From: Wim Taymans Date: Wed, 17 Aug 2005 19:05:51 +0000 (+0000) Subject: configure.ac: Added mpegaudioparse X-Git-Tag: 1.19.3~505^2~2245 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=c3326a7da2be36f393b72d2731fd1e0fdbe13160;p=platform%2Fupstream%2Fgstreamer.git configure.ac: Added mpegaudioparse Original commit message from CVS: * configure.ac: Added mpegaudioparse * ext/lame/gstlame.c: (gst_lame_src_getcaps), (gst_lame_src_setcaps), (gst_lame_sink_setcaps), (gst_lame_sink_event), (gst_lame_chain): Some cleanups. Fix memleak. * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_class_init), (gst_mp3parse_init), (gst_mp3parse_chain), (gst_mp3parse_change_state): * gst/mpegaudioparse/gstmpegaudioparse.h: Ported mpegaudioparse --- diff --git a/ChangeLog b/ChangeLog index 481020e..65a0bac 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,5 +1,22 @@ 2005-08-17 Wim Taymans + * configure.ac: + Added mpegaudioparse + + * ext/lame/gstlame.c: (gst_lame_src_getcaps), + (gst_lame_src_setcaps), (gst_lame_sink_setcaps), + (gst_lame_sink_event), (gst_lame_chain): + Some cleanups. + Fix memleak. + + * gst/mpegaudioparse/gstmpegaudioparse.c: + (gst_mp3parse_class_init), (gst_mp3parse_init), + (gst_mp3parse_chain), (gst_mp3parse_change_state): + * gst/mpegaudioparse/gstmpegaudioparse.h: + Ported mpegaudioparse + +2005-08-17 Wim Taymans + * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open), (gst_rtspsrc_play): Support absolute control urls too. diff --git a/configure.ac b/configure.ac index 45ce24f..9ef0688 100644 --- a/configure.ac +++ b/configure.ac @@ -633,6 +633,7 @@ gst/fdsrc/Makefile gst/goom/Makefile gst/law/Makefile gst/level/Makefile +gst/mpegaudioparse/Makefile gst/realmedia/Makefile gst/rtp/Makefile gst/rtsp/Makefile diff --git a/ext/lame/gstlame.c b/ext/lame/gstlame.c index d2cfd4b..5519d60 100644 --- a/ext/lame/gstlame.c +++ b/ext/lame/gstlame.c @@ -1020,11 +1020,11 @@ static GstFlowReturn gst_lame_chain (GstPad * pad, GstBuffer * buf) { GstLame *lame; - GstBuffer *outbuf; - guchar *mp3_data = NULL; - gint mp3_buffer_size, mp3_size = 0; + guchar *mp3_data; + gint mp3_buffer_size, mp3_size; gint64 duration; GstFlowReturn result; + gint num_samples; lame = GST_LAME (gst_pad_get_parent (pad)); @@ -1033,9 +1033,10 @@ gst_lame_chain (GstPad * pad, GstBuffer * buf) if (!lame->initialized) goto not_initialized; + num_samples = GST_BUFFER_SIZE (buf) / 2; + /* allocate space for output */ - mp3_buffer_size = - ((GST_BUFFER_SIZE (buf) / (2 + lame->num_channels)) * 1.25) + 7200; + mp3_buffer_size = 1.25 * num_samples + 7200; mp3_data = g_malloc (mp3_buffer_size); /* lame seems to be too stupid to get mono interleaved going */ @@ -1043,12 +1044,11 @@ gst_lame_chain (GstPad * pad, GstBuffer * buf) mp3_size = lame_encode_buffer (lame->lgf, (short int *) (GST_BUFFER_DATA (buf)), (short int *) (GST_BUFFER_DATA (buf)), - GST_BUFFER_SIZE (buf) / 2, mp3_data, mp3_buffer_size); + num_samples, mp3_data, mp3_buffer_size); } else { mp3_size = lame_encode_buffer_interleaved (lame->lgf, (short int *) (GST_BUFFER_DATA (buf)), - GST_BUFFER_SIZE (buf) / 2 / lame->num_channels, - mp3_data, mp3_buffer_size); + num_samples / lame->num_channels, mp3_data, mp3_buffer_size); } GST_LOG_OBJECT (lame, "encoded %d bytes of audio to %d bytes of mp3", @@ -1074,9 +1074,16 @@ gst_lame_chain (GstPad * pad, GstBuffer * buf) gst_buffer_unref (buf); + if (mp3_size < 0) { + g_warning ("error %d", mp3_size); + } + if (mp3_size > 0) { + GstBuffer *outbuf; + outbuf = gst_buffer_new (); GST_BUFFER_DATA (outbuf) = mp3_data; + GST_BUFFER_MALLOCDATA (outbuf) = mp3_data; GST_BUFFER_SIZE (outbuf) = mp3_size; GST_BUFFER_TIMESTAMP (outbuf) = lame->last_ts; GST_BUFFER_OFFSET (outbuf) = lame->last_offs; diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c index de128f5..90775ea 100644 --- a/gst/mpegaudioparse/gstmpegaudioparse.c +++ b/gst/mpegaudioparse/gstmpegaudioparse.c @@ -67,8 +67,8 @@ static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass); static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass); static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse); -static void gst_mp3parse_chain (GstPad * pad, GstData * _data); -static long bpf_from_header (GstMPEGAudioParse * parse, unsigned long header); +static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer); + static int head_check (unsigned long head); static void gst_mp3parse_set_property (GObject * object, guint prop_id, @@ -239,14 +239,18 @@ gst_mp3parse_class_init (GstMPEGAudioParseClass * klass) gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; - g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, g_param_spec_int ("skip", "skip", "skip", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); /* CHECKME */ - g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE, g_param_spec_int ("bitrate", "Bitrate", "Bit Rate", G_MININT, G_MAXINT, 0, G_PARAM_READABLE)); /* CHECKME */ - parent_class = g_type_class_ref (GST_TYPE_ELEMENT); gobject_class->set_property = gst_mp3parse_set_property; gobject_class->get_property = gst_mp3parse_get_property; + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, + g_param_spec_int ("skip", "skip", "skip", + G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE, + g_param_spec_int ("bitrate", "Bitrate", "Bit Rate", + G_MININT, G_MAXINT, 0, G_PARAM_READABLE)); + gstelement_class->change_state = gst_mp3parse_change_state; } @@ -256,16 +260,14 @@ gst_mp3parse_init (GstMPEGAudioParse * mp3parse) mp3parse->sinkpad = gst_pad_new_from_template (gst_static_pad_template_get (&mp3_sink_template), "sink"); - gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad); - gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain); - gst_element_set_loop_function (GST_ELEMENT (mp3parse), NULL); + gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad); mp3parse->srcpad = gst_pad_new_from_template (gst_static_pad_template_get (&mp3_src_template), "src"); + gst_pad_use_fixed_caps (mp3parse->srcpad); gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad); - gst_pad_use_explicit_caps (mp3parse->srcpad); /*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */ mp3parse->partialbuf = NULL; @@ -275,10 +277,10 @@ gst_mp3parse_init (GstMPEGAudioParse * mp3parse) mp3parse->rate = mp3parse->channels = mp3parse->layer = -1; } -static void -gst_mp3parse_chain (GstPad * pad, GstData * _data) +/* FIXME, use adapter */ +static GstFlowReturn +gst_mp3parse_chain (GstPad * pad, GstBuffer * buf) { - GstBuffer *buf = GST_BUFFER (_data); GstMPEGAudioParse *mp3parse; guchar *data; glong size, offset = 0; @@ -287,26 +289,12 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data) GstBuffer *outbuf; guint64 last_ts; - g_return_if_fail (pad != NULL); - g_return_if_fail (GST_IS_PAD (pad)); - g_return_if_fail (buf != NULL); -/* g_return_if_fail(GST_IS_BUFFER(buf)); */ - mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad)); GST_DEBUG ("mp3parse: received buffer of %d bytes", GST_BUFFER_SIZE (buf)); last_ts = GST_BUFFER_TIMESTAMP (buf); - /* FIXME, do flush */ - /* - if (mp3parse->partialbuf) { - gst_buffer_unref(mp3parse->partialbuf); - mp3parse->partialbuf = NULL; - } - mp3parse->in_flush = TRUE; - */ - /* if we have something left from the previous frame */ if (mp3parse->partialbuf) { GstBuffer *newbuf; @@ -332,15 +320,19 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data) /* search for a possible start byte */ for (; ((offset < size - 4) && (data[offset] != 0xff)); offset++) skipped++; - if (skipped && !mp3parse->in_flush) { + if (skipped) { GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped); } /* construct the header word */ header = GST_READ_UINT32_BE (data + offset); /* if it's a valid header, go ahead and send off the frame */ if (head_check (header)) { - /* calculate the bpf of the frame */ - bpf = bpf_from_header (mp3parse, header); + guint bitrate = 0, layer = 0, rate = 0, channels = 0; + + if (!(bpf = mp3_type_frame_length_from_header (header, &layer, + &channels, &bitrate, &rate))) { + g_error ("Header failed internal error"); + } /******************************************************************************** * robust seek support @@ -387,18 +379,13 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data) bpf); break; } else { - guint bitrate, layer, rate, channels; - - if (!mp3_type_frame_length_from_header (header, &layer, - &channels, &bitrate, &rate)) { - g_error ("Header failed internal error"); - } if (channels != mp3parse->channels || rate != mp3parse->rate || layer != mp3parse->layer || bitrate != mp3parse->bit_rate) { GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate); - gst_pad_set_explicit_caps (mp3parse->srcpad, caps); + gst_pad_set_caps (mp3parse->srcpad, caps); + gst_caps_unref (caps); mp3parse->channels = channels; mp3parse->layer = layer; @@ -412,23 +399,18 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data) if (mp3parse->skip == 0) { GST_DEBUG ("mp3parse: pushing buffer of %d bytes", GST_BUFFER_SIZE (outbuf)); - if (mp3parse->in_flush) { - /* FIXME do some sort of flush event */ - mp3parse->in_flush = FALSE; - } GST_BUFFER_TIMESTAMP (outbuf) = last_ts; + if (mp3parse->layer == 1) { GST_BUFFER_DURATION (outbuf) = 384 * GST_SECOND / mp3parse->rate; } else { GST_BUFFER_DURATION (outbuf) = 1152 * GST_SECOND / mp3parse->rate; } - if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) { - gst_pad_push (mp3parse->srcpad, GST_DATA (outbuf)); - } else { - GST_DEBUG ("No capsnego yet, delaying buffer push"); - gst_buffer_unref (outbuf); - } + gst_buffer_set_caps (outbuf, GST_PAD_CAPS (pad)); + + gst_pad_push (mp3parse->srcpad, outbuf); + } else { GST_DEBUG ("mp3parse: skipping buffer of %d bytes", GST_BUFFER_SIZE (outbuf)); @@ -438,8 +420,7 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data) } } else { offset++; - if (!mp3parse->in_flush) - GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)"); + GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)"); } } /* if we have processed this block and there are still */ @@ -457,19 +438,10 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data) gst_buffer_unref (mp3parse->partialbuf); mp3parse->partialbuf = NULL; } -} - -static long -bpf_from_header (GstMPEGAudioParse * parse, unsigned long header) -{ - guint bitrate, layer, rate, channels, length; - if (!(length = mp3_type_frame_length_from_header (header, &layer, - &channels, &bitrate, &rate))) { - return 0; - } + gst_object_unref (mp3parse); - return length; + return GST_FLOW_OK; } static gboolean @@ -561,8 +533,8 @@ static GstElementStateReturn gst_mp3parse_change_state (GstElement * element) { GstMPEGAudioParse *src; + GstElementStateReturn result; - g_return_val_if_fail (GST_IS_MP3PARSE (element), GST_STATE_FAILURE); src = GST_MP3PARSE (element); switch (GST_STATE_TRANSITION (element)) { @@ -575,10 +547,9 @@ gst_mp3parse_change_state (GstElement * element) break; } - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + result = GST_ELEMENT_CLASS (parent_class)->change_state (element); - return GST_STATE_SUCCESS; + return result; } static gboolean diff --git a/gst/mpegaudioparse/gstmpegaudioparse.h b/gst/mpegaudioparse/gstmpegaudioparse.h index ce8121a..a8c2a23 100644 --- a/gst/mpegaudioparse/gstmpegaudioparse.h +++ b/gst/mpegaudioparse/gstmpegaudioparse.h @@ -24,11 +24,7 @@ #include - -#ifdef __cplusplus -extern "C" { -#endif /* __cplusplus */ - +G_BEGIN_DECLS #define GST_TYPE_MP3PARSE \ (gst_mp3parse_get_type()) @@ -62,10 +58,6 @@ struct _GstMPEGAudioParseClass { GType gst_mp3parse_get_type(void); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - +G_END_DECLS #endif /* __MP3PARSE_H__ */