* consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
* for a royalty free (and often higher quality) alternative.
* </para>
+ * <title>Output sample rate</title>
+ * <para>
+ * If no fixed output sample rate is negotiated on the element's src pad,
+ * the element will choose an optimal sample rate to resample to internally.
+ * For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
+ * get resampled to 32 KHz. Use filter caps on the src pad to force a
+ * particular sample rate.
+ * </para>
* <title>Writing metadata (tags)</title>
* <para>
* Whilst the lame encoder element does claim to implement the GstTagSetter
* <programlisting>
* gst-launch -v cdda://5 ! audioconvert ! lame bitrate=192 ! filesink location=track5.mp3
* </programlisting>
+ * <para>
+ * Encode to a fixed sample rate:
+ * </para>
+ * <programlisting>
+ * gst-launch -v audiotestsrc num-buffers=10 ! audio/x-raw-int,rate=44100,channels=1 ! lame bitrate=48 mode=3 ! filesink location=test.mp3
+ * </programlisting>
* </refsect2>
*
- * Last reviewed on 2006-10-13 (0.10.4)
+ * Last reviewed on 2007-07-24 (0.10.7)
*/
#ifdef HAVE_CONFIG_H
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_lame_change_state);
}
+static gboolean
+gst_lame_src_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GST_DEBUG_OBJECT (pad, "src_setcaps %s", gst_caps_to_string (caps));
+ return TRUE;
+}
+
static gboolean
gst_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
{
lame->srcpad =
gst_pad_new_from_static_template (&gst_lame_src_template, "src");
+ gst_pad_set_setcaps_function (lame->srcpad,
+ GST_DEBUG_FUNCPTR (gst_lame_src_setcaps));
gst_element_add_pad (GST_ELEMENT (lame), lame->srcpad);
/* create an encoder state so we can ask about defaults */
static gboolean
gst_lame_setup (GstLame * lame)
{
+
#define CHECK_ERROR(command) G_STMT_START {\
if ((command) < 0) { \
GST_ERROR_OBJECT (lame, "setup failed: " G_STRINGIFY (command)); \
return FALSE; \
} \
}G_STMT_END
+
int retval;
+ GstCaps *src_caps;
+ GstStructure *structure;
+ gint samplerate;
GST_DEBUG_OBJECT (lame, "starting setup");
if (lame->lgf == NULL)
return FALSE;
- /* let lame choose a default samplerate */
- lame_set_out_samplerate (lame->lgf, 0);
-
/* copy the parameters over */
lame_set_in_samplerate (lame->lgf, lame->samplerate);
+ /* let lame choose default samplerate unless outgoing sample rate is fixed */
+ src_caps = gst_pad_get_allowed_caps (lame->srcpad);
+ structure = gst_caps_get_structure (src_caps, 0);
+
+ if (gst_structure_get_int (structure, "rate", &samplerate)) {
+ GST_DEBUG_OBJECT (lame, "Setting sample rate to %d as fixed in src caps",
+ samplerate);
+ lame_set_out_samplerate (lame->lgf, samplerate);
+ } else {
+ GST_DEBUG_OBJECT (lame, "Letting lame choose sample rate");
+ lame_set_out_samplerate (lame->lgf, 0);
+ }
+
/* force mono encoding if we only have one channel */
if (lame->num_channels == 1)
lame->mode = 3;
--- /dev/null
+/* GStreamer
+ *
+ * unit test for lame
+ *
+ * Copyright (C) 2007 Thomas Vander Stichele <thomas at apestaart dot 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 (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 <gst/check/gstcheck.h>
+#include <gst/check/gstbufferstraw.h>
+
+#ifndef GST_DISABLE_PARSE
+
+GST_START_TEST (test_format)
+{
+ GstElement *bin;
+ GstPad *pad;
+ gchar *pipe_str;
+ GstBuffer *buffer;
+ GError *error = NULL;
+
+ pipe_str = g_strdup_printf ("audiotestsrc num-buffers=1 "
+ "! audio/x-raw-int, rate=22050, channels=1 "
+ "! lame bitrate=24 mode=3 " "! audio/mpeg,rate=22050 ! fakesink");
+
+ bin = gst_parse_launch (pipe_str, &error);
+ fail_unless (bin != NULL, "Error parsing pipeline: %s",
+ error ? error->message : "(invalid error)");
+ g_free (pipe_str);
+
+ /* get the pad */
+ {
+ GstElement *sink = gst_bin_get_by_name (GST_BIN (bin), "fakesink0");
+
+ fail_unless (sink != NULL, "Could not get fakesink out of bin");
+ pad = gst_element_get_pad (sink, "sink");
+ fail_unless (pad != NULL, "Could not get pad out of fakesink");
+ gst_object_unref (sink);
+ }
+
+ gst_buffer_straw_start_pipeline (bin, pad);
+
+ buffer = gst_buffer_straw_get_buffer (bin, pad);
+
+ gst_buffer_straw_stop_pipeline (bin, pad);
+
+ gst_object_unref (pad);
+ gst_object_unref (bin);
+}
+
+GST_END_TEST;
+
+#endif /* #ifndef GST_DISABLE_PARSE */
+
+Suite *
+lame_suite (void)
+{
+ Suite *s = suite_create ("lame");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+
+#ifndef GST_DISABLE_PARSE
+ tcase_add_test (tc_chain, test_format);
+#endif
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = lame_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}