From e204c5934c35e5af1be7afb70477f10a7b0cc2ac Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 6 Sep 2011 13:16:27 +0200 Subject: [PATCH] -good: port to new audio caps --- ext/jack/gstjack.h | 4 +-- ext/pulse/pulsesink.c | 8 ++--- ext/pulse/pulsesrc.c | 6 ++-- ext/pulse/pulseutil.c | 20 +++++------ gst/audiofx/audiopanorama.c | 85 +++++++++++++++------------------------------ gst/audiofx/audiopanorama.h | 8 ++--- gst/auparse/gstauparse.c | 26 +++++++------- gst/avi/gstavimux.c | 55 ++++++++++++----------------- gst/isomp4/gstqtmux.c | 2 +- gst/isomp4/qtdemux.c | 14 ++++---- gst/law/alaw.c | 4 +-- gst/law/mulaw-decode.c | 4 +-- gst/law/mulaw.c | 4 +-- gst/spectrum/gstspectrum.c | 6 ++-- gst/wavparse/gstwavparse.c | 2 +- 15 files changed, 104 insertions(+), 144 deletions(-) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 9724cc0..2b5dbe8 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -50,9 +50,9 @@ typedef jack_default_audio_sample_t sample_t; #define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ()) #if G_BYTE_ORDER == G_LITTLE_ENDIAN -#define GST_JACK_FORMAT_STR "F32_LE" +#define GST_JACK_FORMAT_STR "F32LE" #else -#define GST_JACK_FORMAT_STR "F32_BE" +#define GST_JACK_FORMAT_STR "F32BE" #endif GType gst_jack_client_get_type(void); diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index dd89f99..b4c8a54 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -1717,11 +1717,11 @@ static GstStateChangeReturn gst_pulsesink_change_state (GstElement * element, GstStateChange transition); #if (G_BYTE_ORDER == G_LITTLE_ENDIAN) -# define FORMATS "{ S16_LE, S16_BE, F32_LE, F32_BE, S32_LE, S32_BE, " \ - "S24_3LE, S24_3BE, S24_LE, S24_BE, S8 }" +# define FORMATS "{ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, " \ + "S24LE, S24BE, S24_32LE, S24_32BE, S8 }" #else -# define FORMATS "{ S16_BE, S16_LE, F32_BE, F32_LE, S32_BE, S32_LE, " \ - "S24_3BE, S24_3LE, S24_BE, S24_LE, S8 }" +# define FORMATS "{ S16BE, S16LE, F32BE, F32LE, S32BE, S32LE, " \ + "S24BE, S24LE, S24_32BE, S24_32LE, S8 }" #endif static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink", diff --git a/ext/pulse/pulsesrc.c b/ext/pulse/pulsesrc.c index e0c1134..112d41b 100644 --- a/ext/pulse/pulsesrc.c +++ b/ext/pulse/pulsesrc.c @@ -99,9 +99,9 @@ static GstStateChangeReturn gst_pulsesrc_change_state (GstElement * element, GstStateChange transition); #if (G_BYTE_ORDER == G_LITTLE_ENDIAN) -# define FORMATS "{ S16_LE, S16_BE, F32_LE, F32_BE, S32_LE, S32_BE, U8 }" +# define FORMATS "{ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, U8 }" #else -# define FORMATS "{ S16_BE, S16_LE, F32_BE, F32_LE, S32_BE, S32_LE, U8 }" +# define FORMATS "{ S16BE, S16LE, F32BE, F32LE, S32BE, S32LE, U8 }" #endif static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("src", @@ -1084,7 +1084,7 @@ gst_pulsesrc_success_cb (pa_stream * s, int success, void *userdata) { GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata); - pulsesrc->operation_success = ! !success; + pulsesrc->operation_success = !!success; pa_threaded_mainloop_signal (pulsesrc->mainloop, 0); } diff --git a/ext/pulse/pulseutil.c b/ext/pulse/pulseutil.c index 196e39c..171ef4c 100644 --- a/ext/pulse/pulseutil.c +++ b/ext/pulse/pulseutil.c @@ -98,34 +98,34 @@ gst_pulse_fill_sample_spec (GstRingBufferSpec * spec, pa_sample_spec * ss) case GST_AUDIO_FORMAT_U8: ss->format = PA_SAMPLE_U8; break; - case GST_AUDIO_FORMAT_S16_LE: + case GST_AUDIO_FORMAT_S16LE: ss->format = PA_SAMPLE_S16LE; break; - case GST_AUDIO_FORMAT_S16_BE: + case GST_AUDIO_FORMAT_S16BE: ss->format = PA_SAMPLE_S16BE; break; - case GST_AUDIO_FORMAT_F32_LE: + case GST_AUDIO_FORMAT_F32LE: ss->format = PA_SAMPLE_FLOAT32LE; break; - case GST_AUDIO_FORMAT_F32_BE: + case GST_AUDIO_FORMAT_F32BE: ss->format = PA_SAMPLE_FLOAT32BE; break; - case GST_AUDIO_FORMAT_S32_LE: + case GST_AUDIO_FORMAT_S32LE: ss->format = PA_SAMPLE_S32LE; break; - case GST_AUDIO_FORMAT_S32_BE: + case GST_AUDIO_FORMAT_S32BE: ss->format = PA_SAMPLE_S32BE; break; - case GST_AUDIO_FORMAT_S24_3LE: + case GST_AUDIO_FORMAT_S24LE: ss->format = PA_SAMPLE_S24LE; break; - case GST_AUDIO_FORMAT_S24_3BE: + case GST_AUDIO_FORMAT_S24BE: ss->format = PA_SAMPLE_S24BE; break; - case GST_AUDIO_FORMAT_S24_LE: + case GST_AUDIO_FORMAT_S24_32LE: ss->format = PA_SAMPLE_S24_32LE; break; - case GST_AUDIO_FORMAT_S24_BE: + case GST_AUDIO_FORMAT_S24_32BE: ss->format = PA_SAMPLE_S24_32BE; break; default: diff --git a/gst/audiofx/audiopanorama.c b/gst/audiofx/audiopanorama.c index aa3bc84..9c43f60 100644 --- a/gst/audiofx/audiopanorama.c +++ b/gst/audiofx/audiopanorama.c @@ -92,29 +92,17 @@ gst_audio_panorama_method_get_type (void) static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, 2 ], " - "endianness = (int) BYTE_ORDER, " "width = (int) 32; " - "audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, 2 ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true") + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) { " GST_AUDIO_NE (S32) ", " GST_AUDIO_NE (S16) "}, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]") ); static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) 2, " - "endianness = (int) BYTE_ORDER, " "width = (int) 32; " - "audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) 2, " - "endianness = (int) BYTE_ORDER, " - "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true") + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) { " GST_AUDIO_NE (S32) ", " GST_AUDIO_NE (S16) "}, " + "rate = (int) [ 1, MAX ], " "channels = (int) 2") ); G_DEFINE_TYPE (GstAudioPanorama, gst_audio_panorama, GST_TYPE_BASE_TRANSFORM); @@ -237,27 +225,27 @@ gst_audio_panorama_init (GstAudioPanorama * filter) filter->panorama = 0; filter->method = METHOD_PSYCHOACOUSTIC; - filter->width = 0; - filter->channels = 0; - filter->format_float = FALSE; + gst_audio_info_init (&filter->info); filter->process = NULL; gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE); } static gboolean -gst_audio_panorama_set_process_function (GstAudioPanorama * filter) +gst_audio_panorama_set_process_function (GstAudioPanorama * filter, + GstAudioInfo * info) { gint channel_index, format_index, method_index; + const GstAudioFormatInfo *finfo = info->finfo; /* set processing function */ - channel_index = filter->channels - 1; + channel_index = GST_AUDIO_INFO_CHANNELS (info) - 1; if (channel_index > 1 || channel_index < 0) { filter->process = NULL; return FALSE; } - format_index = (filter->format_float) ? 1 : 0; + format_index = GST_AUDIO_FORMAT_INFO_IS_FLOAT (finfo) ? 1 : 0; method_index = filter->method; if (method_index >= NUM_METHODS || method_index < 0) @@ -280,7 +268,7 @@ gst_audio_panorama_set_property (GObject * object, guint prop_id, break; case PROP_METHOD: filter->method = g_value_get_enum (value); - gst_audio_panorama_set_process_function (filter); + gst_audio_panorama_set_process_function (filter, &filter->info); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -356,44 +344,27 @@ gst_audio_panorama_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps) { GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base); - const GstStructure *structure; - gboolean ret; - gint width; - const gchar *fmt; + GstAudioInfo info; /*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */ + if (!gst_audio_info_from_caps (&info, incaps)) + goto no_format; - structure = gst_caps_get_structure (incaps, 0); - ret = gst_structure_get_int (structure, "channels", &filter->channels); - if (!ret) - goto no_channels; - - ret = gst_structure_get_int (structure, "width", &width); - if (!ret) - goto no_width; - filter->width = width / 8; - - fmt = gst_structure_get_name (structure); - if (!strcmp (fmt, "audio/x-raw-int")) - filter->format_float = FALSE; - else - filter->format_float = TRUE; - - GST_DEBUG ("try to process %s input with %d channels", fmt, filter->channels); + GST_DEBUG ("try to process %d input with %d channels", + GST_AUDIO_INFO_FORMAT (&info), GST_AUDIO_INFO_CHANNELS (&info)); - ret = gst_audio_panorama_set_process_function (filter); + if (!gst_audio_panorama_set_process_function (filter, &info)) + goto no_format; - if (!ret) - GST_WARNING ("can't process input with %d channels", filter->channels); + filter->info = info; - return ret; + return TRUE; -no_channels: - GST_DEBUG ("no channels in caps"); - return ret; -no_width: - GST_DEBUG ("no width in caps"); - return ret; +no_format: + { + GST_DEBUG ("invalid caps"); + return FALSE; + } } /* psychoacoustic processing functions */ @@ -672,7 +643,7 @@ gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf, GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP); memset (outdata, 0, outsize); } else { - guint num_samples = outsize / (2 * filter->width); + guint num_samples = outsize / GST_AUDIO_INFO_BPF (&filter->info); filter->process (filter, indata, outdata, num_samples); } diff --git a/gst/audiofx/audiopanorama.h b/gst/audiofx/audiopanorama.h index b3cd183..e445f4d 100644 --- a/gst/audiofx/audiopanorama.h +++ b/gst/audiofx/audiopanorama.h @@ -22,6 +22,7 @@ #define __GST_AUDIO_PANORAMA_H__ #include +#include #include G_BEGIN_DECLS @@ -42,12 +43,11 @@ struct _GstAudioPanorama { GstBaseTransform element; gfloat panorama; - + /* < private > */ GstAudioPanoramaProcessFunc process; - gint channels; - gboolean format_float; - gint width; + + GstAudioInfo info; gint method; }; diff --git a/gst/auparse/gstauparse.c b/gst/auparse/gstauparse.c index 054a640..bf7b639 100644 --- a/gst/auparse/gstauparse.c +++ b/gst/auparse/gstauparse.c @@ -45,9 +45,9 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", #define GST_AU_PARSE_RAW_PAD_TEMPLATE_CAPS \ "audio/x-raw, " \ - "format= (string) { S8, S16_LE, S16_BE, S24_3LE, S24_3BE, " \ - "S32_LE, S32_BE, F32_LE, F32_BE, " \ - "F64_LE, F64_BE }, " \ + "format= (string) { S8, S16LE, S16BE, S24LE, S24BE, " \ + "S32LE, S32BE, F32LE, F32BE, " \ + "F64LE, F64BE }, " \ "rate = (int) [ 8000, 192000 ], " \ "channels = (int) [ 1, 2 ]" @@ -257,38 +257,38 @@ gst_au_parse_parse_header (GstAuParse * auparse) break; case 3: /* 16-bit linear PCM */ if (endianness == G_LITTLE_ENDIAN) - format = GST_AUDIO_FORMAT_S16_LE; + format = GST_AUDIO_FORMAT_S16LE; else - format = GST_AUDIO_FORMAT_S16_BE; + format = GST_AUDIO_FORMAT_S16BE; auparse->sample_size = auparse->channels * 2; break; case 4: /* 24-bit linear PCM */ if (endianness == G_LITTLE_ENDIAN) - format = GST_AUDIO_FORMAT_S24_3LE; + format = GST_AUDIO_FORMAT_S24LE; else - format = GST_AUDIO_FORMAT_S24_3BE; + format = GST_AUDIO_FORMAT_S24BE; auparse->sample_size = auparse->channels * 3; break; case 5: /* 32-bit linear PCM */ if (endianness == G_LITTLE_ENDIAN) - format = GST_AUDIO_FORMAT_S32_LE; + format = GST_AUDIO_FORMAT_S32LE; else - format = GST_AUDIO_FORMAT_S32_BE; + format = GST_AUDIO_FORMAT_S32BE; auparse->sample_size = auparse->channels * 4; break; case 6: /* 32-bit IEEE floating point */ if (endianness == G_LITTLE_ENDIAN) - format = GST_AUDIO_FORMAT_F32_LE; + format = GST_AUDIO_FORMAT_F32LE; else - format = GST_AUDIO_FORMAT_F32_BE; + format = GST_AUDIO_FORMAT_F32BE; auparse->sample_size = auparse->channels * 4; break; case 7: /* 64-bit IEEE floating point */ if (endianness == G_LITTLE_ENDIAN) - format = GST_AUDIO_FORMAT_F64_LE; + format = GST_AUDIO_FORMAT_F64LE; else - format = GST_AUDIO_FORMAT_F64_BE; + format = GST_AUDIO_FORMAT_F64BE; auparse->sample_size = auparse->channels * 8; break; diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index 514257e..13e0fff 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -38,7 +38,7 @@ * ! 'video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1' \ * ! queue ! mux. \ * audiotestsrc num-buffers=440 ! audioconvert \ - * ! 'audio/x-raw-int,rate=44100,channels=2' ! queue ! mux. \ + * ! 'audio/x-raw,rate=44100,channels=2' ! queue ! mux. \ * avimux name=mux ! filesink location=test.avi * ]| This will create an .AVI file containing an uncompressed video stream * with a test picture and an uncompressed audio stream containing a @@ -47,7 +47,7 @@ * gst-launch videotestsrc num-buffers=250 \ * ! 'video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1' \ * ! xvidenc ! queue ! mux. \ - * audiotestsrc num-buffers=440 ! audioconvert ! 'audio/x-raw-int,rate=44100,channels=2' \ + * audiotestsrc num-buffers=440 ! audioconvert ! 'audio/x-raw,rate=44100,channels=2' \ * ! lame ! queue ! mux. \ * avimux name=mux ! filesink location=test.avi * ]| This will create an .AVI file containing the same test video and sound @@ -66,6 +66,7 @@ #include #include +#include #include #include "gstavimux.h" @@ -160,11 +161,8 @@ static GstStaticPadTemplate audio_sink_factory = GST_STATIC_PAD_TEMPLATE ("audio_%d", GST_PAD_SINK, GST_PAD_REQUEST, - GST_STATIC_CAPS ("audio/x-raw-int, " - "endianness = (int) LITTLE_ENDIAN, " - "signed = (boolean) { TRUE, FALSE }, " - "width = (int) { 8, 16 }, " - "depth = (int) { 8, 16 }, " + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) { U8, S16LE }, " "rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 2 ]; " "audio/mpeg, " @@ -758,36 +756,27 @@ gst_avi_mux_audsink_set_caps (GstPad * pad, GstCaps * vscaps) avimux->codec_data_size += gst_buffer_get_size (avipad->auds_codec_data); } - if (!strcmp (mimetype, "audio/x-raw-int")) { - gint width, depth; - gboolean signedness; - - avipad->auds.format = GST_RIFF_WAVE_FORMAT_PCM; - - if (!gst_structure_get_int (structure, "width", &width) || - !gst_structure_get_int (structure, "depth", &depth) || - !gst_structure_get_boolean (structure, "signed", &signedness)) { - GST_DEBUG_OBJECT (avimux, - "broken caps, width/depth/signed field missing"); - goto refuse_caps; - } + if (!strcmp (mimetype, "audio/x-raw")) { + const gchar *format; + GstAudioFormat fmt; - /* no clear place to put different values for these while keeping to spec */ - if (width != depth) { - GST_DEBUG_OBJECT (avimux, "width must be same as depth!"); - goto refuse_caps; - } + format = gst_structure_get_string (structure, "format"); + fmt = gst_audio_format_from_string (format); - /* because that's the way the caps will be recreated from riff data */ - if ((width == 8 && signedness) || (width == 16 && !signedness)) { - GST_DEBUG_OBJECT (avimux, - "8-bit PCM must be unsigned, 16-bit PCM signed"); - goto refuse_caps; + switch (fmt) { + case GST_AUDIO_FORMAT_U8: + avipad->auds.blockalign = 8; + avipad->auds.size = 8; + break; + case GST_AUDIO_FORMAT_S16: + avipad->auds.blockalign = 16; + avipad->auds.size = 16; + break; + default: + goto refuse_caps; } - avipad->auds.blockalign = width; - avipad->auds.size = (width == 8) ? 8 : depth; - + avipad->auds.format = GST_RIFF_WAVE_FORMAT_PCM; /* set some more info straight */ avipad->auds.blockalign /= 8; avipad->auds.blockalign *= avipad->auds.channels; diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index 9dc9fba..3e210f7 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -97,7 +97,7 @@ * * Example pipelines * |[ - * gst-launch v4l2src num-buffers=500 ! video/x-raw-yuv,width=320,height=240 ! ffmpegcolorspace ! qtmux ! filesink location=video.mov + * gst-launch v4l2src num-buffers=500 ! video/x-raw,width=320,height=240 ! ffmpegcolorspace ! qtmux ! filesink location=video.mov * ]| * Records a video stream captured from a v4l2 device and muxes it into a qt file. * diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c index 057d03d..d81c2dc 100644 --- a/gst/isomp4/qtdemux.c +++ b/gst/isomp4/qtdemux.c @@ -5256,7 +5256,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* sync sample atom */ stream->stps_present = FALSE; if ((stream->stss_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, &stream->stss) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stss.data = g_memdup (stream->stss.data, stream->stss.size); @@ -5274,7 +5274,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* partial sync sample atom */ if ((stream->stps_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, &stream->stps) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stps.data = g_memdup (stream->stps.data, stream->stps.size); @@ -5393,7 +5393,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* composition time-to-sample */ if ((stream->ctts_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, &stream->ctts) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->ctts.data = g_memdup (stream->ctts.data, stream->ctts.size); @@ -9382,24 +9382,24 @@ qtdemux_audio_caps (GstQTDemux * qtdemux, QtDemuxStream * stream, case GST_MAKE_FOURCC ('f', 'l', '6', '4'): _codec ("Raw 64-bit floating-point audio"); caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, "F64_BE", NULL); + "format", G_TYPE_STRING, "F64BE", NULL); break; case GST_MAKE_FOURCC ('f', 'l', '3', '2'): _codec ("Raw 32-bit floating-point audio"); caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, "F32_BE", NULL); + "format", G_TYPE_STRING, "F32BE", NULL); break; case FOURCC_in24: _codec ("Raw 24-bit PCM audio"); /* we assume BIG ENDIAN, an enda box will tell us to change this to little * endian later */ caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, "S24_3BE", NULL); + "format", G_TYPE_STRING, "S24BE", NULL); break; case GST_MAKE_FOURCC ('i', 'n', '3', '2'): _codec ("Raw 32-bit PCM audio"); caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, "S32_BE", NULL); + "format", G_TYPE_STRING, "S32BE", NULL); break; case GST_MAKE_FOURCC ('u', 'l', 'a', 'w'): _codec ("Mu-law audio"); diff --git a/gst/law/alaw.c b/gst/law/alaw.c index 1e7cc78..9e34205 100644 --- a/gst/law/alaw.c +++ b/gst/law/alaw.c @@ -24,9 +24,9 @@ #include "alaw-decode.h" #if G_BYTE_ORDER == G_LITTLE_ENDIAN -#define INT_FORMAT "S16_LE" +#define INT_FORMAT "S16LE" #else -#define INT_FORMAT "S16_BE" +#define INT_FORMAT "S16BE" #endif GstStaticPadTemplate alaw_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src", diff --git a/gst/law/mulaw-decode.c b/gst/law/mulaw-decode.c index 15bbcd8..e9bbdf2 100644 --- a/gst/law/mulaw-decode.c +++ b/gst/law/mulaw-decode.c @@ -31,9 +31,9 @@ #include "mulaw-conversion.h" #if G_BYTE_ORDER == G_LITTLE_ENDIAN -#define INT_FORMAT "S16_LE" +#define INT_FORMAT "S16LE" #else -#define INT_FORMAT "S16_BE" +#define INT_FORMAT "S16BE" #endif extern GstStaticPadTemplate mulaw_dec_src_factory; diff --git a/gst/law/mulaw.c b/gst/law/mulaw.c index 1784084..ba310db 100644 --- a/gst/law/mulaw.c +++ b/gst/law/mulaw.c @@ -23,9 +23,9 @@ #include "mulaw-decode.h" #if G_BYTE_ORDER == G_LITTLE_ENDIAN -#define INT_FORMAT "S16_LE" +#define INT_FORMAT "S16LE" #else -#define INT_FORMAT "S16_BE" +#define INT_FORMAT "S16BE" #endif GstStaticPadTemplate mulaw_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src", diff --git a/gst/spectrum/gstspectrum.c b/gst/spectrum/gstspectrum.c index fbbdf36..f2d8193 100644 --- a/gst/spectrum/gstspectrum.c +++ b/gst/spectrum/gstspectrum.c @@ -111,9 +111,9 @@ GST_DEBUG_CATEGORY_STATIC (gst_spectrum_debug); /* elementfactory information */ #if G_BYTE_ORDER == G_LITTLE_ENDIAN -# define FORMATS "{ S16_LE, S24_3LE, S32_LE, F32_LE, F64_LE }" +# define FORMATS "{ S16LE, S24LE, S32LE, F32LE, F64LE }" #else -# define FORMATS "{ S16_BE, S24_3BE, S32_BE, F32_BE, F64_BE }" +# define FORMATS "{ S16BE, S24BE, S32BE, F32BE, F64BE }" #endif #define ALLOWED_CAPS \ @@ -626,7 +626,7 @@ gst_spectrum_setup (GstAudioFilter * base, const GstAudioInfo * info) input_data = multi_channel ? input_data_int16_max : input_data_mixed_int16_max; break; - case GST_AUDIO_FORMAT_S24_3: + case GST_AUDIO_FORMAT_S24: input_data = multi_channel ? input_data_int24_max : input_data_mixed_int24_max; break; diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 3377eae..b04da8d 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -1743,7 +1743,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf) if (wav->caps) { s = gst_caps_get_structure (wav->caps, 0); - if (s && gst_structure_has_name (s, "audio/x-raw-int") && buf != NULL) { + if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) { GstTypeFindProbability prob; GstCaps *tf_caps; -- 2.7.4