mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-23 16:50:47 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...

Browse source 
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
This commit is contained in:
Sebastian Dröge 2008-10-29 12:11:20 +00:00
parent b2b865beac
commit 80c3258a6a
5 changed files with 667 additions and 109 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

20
ChangeLog
View file

@ -1,3 +1,23 @@
2008-10-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
2008-10-29 Jan Schmidt <thaytan@noraisin.net> 2008-10-29 Jan Schmidt <thaytan@noraisin.net>
* ext/resindvd/resindvdsrc.c: * ext/resindvd/resindvdsrc.c:

169
gst/speexresample/gstspeexresample.c
View file

@ -183,9 +183,9 @@ gst_speex_resample_start (GstBaseTransform * base)
{ {
GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base); GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
resample->ts_offset = -1; resample->next_offset = -1;
resample->offset = -1;
resample->next_ts = -1; resample->next_ts = -1;
resample->next_upstream_ts = -1;
return TRUE; return TRUE;
} }
@ -224,7 +224,7 @@ gst_speex_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
if (G_UNLIKELY (!ret)) if (G_UNLIKELY (!ret))
return FALSE; return FALSE;
*size = width * channels / 8; *size = gst_util_uint64_scale (width, channels, 8);
return TRUE; return TRUE;
} }
@ -595,26 +595,18 @@ gst_speex_resample_push_drain (GstSpeexResample * resample)
return; return;
} }
GST_BUFFER_OFFSET (buf) = resample->offset; GST_BUFFER_DURATION (buf) =
GST_BUFFER_TIMESTAMP (buf) = resample->next_ts; GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
GST_BUFFER_SIZE (buf) = GST_BUFFER_SIZE (buf) =
out_processed * resample->channels * ((resample->fp) ? 4 : 2); out_processed * resample->channels * ((resample->fp) ? 4 : 2);
if (resample->ts_offset != -1) { if (GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
resample->offset += out_processed; GST_BUFFER_OFFSET (buf) = resample->next_offset;
resample->ts_offset += out_processed; GST_BUFFER_OFFSET_END (buf) = resample->next_offset + out_processed;
resample->next_ts = GST_BUFFER_TIMESTAMP (buf) = resample->next_ts;
GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
GST_BUFFER_OFFSET_END (buf) = resample->offset;
/* we calculate DURATION as the difference between "next" timestamp resample->next_ts += GST_BUFFER_DURATION (buf);
* and current timestamp so we ensure a contiguous stream, instead of resample->next_offset += out_processed;
* having rounding errors. */
GST_BUFFER_DURATION (buf) = resample->next_ts - GST_BUFFER_TIMESTAMP (buf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (buf) =
GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
} }
GST_LOG_OBJECT (resample, GST_LOG_OBJECT (resample,
@ -644,15 +636,15 @@ gst_speex_resample_event (GstBaseTransform * base, GstEvent * event)
break; break;
case GST_EVENT_FLUSH_STOP: case GST_EVENT_FLUSH_STOP:
gst_speex_resample_reset_state (resample); gst_speex_resample_reset_state (resample);
resample->ts_offset = -1; resample->next_offset = -1;
resample->next_ts = -1; resample->next_ts = -1;
resample->offset = -1; resample->next_upstream_ts = -1;
case GST_EVENT_NEWSEGMENT: case GST_EVENT_NEWSEGMENT:
gst_speex_resample_push_drain (resample); gst_speex_resample_push_drain (resample);
gst_speex_resample_reset_state (resample); gst_speex_resample_reset_state (resample);
resample->ts_offset = -1; resample->next_offset = -1;
resample->next_ts = -1; resample->next_ts = -1;
resample->offset = -1; resample->next_upstream_ts = -1;
break; break;
case GST_EVENT_EOS:{ case GST_EVENT_EOS:{
gst_speex_resample_push_drain (resample); gst_speex_resample_push_drain (resample);
@ -671,19 +663,18 @@ gst_speex_resample_check_discont (GstSpeexResample * resample,
GstClockTime timestamp) GstClockTime timestamp)
{ {
if (timestamp != GST_CLOCK_TIME_NONE && if (timestamp != GST_CLOCK_TIME_NONE &&
resample->prev_ts != GST_CLOCK_TIME_NONE && resample->next_upstream_ts != GST_CLOCK_TIME_NONE &&
resample->prev_duration != GST_CLOCK_TIME_NONE && timestamp != resample->next_upstream_ts) {
timestamp != resample->prev_ts + resample->prev_duration) {
/* Potentially a discontinuous buffer. However, it turns out that many /* Potentially a discontinuous buffer. However, it turns out that many
* elements generate imperfect streams due to rounding errors, so we permit * elements generate imperfect streams due to rounding errors, so we permit
* a small error (up to one sample) without triggering a filter * a small error (up to one sample) without triggering a filter
* flush/restart (if triggered incorrectly, this will be audible) */ * flush/restart (if triggered incorrectly, this will be audible) */
GstClockTimeDiff diff = timestamp - GstClockTimeDiff diff = timestamp - resample->next_upstream_ts;
(resample->prev_ts + resample->prev_duration);
if (ABS (diff) > GST_SECOND / resample->inrate) { if (ABS (diff) > (GST_SECOND + resample->inrate - 1) / resample->inrate) {
GST_WARNING_OBJECT (resample, GST_WARNING_OBJECT (resample,
"encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff); "encountered timestamp discontinuity of %s%" GST_TIME_FORMAT,
(diff < 0) ? "-" : "", ABS (diff));
return TRUE; return TRUE;
} }
} }
@ -691,27 +682,6 @@ gst_speex_resample_check_discont (GstSpeexResample * resample,
return FALSE; return FALSE;
} }
static void
gst_speex_fix_output_buffer (GstSpeexResample * resample, GstBuffer * outbuf,
guint diff)
{
GstClockTime timediff = GST_FRAMES_TO_CLOCK_TIME (diff, resample->outrate);
GST_LOG_OBJECT (resample, "Adjusting buffer by %d samples", diff);
GST_BUFFER_DURATION (outbuf) -= timediff;
GST_BUFFER_SIZE (outbuf) -=
diff * ((resample->fp) ? 4 : 2) * resample->channels;
if (resample->ts_offset != -1) {
GST_BUFFER_OFFSET_END (outbuf) -= diff;
resample->offset -= diff;
resample->ts_offset -= diff;
resample->next_ts =
GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
}
}
static GstFlowReturn static GstFlowReturn
gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf, gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
GstBuffer * outbuf) GstBuffer * outbuf)
@ -753,14 +723,6 @@ gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
if (out_processed == 0) { if (out_processed == 0) {
GST_DEBUG_OBJECT (resample, "Converted to 0 samples, buffer dropped"); GST_DEBUG_OBJECT (resample, "Converted to 0 samples, buffer dropped");
if (resample->ts_offset != -1) {
GST_BUFFER_OFFSET_END (outbuf) -= out_len;
resample->offset -= out_len;
resample->ts_offset -= out_len;
resample->next_ts =
GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
}
return GST_BASE_TRANSFORM_FLOW_DROPPED; return GST_BASE_TRANSFORM_FLOW_DROPPED;
} else if (out_len - out_processed != 1) { } else if (out_len - out_processed != 1) {
GST_WARNING_OBJECT (resample, GST_WARNING_OBJECT (resample,
@ -768,9 +730,7 @@ gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
out_len); out_len);
} }
if (G_LIKELY (out_len > out_processed)) { if (G_UNLIKELY (out_len < out_processed)) {
gst_speex_fix_output_buffer (resample, outbuf, out_len - out_processed);
} else {
GST_ERROR_OBJECT (resample, "Wrote more output than allocated!"); GST_ERROR_OBJECT (resample, "Wrote more output than allocated!");
return GST_FLOW_ERROR; return GST_FLOW_ERROR;
} }
@ -781,6 +741,20 @@ gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
resample_resampler_strerror (err)); resample_resampler_strerror (err));
return GST_FLOW_ERROR; return GST_FLOW_ERROR;
} else { } else {
GST_BUFFER_DURATION (outbuf) =
GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
GST_BUFFER_SIZE (outbuf) =
out_processed * resample->channels * ((resample->fp) ? 4 : 2);
if (GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
GST_BUFFER_OFFSET (outbuf) = resample->next_offset;
GST_BUFFER_OFFSET_END (outbuf) = resample->next_offset + out_processed;
resample->next_ts += GST_BUFFER_DURATION (outbuf);
resample->next_offset += out_processed;
}
GST_LOG_OBJECT (resample, GST_LOG_OBJECT (resample,
"Converted to buffer of %u bytes with timestamp %" GST_TIME_FORMAT "Converted to buffer of %u bytes with timestamp %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
@ -801,7 +775,8 @@ gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
guint8 *data; guint8 *data;
gulong size; gulong size;
GstClockTime timestamp; GstClockTime timestamp;
gint outsamples; guint outsamples, insamples;
GstFlowReturn ret;
if (resample->state == NULL) if (resample->state == NULL)
if (G_UNLIKELY (!(resample->state = if (G_UNLIKELY (!(resample->state =
@ -828,53 +803,23 @@ gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
gst_speex_resample_reset_state (resample); gst_speex_resample_reset_state (resample);
/* Inform downstream element about discontinuity */ /* Inform downstream element about discontinuity */
resample->need_discont = TRUE; resample->need_discont = TRUE;
/* We want to recalculate the offset */ /* We want to recalculate the timestamps */
resample->ts_offset = -1; resample->next_ts = -1;
resample->next_upstream_ts = -1;
resample->next_offset = -1;
} }
insamples = GST_BUFFER_SIZE (inbuf) / resample->channels;
insamples /= (resample->fp) ? 4 : 2;
outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels; outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels;
outsamples /= (resample->fp) ? 4 : 2; outsamples /= (resample->fp) ? 4 : 2;
if (resample->ts_offset == -1) { if (GST_CLOCK_TIME_IS_VALID (timestamp)
/* if we don't know the initial offset yet, calculate it based on the && !GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
* input timestamp. */ resample->next_ts = timestamp;
if (GST_CLOCK_TIME_IS_VALID (timestamp)) { resample->next_offset =
GstClockTime stime; GST_CLOCK_TIME_TO_FRAMES (timestamp, resample->outrate);
/* offset used to calculate the timestamps. We use the sample offset for
* this to make it more accurate. We want the first buffer to have the
* same timestamp as the incoming timestamp. */
resample->next_ts = timestamp;
resample->ts_offset =
GST_CLOCK_TIME_TO_FRAMES (timestamp, resample->outrate);
/* offset used to set as the buffer offset, this offset is always
* relative to the stream time, note that timestamp is not... */
stime = (timestamp - base->segment.start) + base->segment.time;
resample->offset = GST_CLOCK_TIME_TO_FRAMES (stime, resample->outrate);
}
}
resample->prev_ts = timestamp;
resample->prev_duration = GST_BUFFER_DURATION (inbuf);
GST_BUFFER_OFFSET (outbuf) = resample->offset;
GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
if (resample->ts_offset != -1) {
resample->offset += outsamples;
resample->ts_offset += outsamples;
resample->next_ts =
GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
GST_BUFFER_OFFSET_END (outbuf) = resample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
GST_BUFFER_DURATION (outbuf) = resample->next_ts -
GST_BUFFER_TIMESTAMP (outbuf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (outbuf) =
GST_FRAMES_TO_CLOCK_TIME (outsamples, resample->outrate);
} }
if (G_UNLIKELY (resample->need_discont)) { if (G_UNLIKELY (resample->need_discont)) {
@ -883,7 +828,19 @@ gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
resample->need_discont = FALSE; resample->need_discont = FALSE;
} }
return gst_speex_resample_process (resample, inbuf, outbuf); ret = gst_speex_resample_process (resample, inbuf, outbuf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
return ret;
if (GST_CLOCK_TIME_IS_VALID (timestamp)
&& !GST_CLOCK_TIME_IS_VALID (resample->next_upstream_ts))
resample->next_upstream_ts = timestamp;
if (GST_CLOCK_TIME_IS_VALID (resample->next_upstream_ts))
resample->next_upstream_ts +=
GST_FRAMES_TO_CLOCK_TIME (insamples, resample->inrate);
return GST_FLOW_OK;
} }
static gboolean static gboolean

5
gst/speexresample/gstspeexresample.h
View file

@ -57,10 +57,9 @@ struct _GstSpeexResample {
gboolean need_discont; gboolean need_discont;
guint64 offset; guint64 next_offset;
guint64 ts_offset;
GstClockTime next_ts; GstClockTime next_ts;
GstClockTime prev_ts, prev_duration; GstClockTime next_upstream_ts;
gboolean fp; gboolean fp;
gint channels; gint channels;

3
tests/check/Makefile.am
View file

@ -88,6 +88,7 @@ check_PROGRAMS = \
$(check_timidity) \ $(check_timidity) \
$(check_x264enc) \ $(check_x264enc) \
elements/selector \ elements/selector \
elements/speexresample \
elements/y4menc \ elements/y4menc \
$(check_metadata) $(check_metadata)
@ -101,3 +102,5 @@ LDADD = $(GST_OBJ_LIBS) $(GST_CHECK_LIBS) $(CHECK_LIBS)
elements_timidity_CFLAGS = $(GST_BASE_CFLAGS) $(AM_CFLAGS) elements_timidity_CFLAGS = $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_timidity_LDADD = $(GST_BASE_LIBS) $(LDADD) elements_timidity_LDADD = $(GST_BASE_LIBS) $(LDADD)
elements_speexresample_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(AM_CFLAGS)
elements_speexresample_LDADD = $(GST_PLUGINS_BASE_LIBS) $(LDADD) -lgstaudio-$(GST_MAJORMINOR)

579
tests/check/elements/speexresample.c Normal file
View file

@ -0,0 +1,579 @@
/* 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_TEMPLATE_STRING \
"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"
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)
{
GstElement *speexresample;
GstCaps *caps;
GstStructure *structure;
GST_DEBUG ("setup_speexresample");
speexresample = gst_check_setup_element ("speexresample");
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, inrate, 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);
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, outrate, 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);
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);
#if 0
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);
#endif
}
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);
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);
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 ("speexresample");
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);
/* 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 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);
return s;
}
GST_CHECK_MAIN (speexresample);