+2007-03-14 Julien MOUTTE <julien@moutte.net>
+
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
+ (audioresample_transform_size), (audioresample_do_output),
+ (audioresample_transform), (audioresample_pushthrough): Handle
+ discontinuous streams.
+ * gst/audioresample/gstaudioresample.h:
+ * tests/check/elements/audioresample.c:
+ (test_discont_stream_instance), (GST_START_TEST),
+ (audioresample_suite): Add a test for discontinuous streams.
+ * win32/common/config.h: Updated.
+
2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
* po/af.po:
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
audioresample->filter_length = DEFAULT_FILTERLEN;
+
+ audioresample->need_discont = FALSE;
}
/* vmethods */
gboolean use_internal = FALSE; /* whether we use the internal state */
gboolean ret = TRUE;
- GST_DEBUG_OBJECT (base, "asked to transform size %d in direction %s",
+ GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
size, direction == GST_PAD_SINK ? "SINK" : "SRC");
if (direction == GST_PAD_SINK) {
sinkcaps = caps;
/* we make room for one extra sample, given that the resampling filter
* can output an extra one for non-integral i_rate/o_rate */
- GST_DEBUG_OBJECT (base, "transformed size %d to %d", size, *othersize);
+ GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);
if (!use_internal) {
resample_free (state);
r = audioresample->resample;
outsize = resample_get_output_size (r);
- GST_DEBUG_OBJECT (audioresample, "audioresample can give me %d bytes",
- outsize);
+ GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
/* protect against mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
}
GST_BUFFER_SIZE (outbuf) = outsize;
+ if (G_UNLIKELY (audioresample->need_discont)) {
+ GST_DEBUG_OBJECT (audioresample,
+ "marking this buffer with the DISCONT flag");
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ audioresample->need_discont = FALSE;
+ }
+
GST_LOG_OBJECT (audioresample, "transformed to buffer of %ld bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
+ /* check for timestamp discontinuities and flush/reset if needed */
+ if (GST_CLOCK_TIME_IS_VALID (audioresample->prev_ts) &&
+ GST_CLOCK_TIME_IS_VALID (audioresample->prev_duration)) {
+ GstClockTime ts_expected = audioresample->prev_ts +
+ audioresample->prev_duration;
+ GstClockTimeDiff ts_diff = GST_CLOCK_DIFF (ts_expected, timestamp);
+
+ if (G_UNLIKELY (ts_diff != 0)) {
+ GST_WARNING_OBJECT (audioresample,
+ "encountered timestamp discontinuity of %" G_GINT64_FORMAT, ts_diff);
+ /* Flush internal samples */
+ audioresample_pushthrough (audioresample);
+ /* Inform downstream element about discontinuity */
+ audioresample->need_discont = TRUE;
+ /* We want to recalculate the offset */
+ audioresample->ts_offset = -1;
+ }
+ }
+
if (audioresample->ts_offset == -1) {
/* if we don't know the initial offset yet, calculate it based on the
* input timestamp. */
gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
}
}
+ audioresample->prev_ts = timestamp;
+ audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
/* need to memdup, resample takes ownership. */
datacopy = g_memdup (data, size);
r = audioresample->resample;
outsize = resample_get_output_size (r);
- if (outsize == 0)
+ if (outsize == 0) {
+ GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
goto done;
+ }
- outbuf = gst_buffer_new_and_alloc (outsize);
+ trans = GST_BASE_TRANSFORM (audioresample);
- res = audioresample_do_output (audioresample, outbuf);
- if (res != GST_FLOW_OK)
+ res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
+ GST_PAD_CAPS (trans->srcpad), &outbuf);
+ if (G_UNLIKELY (res != GST_FLOW_OK)) {
+ GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
+ outsize);
goto done;
+ }
- trans = GST_BASE_TRANSFORM (audioresample);
+ res = audioresample_do_output (audioresample, outbuf);
+ if (G_UNLIKELY (res != GST_FLOW_OK))
+ goto done;
res = gst_pad_push (trans->srcpad, outbuf);
GstCaps *srccaps, *sinkcaps;
gboolean passthru;
+ gboolean need_discont;
guint64 offset;
guint64 ts_offset;
GstClockTime next_ts;
+ GstClockTime prev_ts, prev_duration;
int channels;
int i_rate;
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)
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 *audioresample;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ GstClockTime ints;
+
+ int i, j;
+ gint16 *p;
+
+ audioresample = setup_audioresample (2, inrate, outrate);
+ caps = gst_pad_get_negotiated_caps (mysrcpad);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ fail_unless (gst_element_set_state (audioresample,
+ 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;
+ }
+
+ /* 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));
+ if (j > 1) {
+ fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
+ "expected discont buffer");
+ }
+ }
+
+ /* cleanup */
+ gst_caps_unref (caps);
+ cleanup_audioresample (audioresample);
+}
+
+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 *audioresample;
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);
return s;
#define GST_LICENSE "LGPL"
/* package name in plugins */
-#define GST_PACKAGE_NAME "GStreamer Base Plug-ins source release"
+#define GST_PACKAGE_NAME "GStreamer Base Plug-ins CVS/prerelease"
/* package origin */
#define GST_PACKAGE_ORIGIN "Unknown package origin"
#undef PACKAGE_NAME "GStreamer Base Plug-ins"
/* Define to the full name and version of this package. */
-#undef PACKAGE_STRING "GStreamer Base Plug-ins 0.10.12"
+#undef PACKAGE_STRING "GStreamer Base Plug-ins 0.10.12.1"
/* Define to the one symbol short name of this package. */
#undef PACKAGE_TARNAME "gst-plugins-base"
/* Define to the version of this package. */
-#undef PACKAGE_VERSION "0.10.12"
+#undef PACKAGE_VERSION "0.10.12.1"
/* directory where plugins are located */
#undef PLUGINDIR
#undef STDC_HEADERS
/* Version number of package */
-#define VERSION "0.10.12"
+#define VERSION "0.10.12.1"
/* Define to 1 if your processor stores words with the most significant byte
first (like Motorola and SPARC, unlike Intel and VAX). */
projects / platform / upstream / gstreamer.git / commitdiff