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gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
Original commit message from CVS: * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain): If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather than a single very large buffer. Avoids unreasonably large single buffer allocations when encountering a large gap. * tests/check/elements/audiorate.c: (GST_START_TEST), (audiorate_suite): Add a test for this.
This commit is contained in:
Michael Smith
parent
9730f452ee
commit
9f9e76bc99
3 changed files with 119 additions and 32 deletions
11
ChangeLog
11
ChangeLog
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@ -1,3 +1,14 @@
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2007-08-09 Michael Smith <msmith@fluendo.com>
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* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
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If we have a large (> 1 second) discontinuity, push a series of
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smaller buffers rather than a single very large buffer. Avoids
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unreasonably large single buffer allocations when encountering a
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large gap.
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* tests/check/elements/audiorate.c: (GST_START_TEST),
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(audiorate_suite):
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Add a test for this.
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2007-08-09 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/playback/gstplaybasebin.c: (group_commit),
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@ -543,45 +543,60 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
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gint fillsize;
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guint64 fillsamples;
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/* We don't want to allocate a single unreasonably huge buffer - it might
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be hundreds of megabytes. So, limit each output buffer to one second of
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audio */
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fillsamples = in_offset - audiorate->next_offset;
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fillsize = fillsamples * audiorate->bytes_per_sample;
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fill = gst_buffer_new_and_alloc (fillsize);
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/* FIXME, 0 might not be the silence byte for the negotiated format. */
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memset (GST_BUFFER_DATA (fill), 0, fillsize);
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while (fillsamples > 0) {
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guint64 cursamples = MIN (fillsamples, audiorate->rate);
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GST_DEBUG_OBJECT (audiorate, "inserting %lld samples", fillsamples);
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GST_DEBUG_OBJECT (audiorate,
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"inserting %lld samples of total %lld at ts %lld", cursamples,
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fillsamples, audiorate->next_ts);
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GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
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audiorate->next_offset += fillsamples;
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GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset;
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fillsamples -= cursamples;
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fillsize = cursamples * audiorate->bytes_per_sample;
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/* Use next timestamp, then calculate following timestamp based on in_offset
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* to get duration. Neccesary complexity to get 'perfect' streams */
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GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
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audiorate->next_ts = gst_util_uint64_scale_int (in_offset,
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GST_SECOND, audiorate->rate);
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GST_BUFFER_DURATION (fill) = audiorate->next_ts -
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GST_BUFFER_TIMESTAMP (fill);
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fill = gst_buffer_new_and_alloc (fillsize);
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/* FIXME, 0 might not be the silence byte for the negotiated format. */
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memset (GST_BUFFER_DATA (fill), 0, fillsize);
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/* we created this buffer to fill a gap */
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GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);
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/* set discont if it's pending, this is mostly done for the first buffer and
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* after a flushing seek */
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if (audiorate->discont) {
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GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT);
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audiorate->discont = FALSE;
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GST_DEBUG_OBJECT (audiorate, "inserting %lld samples", cursamples);
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GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
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audiorate->next_offset += cursamples;
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GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset;
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/* Use next timestamp, then calculate following timestamp based on
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* offset to get duration. Neccesary complexity to get 'perfect'
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* streams */
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GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
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audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,
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GST_SECOND, audiorate->rate);
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GST_BUFFER_DURATION (fill) = audiorate->next_ts -
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GST_BUFFER_TIMESTAMP (fill);
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/* we created this buffer to fill a gap */
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GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);
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/* set discont if it's pending, this is mostly done for the first buffer
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* and after a flushing seek */
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if (audiorate->discont) {
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GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT);
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audiorate->discont = FALSE;
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}
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gst_buffer_set_caps (fill, GST_PAD_CAPS (audiorate->srcpad));
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ret = gst_pad_push (audiorate->srcpad, fill);
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if (ret != GST_FLOW_OK)
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goto beach;
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audiorate->out++;
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audiorate->add += cursamples;
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if (!audiorate->silent)
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g_object_notify (G_OBJECT (audiorate), "add");
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}
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gst_buffer_set_caps (fill, GST_PAD_CAPS (audiorate->srcpad));
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ret = gst_pad_push (audiorate->srcpad, fill);
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if (ret != GST_FLOW_OK)
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goto beach;
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audiorate->out++;
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audiorate->add += fillsamples;
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if (!audiorate->silent)
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g_object_notify (G_OBJECT (audiorate), "add");
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} else if (in_offset < audiorate->next_offset) {
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/* need to remove samples */
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if (in_offset_end <= audiorate->next_offset) {
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@ -25,7 +25,6 @@
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#include <gst/check/gstcheck.h>
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/* helper element to insert additional buffers overlapping with previous ones */
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static gdouble injector_inject_probability = 0.0;
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typedef GstElement TestInjector;
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@ -386,6 +385,67 @@ GST_END_TEST;
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/* TODO: also do all tests with channels=1 and channels=2 */
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-float,channels=1,rate=44100,width=32")
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);
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-float,channels=1,rate=44100,width=32")
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);
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GST_START_TEST (test_large_discont)
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{
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GstElement *audiorate;
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GstCaps *caps;
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GstPad *srcpad, *sinkpad;
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GstBuffer *buf;
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audiorate = gst_check_setup_element ("audiorate");
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caps = gst_caps_new_simple ("audio/x-raw-float",
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"channels", G_TYPE_INT, 1,
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"rate", G_TYPE_INT, 44100, "width", G_TYPE_INT, 32, NULL);
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srcpad = gst_check_setup_src_pad (audiorate, &srctemplate, caps);
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sinkpad = gst_check_setup_sink_pad (audiorate, &sinktemplate, caps);
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gst_pad_set_active (srcpad, TRUE);
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gst_pad_set_active (sinkpad, TRUE);
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fail_unless (gst_element_set_state (audiorate,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"failed to set audiorate playing");
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buf = gst_buffer_new_and_alloc (4);
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gst_buffer_set_caps (buf, caps);
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GST_BUFFER_TIMESTAMP (buf) = 0;
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gst_pad_push (srcpad, buf);
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fail_unless_equals_int (g_list_length (buffers), 1);
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buf = gst_buffer_new_and_alloc (4);
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gst_buffer_set_caps (buf, caps);
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GST_BUFFER_TIMESTAMP (buf) = 2 * GST_SECOND;
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gst_pad_push (srcpad, buf);
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/* Now we should have 3 more buffers: the one we injected, plus _two_ filler
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* buffers, because the gap is > 1 second (but less than 2 seconds) */
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fail_unless_equals_int (g_list_length (buffers), 4);
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gst_element_set_state (audiorate, GST_STATE_NULL);
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gst_caps_unref (caps);
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gst_check_teardown_sink_pad (audiorate);
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gst_check_teardown_src_pad (audiorate);
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gst_object_unref (audiorate);
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}
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GST_END_TEST;
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static Suite *
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audiorate_suite (void)
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{
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@ -401,6 +461,7 @@ audiorate_suite (void)
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tcase_add_test (tc_chain, test_perfect_stream_inject10);
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tcase_add_test (tc_chain, test_perfect_stream_inject90);
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tcase_add_test (tc_chain, test_perfect_stream_drop45_inject25);
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tcase_add_test (tc_chain, test_large_discont);
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return s;
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}
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