sinc_scale) * x;
|
||||
}
|
||||
if (acc < -2147483648.0)
|
||||
acc = -2147483648.0;
|
||||
if (acc > 2147483647.0)
|
||||
acc = 2147483647.0;
|
||||
|
||||
*(int32_t *) (r->o_buf + i * sizeof (int32_t)) = rint (acc);
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_F32:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(float *) (r->buffer + i * sizeof (float) +
|
||||
j * r->sample_size);
|
||||
acc +=
|
||||
resample_sinc_window (offset, r->filter_length * 0.5,
|
||||
r->sinc_scale) * x;
|
||||
}
|
||||
|
||||
*(float *) (r->o_buf + i * sizeof (float)) = acc;
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_F64:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(double *) (r->buffer + i * sizeof (double) +
|
||||
j * r->sample_size);
|
||||
acc +=
|
||||
resample_sinc_window (offset, r->filter_length * 0.5,
|
||||
r->sinc_scale) * x;
|
||||
}
|
||||
|
||||
*(double *) (r->o_buf + i * sizeof (double)) = acc;
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
r->i_start -= 1.0;
|
||||
r->o_buf += r->sample_size;
|
||||
r->o_size -= r->sample_size;
|
||||
}
|
||||
|
||||
}
|
||||
|
|
@ -1,271 +0,0 @@
|
|||
/* Resampling library
|
||||
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include <config.h>
|
||||
#endif
|
||||
|
||||
|
||||
#include <string.h>
|
||||
#include <math.h>
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <limits.h>
|
||||
#include <liboil/liboil.h>
|
||||
|
||||
#include "resample.h"
|
||||
#include "buffer.h"
|
||||
#include "debug.h"
|
||||
|
||||
static void
|
||||
func_sinc (double *fx, double *dfx, double x, void *closure)
|
||||
{
|
||||
//double scale = *(double *)closure;
|
||||
double scale = M_PI;
|
||||
|
||||
if (x == 0) {
|
||||
*fx = 1;
|
||||
*dfx = 0;
|
||||
return;
|
||||
}
|
||||
|
||||
x *= scale;
|
||||
*fx = sin (x) / x;
|
||||
*dfx = scale * (cos (x) - sin (x) / x) / x;
|
||||
}
|
||||
|
||||
static void
|
||||
func_hanning (double *fx, double *dfx, double x, void *closure)
|
||||
{
|
||||
double width = *(double *) closure;
|
||||
|
||||
if (x < width && x > -width) {
|
||||
x /= width;
|
||||
*fx = (1 - x * x) * (1 - x * x);
|
||||
*dfx = -2 * 2 * x / width * (1 - x * x);
|
||||
} else {
|
||||
*fx = 0;
|
||||
*dfx = 0;
|
||||
}
|
||||
}
|
||||
|
||||
#if 0
|
||||
static double
|
||||
resample_sinc_window (double x, double halfwidth, double scale)
|
||||
{
|
||||
double y;
|
||||
|
||||
if (x == 0)
|
||||
return 1.0;
|
||||
if (x < -halfwidth || x > halfwidth)
|
||||
return 0.0;
|
||||
|
||||
y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
|
||||
|
||||
x /= halfwidth;
|
||||
y *= (1 - x * x) * (1 - x * x);
|
||||
|
||||
return y;
|
||||
}
|
||||
#endif
|
||||
|
||||
#if 0
|
||||
static void
|
||||
functable_test (Functable * ft, double halfwidth)
|
||||
{
|
||||
int i;
|
||||
double x;
|
||||
|
||||
for (i = 0; i < 100; i++) {
|
||||
x = i * 0.1;
|
||||
printf ("%d %g %g\n", i, resample_sinc_window (x, halfwidth, 1.0),
|
||||
functable_evaluate (ft, x));
|
||||
}
|
||||
exit (0);
|
||||
|
||||
}
|
||||
#endif
|
||||
|
||||
|
||||
void
|
||||
resample_scale_functable (ResampleState * r)
|
||||
{
|
||||
if (r->need_reinit) {
|
||||
double hanning_width;
|
||||
|
||||
RESAMPLE_DEBUG ("sample size %d", r->sample_size);
|
||||
|
||||
if (r->buffer)
|
||||
free (r->buffer);
|
||||
r->buffer_len = r->sample_size * r->filter_length;
|
||||
r->buffer = malloc (r->buffer_len);
|
||||
memset (r->buffer, 0, r->buffer_len);
|
||||
|
||||
r->i_inc = r->o_rate / r->i_rate;
|
||||
r->o_inc = r->i_rate / r->o_rate;
|
||||
RESAMPLE_DEBUG ("i_inc %g o_inc %g", r->i_inc, r->o_inc);
|
||||
|
||||
r->i_start = -r->i_inc * r->filter_length;
|
||||
|
||||
if (r->ft) {
|
||||
functable_free (r->ft);
|
||||
}
|
||||
r->ft = functable_new ();
|
||||
functable_set_length (r->ft, r->filter_length * 16);
|
||||
functable_set_offset (r->ft, -r->filter_length / 2);
|
||||
functable_set_multiplier (r->ft, 1 / 16.0);
|
||||
|
||||
hanning_width = r->filter_length / 2;
|
||||
functable_calculate (r->ft, func_sinc, NULL);
|
||||
functable_calculate_multiply (r->ft, func_hanning, &hanning_width);
|
||||
|
||||
//functable_test(r->ft, 0.5 * r->filter_length);
|
||||
#if 0
|
||||
if (r->i_inc < 1.0) {
|
||||
r->sinc_scale = r->i_inc;
|
||||
if (r->sinc_scale == 0.5) {
|
||||
/* strange things happen at integer multiples */
|
||||
r->sinc_scale = 1.0;
|
||||
}
|
||||
} else {
|
||||
r->sinc_scale = 1.0;
|
||||
}
|
||||
#else
|
||||
r->sinc_scale = 1.0;
|
||||
#endif
|
||||
|
||||
r->need_reinit = 0;
|
||||
}
|
||||
|
||||
while (r->o_size > 0) {
|
||||
double midpoint;
|
||||
int i;
|
||||
int j;
|
||||
|
||||
RESAMPLE_DEBUG ("i_start %g", r->i_start);
|
||||
midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
|
||||
if (midpoint > 0.5 * r->i_inc) {
|
||||
RESAMPLE_ERROR ("inconsistent state");
|
||||
}
|
||||
while (midpoint < -0.5 * r->i_inc) {
|
||||
AudioresampleBuffer *buffer;
|
||||
|
||||
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
|
||||
if (buffer == NULL) {
|
||||
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
|
||||
return;
|
||||
}
|
||||
|
||||
r->i_start += r->i_inc;
|
||||
RESAMPLE_DEBUG ("pulling (i_start = %g)", r->i_start);
|
||||
|
||||
midpoint += r->i_inc;
|
||||
memmove (r->buffer, r->buffer + r->sample_size,
|
||||
r->buffer_len - r->sample_size);
|
||||
|
||||
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
|
||||
r->sample_size);
|
||||
audioresample_buffer_unref (buffer);
|
||||
}
|
||||
|
||||
switch (r->format) {
|
||||
case RESAMPLE_FORMAT_S16:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(int16_t *) (r->buffer + i * sizeof (int16_t) +
|
||||
j * r->sample_size);
|
||||
acc += functable_evaluate (r->ft, offset) * x;
|
||||
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
|
||||
}
|
||||
if (acc < -32768.0)
|
||||
acc = -32768.0;
|
||||
if (acc > 32767.0)
|
||||
acc = 32767.0;
|
||||
|
||||
*(int16_t *) (r->o_buf + i * sizeof (int16_t)) = rint (acc);
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_S32:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(int32_t *) (r->buffer + i * sizeof (int32_t) +
|
||||
j * r->sample_size);
|
||||
acc += functable_evaluate (r->ft, offset) * x;
|
||||
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
|
||||
}
|
||||
if (acc < -2147483648.0)
|
||||
acc = -2147483648.0;
|
||||
if (acc > 2147483647.0)
|
||||
acc = 2147483647.0;
|
||||
|
||||
*(int32_t *) (r->o_buf + i * sizeof (int32_t)) = rint (acc);
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_F32:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(float *) (r->buffer + i * sizeof (float) +
|
||||
j * r->sample_size);
|
||||
acc += functable_evaluate (r->ft, offset) * x;
|
||||
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
|
||||
}
|
||||
|
||||
*(float *) (r->o_buf + i * sizeof (float)) = acc;
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_F64:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(double *) (r->buffer + i * sizeof (double) +
|
||||
j * r->sample_size);
|
||||
acc += functable_evaluate (r->ft, offset) * x;
|
||||
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
|
||||
}
|
||||
|
||||
*(double *) (r->o_buf + i * sizeof (double)) = acc;
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
r->i_start -= 1.0;
|
||||
r->o_buf += r->sample_size;
|
||||
r->o_size -= r->sample_size;
|
||||
}
|
||||
|
||||
}
|
||||
|
|
@ -1,223 +0,0 @@
|
|||
/* Resampling library
|
||||
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include <config.h>
|
||||
#endif
|
||||
|
||||
|
||||
#include <string.h>
|
||||
#include <math.h>
|
||||
#include <stdio.h>
|
||||
#include <stdlib.h>
|
||||
#include <limits.h>
|
||||
#include <liboil/liboil.h>
|
||||
|
||||
#include "resample.h"
|
||||
#include "buffer.h"
|
||||
#include "debug.h"
|
||||
|
||||
|
||||
static double
|
||||
resample_sinc_window (double x, double halfwidth, double scale)
|
||||
{
|
||||
double y;
|
||||
|
||||
if (x == 0)
|
||||
return 1.0;
|
||||
if (x < -halfwidth || x > halfwidth)
|
||||
return 0.0;
|
||||
|
||||
y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
|
||||
|
||||
x /= halfwidth;
|
||||
y *= (1 - x * x) * (1 - x * x);
|
||||
|
||||
return y;
|
||||
}
|
||||
|
||||
void
|
||||
resample_scale_ref (ResampleState * r)
|
||||
{
|
||||
if (r->need_reinit) {
|
||||
RESAMPLE_DEBUG ("sample size %d", r->sample_size);
|
||||
|
||||
if (r->buffer)
|
||||
free (r->buffer);
|
||||
r->buffer_len = r->sample_size * r->filter_length;
|
||||
r->buffer = malloc (r->buffer_len);
|
||||
memset (r->buffer, 0, r->buffer_len);
|
||||
r->buffer_filled = 0;
|
||||
|
||||
r->i_inc = r->o_rate / r->i_rate;
|
||||
r->o_inc = r->i_rate / r->o_rate;
|
||||
RESAMPLE_DEBUG ("i_inc %g o_inc %g", r->i_inc, r->o_inc);
|
||||
|
||||
r->i_start = -r->i_inc * r->filter_length;
|
||||
|
||||
r->need_reinit = 0;
|
||||
|
||||
#if 0
|
||||
if (r->i_inc < 1.0) {
|
||||
r->sinc_scale = r->i_inc;
|
||||
if (r->sinc_scale == 0.5) {
|
||||
/* strange things happen at integer multiples */
|
||||
r->sinc_scale = 1.0;
|
||||
}
|
||||
} else {
|
||||
r->sinc_scale = 1.0;
|
||||
}
|
||||
#else
|
||||
r->sinc_scale = 1.0;
|
||||
#endif
|
||||
}
|
||||
|
||||
RESAMPLE_DEBUG ("asked to resample %d bytes", r->o_size);
|
||||
RESAMPLE_DEBUG ("%d bytes in queue",
|
||||
audioresample_buffer_queue_get_depth (r->queue));
|
||||
|
||||
while (r->o_size >= r->sample_size) {
|
||||
double midpoint;
|
||||
int i;
|
||||
int j;
|
||||
|
||||
midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
|
||||
RESAMPLE_DEBUG
|
||||
("still need to output %d bytes, %d input left, i_start %g, midpoint %f",
|
||||
r->o_size, audioresample_buffer_queue_get_depth (r->queue), r->i_start,
|
||||
midpoint);
|
||||
if (midpoint > 0.5 * r->i_inc) {
|
||||
RESAMPLE_ERROR ("inconsistent state");
|
||||
}
|
||||
while (midpoint < -0.5 * r->i_inc) {
|
||||
AudioresampleBuffer *buffer;
|
||||
|
||||
RESAMPLE_DEBUG ("midpoint %f < %f, r->i_inc %f", midpoint,
|
||||
-0.5 * r->i_inc, r->i_inc);
|
||||
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
|
||||
if (buffer == NULL) {
|
||||
/* FIXME: for the first buffer, this isn't necessarily an error,
|
||||
* since because of the filter length we'll output less buffers.
|
||||
* deal with that so we don't print to console */
|
||||
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
|
||||
return;
|
||||
}
|
||||
|
||||
r->i_start += r->i_inc;
|
||||
RESAMPLE_DEBUG ("pulling (i_start = %g)", r->i_start);
|
||||
|
||||
midpoint += r->i_inc;
|
||||
memmove (r->buffer, r->buffer + r->sample_size,
|
||||
r->buffer_len - r->sample_size);
|
||||
|
||||
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
|
||||
r->sample_size);
|
||||
r->buffer_filled = MIN (r->buffer_filled + r->sample_size, r->buffer_len);
|
||||
|
||||
audioresample_buffer_unref (buffer);
|
||||
}
|
||||
|
||||
switch (r->format) {
|
||||
case RESAMPLE_FORMAT_S16:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(int16_t *) (r->buffer + i * sizeof (int16_t) +
|
||||
j * r->sample_size);
|
||||
acc +=
|
||||
resample_sinc_window (offset, r->filter_length * 0.5,
|
||||
r->sinc_scale) * x;
|
||||
}
|
||||
if (acc < -32768.0)
|
||||
acc = -32768.0;
|
||||
if (acc > 32767.0)
|
||||
acc = 32767.0;
|
||||
|
||||
*(int16_t *) (r->o_buf + i * sizeof (int16_t)) = rint (acc);
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_S32:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(int32_t *) (r->buffer + i * sizeof (int32_t) +
|
||||
j * r->sample_size);
|
||||
acc +=
|
||||
resample_sinc_window (offset, r->filter_length * 0.5,
|
||||
r->sinc_scale) * x;
|
||||
}
|
||||
if (acc < -2147483648.0)
|
||||
acc = -2147483648.0;
|
||||
if (acc > 2147483647.0)
|
||||
acc = 2147483647.0;
|
||||
|
||||
*(int32_t *) (r->o_buf + i * sizeof (int32_t)) = rint (acc);
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_F32:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(float *) (r->buffer + i * sizeof (float) +
|
||||
j * r->sample_size);
|
||||
acc +=
|
||||
resample_sinc_window (offset, r->filter_length * 0.5,
|
||||
r->sinc_scale) * x;
|
||||
}
|
||||
|
||||
*(float *) (r->o_buf + i * sizeof (float)) = acc;
|
||||
}
|
||||
break;
|
||||
case RESAMPLE_FORMAT_F64:
|
||||
for (i = 0; i < r->n_channels; i++) {
|
||||
double acc = 0;
|
||||
double offset;
|
||||
double x;
|
||||
|
||||
for (j = 0; j < r->filter_length; j++) {
|
||||
offset = (r->i_start + j * r->i_inc) * r->o_inc;
|
||||
x = *(double *) (r->buffer + i * sizeof (double) +
|
||||
j * r->sample_size);
|
||||
acc +=
|
||||
resample_sinc_window (offset, r->filter_length * 0.5,
|
||||
r->sinc_scale) * x;
|
||||
}
|
||||
|
||||
*(double *) (r->o_buf + i * sizeof (double)) = acc;
|
||||
}
|
||||
break;
|
||||
}
|
||||
|
||||
r->i_start -= 1.0;
|
||||
r->o_buf += r->sample_size;
|
||||
r->o_size -= r->sample_size;
|
||||
}
|
||||
}
|
||||
|
|
@ -1,37 +0,0 @@
|
|||
plugin_LTLIBRARIES = libgstaudioresample.la
|
||||
|
||||
libgstaudioresample_la_SOURCES = \
|
||||
gstspeexresample.c \
|
||||
speex_resampler_int.c \
|
||||
speex_resampler_float.c \
|
||||
speex_resampler_double.c
|
||||
|
||||
libgstaudioresample_la_CFLAGS = \
|
||||
$(GST_PLUGINS_BASE_CFLAGS) \
|
||||
$(GST_BASE_CFLAGS) \
|
||||
$(GST_CFLAGS) \
|
||||
$(LIBOIL_CFLAGS)
|
||||
|
||||
libgstaudioresample_la_LIBADD = \
|
||||
$(GST_PLUGINS_BASE_LIBS) \
|
||||
$(GST_BASE_LIBS) \
|
||||
$(GST_LIBS) \
|
||||
$(LIBOIL_LIBS) \
|
||||
$(LIBM)
|
||||
|
||||
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
|
||||
libgstaudioresample_la_LIBTOOLFLAGS = --tag=disable-static
|
||||
|
||||
noinst_HEADERS = \
|
||||
arch.h \
|
||||
fixed_arm4.h \
|
||||
fixed_arm5e.h \
|
||||
fixed_bfin.h \
|
||||
fixed_debug.h \
|
||||
fixed_generic.h \
|
||||
gstspeexresample.h \
|
||||
resample.c \
|
||||
resample_sse.h \
|
||||
speex_resampler.h \
|
||||
speex_resampler_wrapper.h
|
||||
|
||||
File diff suppressed because it is too large
Load diff
|
|
@ -1,90 +0,0 @@
|
|||
/* GStreamer
|
||||
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
|
||||
* Copyright (C) <2007-2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or (at your option) any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
|
||||
#ifndef __SPEEX_RESAMPLE_H__
|
||||
#define __SPEEX_RESAMPLE_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstbasetransform.h>
|
||||
#include <gst/audio/audio.h>
|
||||
|
||||
#include "speex_resampler_wrapper.h"
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
#define GST_TYPE_SPEEX_RESAMPLE \
|
||||
(gst_speex_resample_get_type())
|
||||
#define GST_SPEEX_RESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResample))
|
||||
#define GST_SPEEX_RESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResampleClass))
|
||||
#define GST_IS_SPEEX_RESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SPEEX_RESAMPLE))
|
||||
#define GST_IS_SPEEX_RESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SPEEX_RESAMPLE))
|
||||
|
||||
typedef struct _GstSpeexResample GstSpeexResample;
|
||||
typedef struct _GstSpeexResampleClass GstSpeexResampleClass;
|
||||
|
||||
/**
|
||||
* GstSpeexResample:
|
||||
*
|
||||
* Opaque data structure.
|
||||
*/
|
||||
struct _GstSpeexResample {
|
||||
GstBaseTransform element;
|
||||
|
||||
/* <private> */
|
||||
|
||||
GstCaps *srccaps, *sinkcaps;
|
||||
|
||||
gboolean need_discont;
|
||||
|
||||
guint64 next_offset;
|
||||
GstClockTime next_ts;
|
||||
GstClockTime next_upstream_ts;
|
||||
|
||||
gint channels;
|
||||
gint inrate;
|
||||
gint outrate;
|
||||
gint quality;
|
||||
gint width;
|
||||
gboolean fp;
|
||||
|
||||
guint8 *tmp_in;
|
||||
guint tmp_in_size;
|
||||
|
||||
guint8 *tmp_out;
|
||||
guint tmp_out_size;
|
||||
|
||||
SpeexResamplerState *state;
|
||||
const SpeexResampleFuncs *funcs;
|
||||
};
|
||||
|
||||
struct _GstSpeexResampleClass {
|
||||
GstBaseTransformClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_speex_resample_get_type(void);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
#endif /* __SPEEX_RESAMPLE_H__ */
|
||||
File diff suppressed because it is too large
Load diff
|
|
@ -412,6 +412,7 @@ mid_type_find (GstTypeFind * tf, gpointer unused)
|
|||
&& data[3] == 'd')
|
||||
gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MID_CAPS);
|
||||
}
|
||||
|
||||
/*** audio/mobile-xmf ***/
|
||||
|
||||
static GstStaticCaps mxmf_caps = GST_STATIC_CAPS ("audio/mobile-xmf");
|
||||
|
|
@ -432,7 +433,7 @@ mxmf_type_find (GstTypeFind * tf, gpointer unused)
|
|||
&& data[3] == '0') {
|
||||
/* Search TypeId 2 1 byte */
|
||||
data = gst_type_find_peek (tf, 11, 1);
|
||||
if (data && data[0] == 2 ) {
|
||||
if (data && data[0] == 2) {
|
||||
gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MXMF_CAPS);
|
||||
}
|
||||
}
|
||||
|
|
@ -3208,8 +3209,8 @@ plugin_init (GstPlugin * plugin)
|
|||
#endif
|
||||
TYPE_FIND_REGISTER (plugin, "audio/midi", GST_RANK_PRIMARY, mid_type_find,
|
||||
mid_exts, MID_CAPS, NULL, NULL);
|
||||
TYPE_FIND_REGISTER (plugin, "audio/mobile-xmf", GST_RANK_PRIMARY, mxmf_type_find,
|
||||
mxmf_exts, MXMF_CAPS, NULL, NULL);
|
||||
TYPE_FIND_REGISTER (plugin, "audio/mobile-xmf", GST_RANK_PRIMARY,
|
||||
mxmf_type_find, mxmf_exts, MXMF_CAPS, NULL, NULL);
|
||||
TYPE_FIND_REGISTER (plugin, "video/x-fli", GST_RANK_MARGINAL, flx_type_find,
|
||||
flx_exts, FLX_CAPS, NULL, NULL);
|
||||
TYPE_FIND_REGISTER (plugin, "application/x-id3v2", GST_RANK_PRIMARY + 103,
|
||||
|
|
@ -3241,8 +3242,8 @@ plugin_init (GstPlugin * plugin)
|
|||
mpeg4_video_type_find, m4v_exts, MPEG_VIDEO_CAPS, NULL, NULL);
|
||||
TYPE_FIND_REGISTER (plugin, "video/x-h264", GST_RANK_PRIMARY,
|
||||
h264_video_type_find, h264_exts, MPEG_VIDEO_CAPS, NULL, NULL);
|
||||
TYPE_FIND_REGISTER (plugin, "video/x-nuv", GST_RANK_SECONDARY,
|
||||
nuv_type_find, nuv_exts, NUV_CAPS, NULL, NULL);
|
||||
TYPE_FIND_REGISTER (plugin, "video/x-nuv", GST_RANK_SECONDARY, nuv_type_find,
|
||||
nuv_exts, NUV_CAPS, NULL, NULL);
|
||||
|
||||
/* ISO formats */
|
||||
TYPE_FIND_REGISTER (plugin, "audio/x-m4a", GST_RANK_PRIMARY, m4a_type_find,
|
||||
|
|
|
|||
|
|
@ -91,6 +91,7 @@ check_PROGRAMS = \
|
|||
elements/adder \
|
||||
elements/audioconvert \
|
||||
elements/audiorate \
|
||||
elements/audioresample \
|
||||
elements/audiotestsrc \
|
||||
elements/decodebin \
|
||||
elements/ffmpegcolorspace \
|
||||
|
|
@ -99,7 +100,6 @@ check_PROGRAMS = \
|
|||
elements/multifdsink \
|
||||
elements/playbin \
|
||||
$(check_subparse) \
|
||||
elements/speexresample \
|
||||
elements/videorate \
|
||||
elements/videotestsrc \
|
||||
elements/volume \
|
||||
|
|
@ -241,12 +241,12 @@ elements_decodebin_CFLAGS = $(GST_BASE_CFLAGS) $(AM_CFLAGS)
|
|||
elements_subparse_LDADD = $(LDADD)
|
||||
elements_subparse_CFLAGS = $(CFLAGS) $(AM_CFLAGS)
|
||||
|
||||
elements_speexresample_CFLAGS = \
|
||||
elements_audioresample_CFLAGS = \
|
||||
$(GST_PLUGINS_BASE_CFLAGS) \
|
||||
$(GST_BASE_CFLAGS) \
|
||||
$(AM_CFLAGS)
|
||||
|
||||
elements_speexresample_LDADD = \
|
||||
elements_audioresample_LDADD = \
|
||||
$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
|
||||
$(top_builddir)/gst-libs/gst/interfaces/libgstinterfaces-@GST_MAJORMINOR@.la \
|
||||
$(GST_BASE_LIBS) \
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
/* GStreamer
|
||||
*
|
||||
* unit test for audioresample
|
||||
* unit test for audioresample, based on the audioresample unit test
|
||||
*
|
||||
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
|
||||
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
|
||||
|
|
@ -25,13 +25,21 @@
|
|||
|
||||
#include <gst/check/gstcheck.h>
|
||||
|
||||
#include <gst/audio/audio.h>
|
||||
|
||||
/* For ease of programming we use globals to keep refs for our floating
|
||||
* src and sink pads we create; otherwise we always have to do get_pad,
|
||||
* get_peer, and then remove references in every test function */
|
||||
static GstPad *mysrcpad, *mysinkpad;
|
||||
|
||||
#define RESAMPLE_CAPS_FLOAT \
|
||||
"audio/x-raw-float, " \
|
||||
"channels = (int) [ 1, MAX ], " \
|
||||
"rate = (int) [ 1, MAX ], " \
|
||||
"endianness = (int) BYTE_ORDER, " \
|
||||
"width = (int) { 32, 64 }"
|
||||
|
||||
#define RESAMPLE_CAPS_TEMPLATE_STRING \
|
||||
#define RESAMPLE_CAPS_INT \
|
||||
"audio/x-raw-int, " \
|
||||
"channels = (int) [ 1, MAX ], " \
|
||||
"rate = (int) [ 1, MAX ], " \
|
||||
|
|
@ -40,6 +48,10 @@ static GstPad *mysrcpad, *mysinkpad;
|
|||
"depth = (int) 16, " \
|
||||
"signed = (bool) TRUE"
|
||||
|
||||
#define RESAMPLE_CAPS_TEMPLATE_STRING \
|
||||
RESAMPLE_CAPS_FLOAT " ; " \
|
||||
RESAMPLE_CAPS_INT
|
||||
|
||||
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
|
||||
GST_PAD_SINK,
|
||||
GST_PAD_ALWAYS,
|
||||
|
|
@ -52,7 +64,8 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
|
|||
);
|
||||
|
||||
static GstElement *
|
||||
setup_audioresample (int channels, int inrate, int outrate)
|
||||
setup_audioresample (int channels, int inrate, int outrate, int width,
|
||||
gboolean fp)
|
||||
{
|
||||
GstElement *audioresample;
|
||||
GstCaps *caps;
|
||||
|
|
@ -61,10 +74,15 @@ setup_audioresample (int channels, int inrate, int outrate)
|
|||
GST_DEBUG ("setup_audioresample");
|
||||
audioresample = gst_check_setup_element ("audioresample");
|
||||
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
|
||||
if (fp)
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
|
||||
else
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
|
||||
"rate", G_TYPE_INT, inrate, NULL);
|
||||
"rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
|
||||
if (!fp)
|
||||
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
fail_unless (gst_element_set_state (audioresample,
|
||||
|
|
@ -75,10 +93,15 @@ setup_audioresample (int channels, int inrate, int outrate)
|
|||
gst_pad_set_caps (mysrcpad, caps);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
|
||||
if (fp)
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
|
||||
else
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
|
||||
"rate", G_TYPE_INT, outrate, NULL);
|
||||
"rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
|
||||
if (!fp)
|
||||
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
|
||||
|
|
@ -90,6 +113,8 @@ setup_audioresample (int channels, int inrate, int outrate)
|
|||
gst_pad_set_active (mysinkpad, TRUE);
|
||||
gst_pad_set_active (mysrcpad, TRUE);
|
||||
|
||||
gst_caps_unref (caps);
|
||||
|
||||
return audioresample;
|
||||
}
|
||||
|
||||
|
|
@ -153,7 +178,7 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
|
|||
int i, j;
|
||||
gint16 *p;
|
||||
|
||||
audioresample = setup_audioresample (2, inrate, outrate);
|
||||
audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
|
||||
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
|
|
@ -164,7 +189,7 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
|
|||
for (j = 1; j <= numbuffers; ++j) {
|
||||
|
||||
inbuffer = gst_buffer_new_and_alloc (samples * 4);
|
||||
GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
|
||||
GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
|
||||
GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
|
||||
GST_BUFFER_OFFSET (inbuffer) = offset;
|
||||
offset += samples;
|
||||
|
|
@ -240,7 +265,7 @@ test_discont_stream_instance (int inrate, int outrate, int samples,
|
|||
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
|
||||
inrate, outrate, samples, numbuffers);
|
||||
|
||||
audioresample = setup_audioresample (2, inrate, outrate);
|
||||
audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
|
||||
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
|
|
@ -326,7 +351,7 @@ GST_START_TEST (test_reuse)
|
|||
GstBuffer *inbuffer;
|
||||
GstCaps *caps;
|
||||
|
||||
audioresample = setup_audioresample (1, 9343, 48000);
|
||||
audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE);
|
||||
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
|
|
@ -477,6 +502,7 @@ live_switch_push (int rate, GstCaps * caps)
|
|||
|
||||
desired = gst_caps_copy (caps);
|
||||
gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
|
||||
gst_pad_set_caps (mysrcpad, desired);
|
||||
|
||||
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
|
||||
GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
|
||||
|
|
@ -485,6 +511,8 @@ live_switch_push (int rate, GstCaps * caps)
|
|||
* returns a buffer with exactly the same caps as we requested so the actual
|
||||
* renegotiation (if needed) will be done in the _chain*/
|
||||
fail_unless (inbuffer != NULL);
|
||||
GST_DEBUG ("desired: %" GST_PTR_FORMAT ".... got: %" GST_PTR_FORMAT,
|
||||
desired, GST_BUFFER_CAPS (inbuffer));
|
||||
fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer)));
|
||||
|
||||
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
|
||||
|
|
@ -516,7 +544,7 @@ GST_START_TEST (test_live_switch)
|
|||
GstEvent *newseg;
|
||||
GstCaps *caps;
|
||||
|
||||
audioresample = setup_audioresample (4, 48000, 48000);
|
||||
audioresample = setup_audioresample (4, 48000, 48000, 16, FALSE);
|
||||
|
||||
/* Let the sinkpad act like something that can only handle things of
|
||||
* rate 48000- and can only allocate buffers for that rate, but if someone
|
||||
|
|
@ -525,6 +553,8 @@ GST_START_TEST (test_live_switch)
|
|||
gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
|
||||
gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
|
||||
|
||||
gst_pad_use_fixed_caps (mysrcpad);
|
||||
|
||||
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
|
|
@ -549,7 +579,104 @@ GST_START_TEST (test_live_switch)
|
|||
gst_caps_unref (caps);
|
||||
}
|
||||
|
||||
GST_END_TEST static Suite *
|
||||
GST_END_TEST;
|
||||
|
||||
#ifndef GST_DISABLE_PARSE
|
||||
|
||||
static GMainLoop *loop;
|
||||
static gint messages = 0;
|
||||
|
||||
static void
|
||||
element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
|
||||
{
|
||||
gchar *s;
|
||||
|
||||
s = gst_structure_to_string (gst_message_get_structure (message));
|
||||
GST_DEBUG ("Received message: %s", s);
|
||||
g_free (s);
|
||||
|
||||
messages++;
|
||||
}
|
||||
|
||||
static void
|
||||
eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
|
||||
{
|
||||
GST_DEBUG ("Received eos");
|
||||
g_main_loop_quit (loop);
|
||||
}
|
||||
|
||||
static void
|
||||
test_pipeline (gint width, gboolean fp, gint inrate, gint outrate, gint quality)
|
||||
{
|
||||
GstElement *pipeline;
|
||||
GstBus *bus;
|
||||
GError *error = NULL;
|
||||
gchar *pipe_str;
|
||||
|
||||
pipe_str =
|
||||
g_strdup_printf
|
||||
("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw-%s,rate=%d,width=%d,channels=2 ! audioresample quality=%d ! audio/x-raw-%s,rate=%d,width=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
|
||||
(fp) ? "float" : "int", inrate, width, quality, (fp) ? "float" : "int",
|
||||
outrate, width);
|
||||
|
||||
pipeline = gst_parse_launch (pipe_str, &error);
|
||||
fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
|
||||
error ? error->message : "(invalid error)");
|
||||
g_free (pipe_str);
|
||||
|
||||
bus = gst_element_get_bus (pipeline);
|
||||
fail_if (bus == NULL);
|
||||
gst_bus_add_signal_watch (bus);
|
||||
g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
|
||||
NULL);
|
||||
g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
|
||||
|
||||
gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
||||
|
||||
/* run until we receive EOS */
|
||||
loop = g_main_loop_new (NULL, FALSE);
|
||||
|
||||
g_main_loop_run (loop);
|
||||
|
||||
g_main_loop_unref (loop);
|
||||
loop = NULL;
|
||||
|
||||
gst_element_set_state (pipeline, GST_STATE_NULL);
|
||||
|
||||
fail_if (messages > 0, "Received imperfect timestamp messages");
|
||||
gst_object_unref (pipeline);
|
||||
}
|
||||
|
||||
GST_START_TEST (test_pipelines)
|
||||
{
|
||||
gint quality;
|
||||
|
||||
/* Test qualities 0, 5 and 10 */
|
||||
for (quality = 0; quality < 11; quality += 5) {
|
||||
test_pipeline (8, FALSE, 44100, 48000, quality);
|
||||
test_pipeline (8, FALSE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (16, FALSE, 44100, 48000, quality);
|
||||
test_pipeline (16, FALSE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (24, FALSE, 44100, 48000, quality);
|
||||
test_pipeline (24, FALSE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (32, FALSE, 44100, 48000, quality);
|
||||
test_pipeline (32, FALSE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (32, TRUE, 44100, 48000, quality);
|
||||
test_pipeline (32, TRUE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (64, TRUE, 44100, 48000, quality);
|
||||
test_pipeline (64, TRUE, 48000, 44100, quality);
|
||||
}
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
#endif
|
||||
|
||||
static Suite *
|
||||
audioresample_suite (void)
|
||||
{
|
||||
Suite *s = suite_create ("audioresample");
|
||||
|
|
@ -562,6 +689,11 @@ audioresample_suite (void)
|
|||
tcase_add_test (tc_chain, test_shutdown);
|
||||
tcase_add_test (tc_chain, test_live_switch);
|
||||
|
||||
#ifndef GST_DISABLE_PARSE
|
||||
tcase_set_timeout (tc_chain, 360);
|
||||
tcase_add_test (tc_chain, test_pipelines);
|
||||
#endif
|
||||
|
||||
return s;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -1,700 +0,0 @@
|
|||
/* GStreamer
|
||||
*
|
||||
* unit test for speexresample, based on the audioresample unit test
|
||||
*
|
||||
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
|
||||
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or (at your option) any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#include <unistd.h>
|
||||
|
||||
#include <gst/check/gstcheck.h>
|
||||
|
||||
#include <gst/audio/audio.h>
|
||||
|
||||
/* For ease of programming we use globals to keep refs for our floating
|
||||
* src and sink pads we create; otherwise we always have to do get_pad,
|
||||
* get_peer, and then remove references in every test function */
|
||||
static GstPad *mysrcpad, *mysinkpad;
|
||||
|
||||
#define RESAMPLE_CAPS_FLOAT \
|
||||
"audio/x-raw-float, " \
|
||||
"channels = (int) [ 1, MAX ], " \
|
||||
"rate = (int) [ 1, MAX ], " \
|
||||
"endianness = (int) BYTE_ORDER, " \
|
||||
"width = (int) { 32, 64 }"
|
||||
|
||||
#define RESAMPLE_CAPS_INT \
|
||||
"audio/x-raw-int, " \
|
||||
"channels = (int) [ 1, MAX ], " \
|
||||
"rate = (int) [ 1, MAX ], " \
|
||||
"endianness = (int) BYTE_ORDER, " \
|
||||
"width = (int) 16, " \
|
||||
"depth = (int) 16, " \
|
||||
"signed = (bool) TRUE"
|
||||
|
||||
#define RESAMPLE_CAPS_TEMPLATE_STRING \
|
||||
RESAMPLE_CAPS_FLOAT " ; " \
|
||||
RESAMPLE_CAPS_INT
|
||||
|
||||
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
|
||||
GST_PAD_SINK,
|
||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
|
||||
);
|
||||
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
|
||||
GST_PAD_SRC,
|
||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
|
||||
);
|
||||
|
||||
static GstElement *
|
||||
setup_speexresample (int channels, int inrate, int outrate, int width,
|
||||
gboolean fp)
|
||||
{
|
||||
GstElement *speexresample;
|
||||
GstCaps *caps;
|
||||
GstStructure *structure;
|
||||
|
||||
GST_DEBUG ("setup_speexresample");
|
||||
speexresample = gst_check_setup_element ("audioresample");
|
||||
|
||||
if (fp)
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
|
||||
else
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
|
||||
"rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
|
||||
if (!fp)
|
||||
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
fail_unless (gst_element_set_state (speexresample,
|
||||
GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to paused");
|
||||
|
||||
mysrcpad = gst_check_setup_src_pad (speexresample, &srctemplate, caps);
|
||||
gst_pad_set_caps (mysrcpad, caps);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
if (fp)
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
|
||||
else
|
||||
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
|
||||
"rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
|
||||
if (!fp)
|
||||
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
mysinkpad = gst_check_setup_sink_pad (speexresample, &sinktemplate, caps);
|
||||
/* this installs a getcaps func that will always return the caps we set
|
||||
* later */
|
||||
gst_pad_set_caps (mysinkpad, caps);
|
||||
gst_pad_use_fixed_caps (mysinkpad);
|
||||
|
||||
gst_pad_set_active (mysinkpad, TRUE);
|
||||
gst_pad_set_active (mysrcpad, TRUE);
|
||||
|
||||
gst_caps_unref (caps);
|
||||
|
||||
return speexresample;
|
||||
}
|
||||
|
||||
static void
|
||||
cleanup_speexresample (GstElement * speexresample)
|
||||
{
|
||||
GST_DEBUG ("cleanup_speexresample");
|
||||
|
||||
fail_unless (gst_element_set_state (speexresample,
|
||||
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
|
||||
|
||||
gst_pad_set_active (mysrcpad, FALSE);
|
||||
gst_pad_set_active (mysinkpad, FALSE);
|
||||
gst_check_teardown_src_pad (speexresample);
|
||||
gst_check_teardown_sink_pad (speexresample);
|
||||
gst_check_teardown_element (speexresample);
|
||||
}
|
||||
|
||||
static void
|
||||
fail_unless_perfect_stream (void)
|
||||
{
|
||||
guint64 timestamp = 0L, duration = 0L;
|
||||
guint64 offset = 0L, offset_end = 0L;
|
||||
|
||||
GList *l;
|
||||
GstBuffer *buffer;
|
||||
|
||||
for (l = buffers; l; l = l->next) {
|
||||
buffer = GST_BUFFER (l->data);
|
||||
ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
|
||||
GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
|
||||
G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
|
||||
G_GUINT64_FORMAT,
|
||||
GST_BUFFER_TIMESTAMP (buffer),
|
||||
GST_BUFFER_DURATION (buffer),
|
||||
GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
|
||||
|
||||
fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
|
||||
fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
|
||||
duration = GST_BUFFER_DURATION (buffer);
|
||||
offset_end = GST_BUFFER_OFFSET_END (buffer);
|
||||
|
||||
timestamp += duration;
|
||||
offset = offset_end;
|
||||
gst_buffer_unref (buffer);
|
||||
}
|
||||
g_list_free (buffers);
|
||||
buffers = NULL;
|
||||
}
|
||||
|
||||
/* this tests that the output is a perfect stream if the input is */
|
||||
static void
|
||||
test_perfect_stream_instance (int inrate, int outrate, int samples,
|
||||
int numbuffers)
|
||||
{
|
||||
GstElement *speexresample;
|
||||
GstBuffer *inbuffer, *outbuffer;
|
||||
GstCaps *caps;
|
||||
guint64 offset = 0;
|
||||
|
||||
int i, j;
|
||||
gint16 *p;
|
||||
|
||||
speexresample = setup_speexresample (2, inrate, outrate, 16, FALSE);
|
||||
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
fail_unless (gst_element_set_state (speexresample,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
for (j = 1; j <= numbuffers; ++j) {
|
||||
|
||||
inbuffer = gst_buffer_new_and_alloc (samples * 4);
|
||||
GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
|
||||
GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
|
||||
GST_BUFFER_OFFSET (inbuffer) = offset;
|
||||
offset += samples;
|
||||
GST_BUFFER_OFFSET_END (inbuffer) = offset;
|
||||
|
||||
gst_buffer_set_caps (inbuffer, caps);
|
||||
|
||||
p = (gint16 *) GST_BUFFER_DATA (inbuffer);
|
||||
|
||||
/* create a 16 bit signed ramp */
|
||||
for (i = 0; i < samples; ++i) {
|
||||
*p = -32767 + i * (65535 / samples);
|
||||
++p;
|
||||
*p = -32767 + i * (65535 / samples);
|
||||
++p;
|
||||
}
|
||||
|
||||
/* pushing gives away my reference ... */
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
/* ... but it ends up being collected on the global buffer list */
|
||||
fail_unless_equals_int (g_list_length (buffers), j);
|
||||
}
|
||||
|
||||
/* FIXME: we should make speexresample handle eos by flushing out the last
|
||||
* samples, which will give us one more, small, buffer */
|
||||
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
|
||||
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
|
||||
|
||||
fail_unless_perfect_stream ();
|
||||
|
||||
/* cleanup */
|
||||
gst_caps_unref (caps);
|
||||
cleanup_speexresample (speexresample);
|
||||
}
|
||||
|
||||
|
||||
/* make sure that outgoing buffers are contiguous in timestamp/duration and
|
||||
* offset/offsetend
|
||||
*/
|
||||
GST_START_TEST (test_perfect_stream)
|
||||
{
|
||||
/* integral scalings */
|
||||
test_perfect_stream_instance (48000, 24000, 500, 20);
|
||||
test_perfect_stream_instance (48000, 12000, 500, 20);
|
||||
test_perfect_stream_instance (12000, 24000, 500, 20);
|
||||
test_perfect_stream_instance (12000, 48000, 500, 20);
|
||||
|
||||
/* non-integral scalings */
|
||||
test_perfect_stream_instance (44100, 8000, 500, 20);
|
||||
test_perfect_stream_instance (8000, 44100, 500, 20);
|
||||
|
||||
/* wacky scalings */
|
||||
test_perfect_stream_instance (12345, 54321, 500, 20);
|
||||
test_perfect_stream_instance (101, 99, 500, 20);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
/* this tests that the output is a correct discontinuous stream
|
||||
* if the input is; ie input drops in time come out the same way */
|
||||
static void
|
||||
test_discont_stream_instance (int inrate, int outrate, int samples,
|
||||
int numbuffers)
|
||||
{
|
||||
GstElement *speexresample;
|
||||
GstBuffer *inbuffer, *outbuffer;
|
||||
GstCaps *caps;
|
||||
GstClockTime ints;
|
||||
|
||||
int i, j;
|
||||
gint16 *p;
|
||||
|
||||
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
|
||||
inrate, outrate, samples, numbuffers);
|
||||
|
||||
speexresample = setup_speexresample (2, inrate, outrate, 16, FALSE);
|
||||
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
fail_unless (gst_element_set_state (speexresample,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
for (j = 1; j <= numbuffers; ++j) {
|
||||
|
||||
inbuffer = gst_buffer_new_and_alloc (samples * 4);
|
||||
GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
|
||||
/* "drop" half the buffers */
|
||||
ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
|
||||
GST_BUFFER_TIMESTAMP (inbuffer) = ints;
|
||||
GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
|
||||
GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
|
||||
|
||||
gst_buffer_set_caps (inbuffer, caps);
|
||||
|
||||
p = (gint16 *) GST_BUFFER_DATA (inbuffer);
|
||||
|
||||
/* create a 16 bit signed ramp */
|
||||
for (i = 0; i < samples; ++i) {
|
||||
*p = -32767 + i * (65535 / samples);
|
||||
++p;
|
||||
*p = -32767 + i * (65535 / samples);
|
||||
++p;
|
||||
}
|
||||
|
||||
GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
|
||||
G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
|
||||
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
|
||||
GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
|
||||
GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
|
||||
/* pushing gives away my reference ... */
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
|
||||
/* check if the timestamp of the pushed buffer matches the incoming one */
|
||||
outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
|
||||
fail_if (outbuffer == NULL);
|
||||
fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
|
||||
GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
|
||||
G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
|
||||
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
|
||||
GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
|
||||
GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
|
||||
if (j > 1) {
|
||||
fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
|
||||
"expected discont for buffer #%d", j);
|
||||
}
|
||||
}
|
||||
|
||||
/* cleanup */
|
||||
gst_caps_unref (caps);
|
||||
cleanup_speexresample (speexresample);
|
||||
}
|
||||
|
||||
GST_START_TEST (test_discont_stream)
|
||||
{
|
||||
/* integral scalings */
|
||||
test_discont_stream_instance (48000, 24000, 500, 20);
|
||||
test_discont_stream_instance (48000, 12000, 500, 20);
|
||||
test_discont_stream_instance (12000, 24000, 500, 20);
|
||||
test_discont_stream_instance (12000, 48000, 500, 20);
|
||||
|
||||
/* non-integral scalings */
|
||||
test_discont_stream_instance (44100, 8000, 500, 20);
|
||||
test_discont_stream_instance (8000, 44100, 500, 20);
|
||||
|
||||
/* wacky scalings */
|
||||
test_discont_stream_instance (12345, 54321, 500, 20);
|
||||
test_discont_stream_instance (101, 99, 500, 20);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
|
||||
|
||||
GST_START_TEST (test_reuse)
|
||||
{
|
||||
GstElement *speexresample;
|
||||
GstEvent *newseg;
|
||||
GstBuffer *inbuffer;
|
||||
GstCaps *caps;
|
||||
|
||||
speexresample = setup_speexresample (1, 9343, 48000, 16, FALSE);
|
||||
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
fail_unless (gst_element_set_state (speexresample,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
|
||||
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
|
||||
|
||||
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
|
||||
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
|
||||
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
|
||||
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
|
||||
GST_BUFFER_OFFSET (inbuffer) = 0;
|
||||
gst_buffer_set_caps (inbuffer, caps);
|
||||
|
||||
/* pushing gives away my reference ... */
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
|
||||
/* ... but it ends up being collected on the global buffer list */
|
||||
fail_unless_equals_int (g_list_length (buffers), 1);
|
||||
|
||||
/* now reset and try again ... */
|
||||
fail_unless (gst_element_set_state (speexresample,
|
||||
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
|
||||
|
||||
fail_unless (gst_element_set_state (speexresample,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
|
||||
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
|
||||
|
||||
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
|
||||
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
|
||||
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
|
||||
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
|
||||
GST_BUFFER_OFFSET (inbuffer) = 0;
|
||||
gst_buffer_set_caps (inbuffer, caps);
|
||||
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
|
||||
/* ... it also ends up being collected on the global buffer list. If we
|
||||
* now have more than 2 buffers, then speexresample probably didn't clean
|
||||
* up its internal buffer properly and tried to push the remaining samples
|
||||
* when it got the second NEWSEGMENT event */
|
||||
fail_unless_equals_int (g_list_length (buffers), 2);
|
||||
|
||||
cleanup_speexresample (speexresample);
|
||||
gst_caps_unref (caps);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
GST_START_TEST (test_shutdown)
|
||||
{
|
||||
GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink;
|
||||
GstCaps *caps;
|
||||
guint i;
|
||||
|
||||
/* create pipeline, force speexresample to actually resample */
|
||||
pipeline = gst_pipeline_new (NULL);
|
||||
|
||||
src = gst_check_setup_element ("audiotestsrc");
|
||||
cf1 = gst_check_setup_element ("capsfilter");
|
||||
ar = gst_check_setup_element ("audioresample");
|
||||
cf2 = gst_check_setup_element ("capsfilter");
|
||||
g_object_set (cf2, "name", "capsfilter2", NULL);
|
||||
sink = gst_check_setup_element ("fakesink");
|
||||
|
||||
caps =
|
||||
gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 11025, NULL);
|
||||
g_object_set (cf1, "caps", caps, NULL);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
caps =
|
||||
gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 48000, NULL);
|
||||
g_object_set (cf2, "caps", caps, NULL);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
/* don't want to sync against the clock, the more throughput the better */
|
||||
g_object_set (src, "is-live", FALSE, NULL);
|
||||
g_object_set (sink, "sync", FALSE, NULL);
|
||||
|
||||
gst_bin_add_many (GST_BIN (pipeline), src, cf1, ar, cf2, sink, NULL);
|
||||
fail_if (!gst_element_link_many (src, cf1, ar, cf2, sink, NULL));
|
||||
|
||||
/* now, wait until pipeline is running and then shut it down again; repeat */
|
||||
for (i = 0; i < 20; ++i) {
|
||||
gst_element_set_state (pipeline, GST_STATE_PAUSED);
|
||||
gst_element_get_state (pipeline, NULL, NULL, -1);
|
||||
gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
||||
g_usleep (100);
|
||||
gst_element_set_state (pipeline, GST_STATE_NULL);
|
||||
}
|
||||
|
||||
gst_object_unref (pipeline);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
static GstFlowReturn
|
||||
live_switch_alloc_only_48000 (GstPad * pad, guint64 offset,
|
||||
guint size, GstCaps * caps, GstBuffer ** buf)
|
||||
{
|
||||
GstStructure *structure;
|
||||
gint rate;
|
||||
gint channels;
|
||||
GstCaps *desired;
|
||||
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
fail_unless (gst_structure_get_int (structure, "rate", &rate));
|
||||
fail_unless (gst_structure_get_int (structure, "channels", &channels));
|
||||
|
||||
if (rate < 48000)
|
||||
return GST_FLOW_NOT_NEGOTIATED;
|
||||
|
||||
desired = gst_caps_copy (caps);
|
||||
gst_caps_set_simple (desired, "rate", G_TYPE_INT, 48000, NULL);
|
||||
|
||||
*buf = gst_buffer_new_and_alloc (channels * 48000);
|
||||
gst_buffer_set_caps (*buf, desired);
|
||||
gst_caps_unref (desired);
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
static GstCaps *
|
||||
live_switch_get_sink_caps (GstPad * pad)
|
||||
{
|
||||
GstCaps *result;
|
||||
|
||||
result = gst_caps_copy (GST_PAD_CAPS (pad));
|
||||
|
||||
gst_caps_set_simple (result,
|
||||
"rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
static void
|
||||
live_switch_push (int rate, GstCaps * caps)
|
||||
{
|
||||
GstBuffer *inbuffer;
|
||||
GstCaps *desired;
|
||||
GList *l;
|
||||
|
||||
desired = gst_caps_copy (caps);
|
||||
gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
|
||||
gst_pad_set_caps (mysrcpad, desired);
|
||||
|
||||
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
|
||||
GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
|
||||
|
||||
/* When the basetransform hits the non-configured case it always
|
||||
* returns a buffer with exactly the same caps as we requested so the actual
|
||||
* renegotiation (if needed) will be done in the _chain*/
|
||||
fail_unless (inbuffer != NULL);
|
||||
GST_DEBUG ("desired: %" GST_PTR_FORMAT ".... got: %" GST_PTR_FORMAT,
|
||||
desired, GST_BUFFER_CAPS (inbuffer));
|
||||
fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer)));
|
||||
|
||||
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
|
||||
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
|
||||
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
|
||||
GST_BUFFER_OFFSET (inbuffer) = 0;
|
||||
|
||||
/* pushing gives away my reference ... */
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
|
||||
/* ... but it ends up being collected on the global buffer list */
|
||||
fail_unless_equals_int (g_list_length (buffers), 1);
|
||||
|
||||
for (l = buffers; l; l = l->next) {
|
||||
GstBuffer *buffer = GST_BUFFER (l->data);
|
||||
|
||||
gst_buffer_unref (buffer);
|
||||
}
|
||||
|
||||
g_list_free (buffers);
|
||||
buffers = NULL;
|
||||
|
||||
gst_caps_unref (desired);
|
||||
}
|
||||
|
||||
GST_START_TEST (test_live_switch)
|
||||
{
|
||||
GstElement *speexresample;
|
||||
GstEvent *newseg;
|
||||
GstCaps *caps;
|
||||
|
||||
speexresample = setup_speexresample (4, 48000, 48000, 16, FALSE);
|
||||
|
||||
/* Let the sinkpad act like something that can only handle things of
|
||||
* rate 48000- and can only allocate buffers for that rate, but if someone
|
||||
* tries to get a buffer with a rate higher then 48000 tries to renegotiate
|
||||
* */
|
||||
gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
|
||||
gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
|
||||
|
||||
gst_pad_use_fixed_caps (mysrcpad);
|
||||
|
||||
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
||||
fail_unless (gst_caps_is_fixed (caps));
|
||||
|
||||
fail_unless (gst_element_set_state (speexresample,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
|
||||
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
|
||||
|
||||
/* downstream can provide the requested rate, a buffer alloc will be passed
|
||||
* on */
|
||||
live_switch_push (48000, caps);
|
||||
|
||||
/* Downstream can never accept this rate, buffer alloc isn't passed on */
|
||||
live_switch_push (40000, caps);
|
||||
|
||||
/* Downstream can provide the requested rate but will re-negotiate */
|
||||
live_switch_push (50000, caps);
|
||||
|
||||
cleanup_speexresample (speexresample);
|
||||
gst_caps_unref (caps);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
#ifndef GST_DISABLE_PARSE
|
||||
|
||||
static GMainLoop *loop;
|
||||
static gint messages = 0;
|
||||
|
||||
static void
|
||||
element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
|
||||
{
|
||||
gchar *s;
|
||||
|
||||
s = gst_structure_to_string (gst_message_get_structure (message));
|
||||
GST_DEBUG ("Received message: %s", s);
|
||||
g_free (s);
|
||||
|
||||
messages++;
|
||||
}
|
||||
|
||||
static void
|
||||
eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
|
||||
{
|
||||
GST_DEBUG ("Received eos");
|
||||
g_main_loop_quit (loop);
|
||||
}
|
||||
|
||||
static void
|
||||
test_pipeline (gint width, gboolean fp, gint inrate, gint outrate, gint quality)
|
||||
{
|
||||
GstElement *pipeline;
|
||||
GstBus *bus;
|
||||
GError *error = NULL;
|
||||
gchar *pipe_str;
|
||||
|
||||
pipe_str =
|
||||
g_strdup_printf
|
||||
("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw-%s,rate=%d,width=%d,channels=2 ! audioresample quality=%d ! audio/x-raw-%s,rate=%d,width=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
|
||||
(fp) ? "float" : "int", inrate, width, quality, (fp) ? "float" : "int",
|
||||
outrate, width);
|
||||
|
||||
pipeline = gst_parse_launch (pipe_str, &error);
|
||||
fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
|
||||
error ? error->message : "(invalid error)");
|
||||
g_free (pipe_str);
|
||||
|
||||
bus = gst_element_get_bus (pipeline);
|
||||
fail_if (bus == NULL);
|
||||
gst_bus_add_signal_watch (bus);
|
||||
g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
|
||||
NULL);
|
||||
g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
|
||||
|
||||
gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
||||
|
||||
/* run until we receive EOS */
|
||||
loop = g_main_loop_new (NULL, FALSE);
|
||||
|
||||
g_main_loop_run (loop);
|
||||
|
||||
g_main_loop_unref (loop);
|
||||
loop = NULL;
|
||||
|
||||
gst_element_set_state (pipeline, GST_STATE_NULL);
|
||||
|
||||
fail_if (messages > 0, "Received imperfect timestamp messages");
|
||||
gst_object_unref (pipeline);
|
||||
}
|
||||
|
||||
GST_START_TEST (test_pipelines)
|
||||
{
|
||||
gint quality;
|
||||
|
||||
/* Test qualities 0, 5 and 10 */
|
||||
for (quality = 0; quality < 11; quality += 5) {
|
||||
test_pipeline (8, FALSE, 44100, 48000, quality);
|
||||
test_pipeline (8, FALSE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (16, FALSE, 44100, 48000, quality);
|
||||
test_pipeline (16, FALSE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (24, FALSE, 44100, 48000, quality);
|
||||
test_pipeline (24, FALSE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (32, FALSE, 44100, 48000, quality);
|
||||
test_pipeline (32, FALSE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (32, TRUE, 44100, 48000, quality);
|
||||
test_pipeline (32, TRUE, 48000, 44100, quality);
|
||||
|
||||
test_pipeline (64, TRUE, 44100, 48000, quality);
|
||||
test_pipeline (64, TRUE, 48000, 44100, quality);
|
||||
}
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
#endif
|
||||
|
||||
static Suite *
|
||||
speexresample_suite (void)
|
||||
{
|
||||
Suite *s = suite_create ("speexresample");
|
||||
TCase *tc_chain = tcase_create ("general");
|
||||
|
||||
suite_add_tcase (s, tc_chain);
|
||||
tcase_add_test (tc_chain, test_perfect_stream);
|
||||
tcase_add_test (tc_chain, test_discont_stream);
|
||||
tcase_add_test (tc_chain, test_reuse);
|
||||
tcase_add_test (tc_chain, test_shutdown);
|
||||
tcase_add_test (tc_chain, test_live_switch);
|
||||
|
||||
#ifndef GST_DISABLE_PARSE
|
||||
tcase_set_timeout (tc_chain, 360);
|
||||
tcase_add_test (tc_chain, test_pipelines);
|
||||
#endif
|
||||
|
||||
return s;
|
||||
}
|
||||
|
||||
GST_CHECK_MAIN (speexresample);
|
||||
Loading…
Reference in a new issue