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gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
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
parent
b2b865beac
commit
80c3258a6a
5 changed files with 667 additions and 109 deletions
20
ChangeLog
20
ChangeLog
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@ -1,3 +1,23 @@
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2008-10-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
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(gst_speex_resample_get_unit_size),
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(gst_speex_resample_push_drain), (gst_speex_resample_event),
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(gst_speex_resample_check_discont), (gst_speex_resample_process),
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(gst_speex_resample_transform):
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* gst/speexresample/gstspeexresample.h:
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Rewrite timestamp tracking to make it more robust and guarantee
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a continous stream.
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* tests/check/Makefile.am:
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* tests/check/elements/speexresample.c: (setup_speexresample),
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(cleanup_speexresample), (fail_unless_perfect_stream),
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(test_perfect_stream_instance), (GST_START_TEST),
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(test_discont_stream_instance), (live_switch_alloc_only_48000),
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(live_switch_get_sink_caps), (live_switch_push),
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(speexresample_suite):
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Add unit tests for speexresample based on the audioresample unit tests.
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2008-10-29 Jan Schmidt <thaytan@noraisin.net>
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* ext/resindvd/resindvdsrc.c:
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@ -183,9 +183,9 @@ gst_speex_resample_start (GstBaseTransform * base)
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{
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GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
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resample->ts_offset = -1;
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resample->offset = -1;
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resample->next_offset = -1;
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resample->next_ts = -1;
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resample->next_upstream_ts = -1;
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return TRUE;
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}
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@ -224,7 +224,7 @@ gst_speex_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
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if (G_UNLIKELY (!ret))
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return FALSE;
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*size = width * channels / 8;
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*size = gst_util_uint64_scale (width, channels, 8);
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return TRUE;
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}
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@ -595,26 +595,18 @@ gst_speex_resample_push_drain (GstSpeexResample * resample)
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return;
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}
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GST_BUFFER_OFFSET (buf) = resample->offset;
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GST_BUFFER_TIMESTAMP (buf) = resample->next_ts;
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GST_BUFFER_DURATION (buf) =
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GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
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GST_BUFFER_SIZE (buf) =
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out_processed * resample->channels * ((resample->fp) ? 4 : 2);
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if (resample->ts_offset != -1) {
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resample->offset += out_processed;
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resample->ts_offset += out_processed;
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resample->next_ts =
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GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
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GST_BUFFER_OFFSET_END (buf) = resample->offset;
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if (GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
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GST_BUFFER_OFFSET (buf) = resample->next_offset;
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GST_BUFFER_OFFSET_END (buf) = resample->next_offset + out_processed;
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GST_BUFFER_TIMESTAMP (buf) = resample->next_ts;
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/* we calculate DURATION as the difference between "next" timestamp
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* and current timestamp so we ensure a contiguous stream, instead of
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* having rounding errors. */
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GST_BUFFER_DURATION (buf) = resample->next_ts - GST_BUFFER_TIMESTAMP (buf);
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} else {
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/* no valid offset know, we can still sortof calculate the duration though */
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GST_BUFFER_DURATION (buf) =
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GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
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resample->next_ts += GST_BUFFER_DURATION (buf);
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resample->next_offset += out_processed;
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}
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GST_LOG_OBJECT (resample,
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@ -644,15 +636,15 @@ gst_speex_resample_event (GstBaseTransform * base, GstEvent * event)
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break;
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case GST_EVENT_FLUSH_STOP:
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gst_speex_resample_reset_state (resample);
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resample->ts_offset = -1;
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resample->next_offset = -1;
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resample->next_ts = -1;
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resample->offset = -1;
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resample->next_upstream_ts = -1;
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case GST_EVENT_NEWSEGMENT:
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gst_speex_resample_push_drain (resample);
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gst_speex_resample_reset_state (resample);
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resample->ts_offset = -1;
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resample->next_offset = -1;
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resample->next_ts = -1;
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resample->offset = -1;
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resample->next_upstream_ts = -1;
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break;
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case GST_EVENT_EOS:{
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gst_speex_resample_push_drain (resample);
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@ -671,19 +663,18 @@ gst_speex_resample_check_discont (GstSpeexResample * resample,
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GstClockTime timestamp)
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{
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if (timestamp != GST_CLOCK_TIME_NONE &&
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resample->prev_ts != GST_CLOCK_TIME_NONE &&
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resample->prev_duration != GST_CLOCK_TIME_NONE &&
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timestamp != resample->prev_ts + resample->prev_duration) {
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resample->next_upstream_ts != GST_CLOCK_TIME_NONE &&
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timestamp != resample->next_upstream_ts) {
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/* Potentially a discontinuous buffer. However, it turns out that many
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* elements generate imperfect streams due to rounding errors, so we permit
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* a small error (up to one sample) without triggering a filter
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* flush/restart (if triggered incorrectly, this will be audible) */
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GstClockTimeDiff diff = timestamp -
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(resample->prev_ts + resample->prev_duration);
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GstClockTimeDiff diff = timestamp - resample->next_upstream_ts;
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if (ABS (diff) > GST_SECOND / resample->inrate) {
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if (ABS (diff) > (GST_SECOND + resample->inrate - 1) / resample->inrate) {
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GST_WARNING_OBJECT (resample,
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"encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
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"encountered timestamp discontinuity of %s%" GST_TIME_FORMAT,
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(diff < 0) ? "-" : "", ABS (diff));
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return TRUE;
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}
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}
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@ -691,27 +682,6 @@ gst_speex_resample_check_discont (GstSpeexResample * resample,
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return FALSE;
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}
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static void
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gst_speex_fix_output_buffer (GstSpeexResample * resample, GstBuffer * outbuf,
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guint diff)
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{
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GstClockTime timediff = GST_FRAMES_TO_CLOCK_TIME (diff, resample->outrate);
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GST_LOG_OBJECT (resample, "Adjusting buffer by %d samples", diff);
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GST_BUFFER_DURATION (outbuf) -= timediff;
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GST_BUFFER_SIZE (outbuf) -=
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diff * ((resample->fp) ? 4 : 2) * resample->channels;
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if (resample->ts_offset != -1) {
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GST_BUFFER_OFFSET_END (outbuf) -= diff;
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resample->offset -= diff;
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resample->ts_offset -= diff;
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resample->next_ts =
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GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
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}
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}
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static GstFlowReturn
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gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
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GstBuffer * outbuf)
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@ -753,14 +723,6 @@ gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
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if (out_processed == 0) {
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GST_DEBUG_OBJECT (resample, "Converted to 0 samples, buffer dropped");
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if (resample->ts_offset != -1) {
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GST_BUFFER_OFFSET_END (outbuf) -= out_len;
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resample->offset -= out_len;
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resample->ts_offset -= out_len;
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resample->next_ts =
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GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
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}
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return GST_BASE_TRANSFORM_FLOW_DROPPED;
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} else if (out_len - out_processed != 1) {
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GST_WARNING_OBJECT (resample,
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@ -768,9 +730,7 @@ gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
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out_len);
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}
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if (G_LIKELY (out_len > out_processed)) {
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gst_speex_fix_output_buffer (resample, outbuf, out_len - out_processed);
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} else {
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if (G_UNLIKELY (out_len < out_processed)) {
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GST_ERROR_OBJECT (resample, "Wrote more output than allocated!");
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return GST_FLOW_ERROR;
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}
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@ -781,6 +741,20 @@ gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
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resample_resampler_strerror (err));
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return GST_FLOW_ERROR;
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} else {
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GST_BUFFER_DURATION (outbuf) =
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GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
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GST_BUFFER_SIZE (outbuf) =
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out_processed * resample->channels * ((resample->fp) ? 4 : 2);
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if (GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
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GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
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GST_BUFFER_OFFSET (outbuf) = resample->next_offset;
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GST_BUFFER_OFFSET_END (outbuf) = resample->next_offset + out_processed;
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resample->next_ts += GST_BUFFER_DURATION (outbuf);
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resample->next_offset += out_processed;
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}
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GST_LOG_OBJECT (resample,
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"Converted to buffer of %u bytes with timestamp %" GST_TIME_FORMAT
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", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
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@ -801,7 +775,8 @@ gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
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guint8 *data;
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gulong size;
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GstClockTime timestamp;
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gint outsamples;
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guint outsamples, insamples;
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GstFlowReturn ret;
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if (resample->state == NULL)
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if (G_UNLIKELY (!(resample->state =
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@ -828,53 +803,23 @@ gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
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gst_speex_resample_reset_state (resample);
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/* Inform downstream element about discontinuity */
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resample->need_discont = TRUE;
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/* We want to recalculate the offset */
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resample->ts_offset = -1;
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/* We want to recalculate the timestamps */
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resample->next_ts = -1;
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resample->next_upstream_ts = -1;
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resample->next_offset = -1;
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}
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insamples = GST_BUFFER_SIZE (inbuf) / resample->channels;
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insamples /= (resample->fp) ? 4 : 2;
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outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels;
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outsamples /= (resample->fp) ? 4 : 2;
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if (resample->ts_offset == -1) {
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/* if we don't know the initial offset yet, calculate it based on the
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* input timestamp. */
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if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
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GstClockTime stime;
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/* offset used to calculate the timestamps. We use the sample offset for
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* this to make it more accurate. We want the first buffer to have the
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* same timestamp as the incoming timestamp. */
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resample->next_ts = timestamp;
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resample->ts_offset =
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GST_CLOCK_TIME_TO_FRAMES (timestamp, resample->outrate);
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/* offset used to set as the buffer offset, this offset is always
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* relative to the stream time, note that timestamp is not... */
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stime = (timestamp - base->segment.start) + base->segment.time;
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resample->offset = GST_CLOCK_TIME_TO_FRAMES (stime, resample->outrate);
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}
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}
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resample->prev_ts = timestamp;
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resample->prev_duration = GST_BUFFER_DURATION (inbuf);
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GST_BUFFER_OFFSET (outbuf) = resample->offset;
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GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
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if (resample->ts_offset != -1) {
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resample->offset += outsamples;
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resample->ts_offset += outsamples;
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resample->next_ts =
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GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
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GST_BUFFER_OFFSET_END (outbuf) = resample->offset;
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/* we calculate DURATION as the difference between "next" timestamp
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* and current timestamp so we ensure a contiguous stream, instead of
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* having rounding errors. */
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GST_BUFFER_DURATION (outbuf) = resample->next_ts -
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GST_BUFFER_TIMESTAMP (outbuf);
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} else {
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/* no valid offset know, we can still sortof calculate the duration though */
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GST_BUFFER_DURATION (outbuf) =
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GST_FRAMES_TO_CLOCK_TIME (outsamples, resample->outrate);
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if (GST_CLOCK_TIME_IS_VALID (timestamp)
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&& !GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
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resample->next_ts = timestamp;
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resample->next_offset =
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GST_CLOCK_TIME_TO_FRAMES (timestamp, resample->outrate);
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}
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if (G_UNLIKELY (resample->need_discont)) {
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@ -883,7 +828,19 @@ gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
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resample->need_discont = FALSE;
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}
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return gst_speex_resample_process (resample, inbuf, outbuf);
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ret = gst_speex_resample_process (resample, inbuf, outbuf);
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if (G_UNLIKELY (ret != GST_FLOW_OK))
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return ret;
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if (GST_CLOCK_TIME_IS_VALID (timestamp)
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&& !GST_CLOCK_TIME_IS_VALID (resample->next_upstream_ts))
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resample->next_upstream_ts = timestamp;
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if (GST_CLOCK_TIME_IS_VALID (resample->next_upstream_ts))
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resample->next_upstream_ts +=
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GST_FRAMES_TO_CLOCK_TIME (insamples, resample->inrate);
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return GST_FLOW_OK;
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}
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static gboolean
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@ -57,10 +57,9 @@ struct _GstSpeexResample {
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gboolean need_discont;
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guint64 offset;
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guint64 ts_offset;
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guint64 next_offset;
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GstClockTime next_ts;
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GstClockTime prev_ts, prev_duration;
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GstClockTime next_upstream_ts;
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gboolean fp;
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gint channels;
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@ -88,6 +88,7 @@ check_PROGRAMS = \
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$(check_timidity) \
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$(check_x264enc) \
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elements/selector \
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elements/speexresample \
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elements/y4menc \
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$(check_metadata)
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@ -101,3 +102,5 @@ LDADD = $(GST_OBJ_LIBS) $(GST_CHECK_LIBS) $(CHECK_LIBS)
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elements_timidity_CFLAGS = $(GST_BASE_CFLAGS) $(AM_CFLAGS)
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elements_timidity_LDADD = $(GST_BASE_LIBS) $(LDADD)
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elements_speexresample_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(AM_CFLAGS)
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elements_speexresample_LDADD = $(GST_PLUGINS_BASE_LIBS) $(LDADD) -lgstaudio-$(GST_MAJORMINOR)
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579
tests/check/elements/speexresample.c
Normal file
579
tests/check/elements/speexresample.c
Normal file
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@ -0,0 +1,579 @@
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/* GStreamer
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*
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* unit test for speexresample, based on the audioresample unit test
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
|
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
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*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <unistd.h>
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#include <gst/check/gstcheck.h>
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#include <gst/audio/audio.h>
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/* For ease of programming we use globals to keep refs for our floating
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* src and sink pads we create; otherwise we always have to do get_pad,
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* get_peer, and then remove references in every test function */
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static GstPad *mysrcpad, *mysinkpad;
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#define RESAMPLE_CAPS_TEMPLATE_STRING \
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"audio/x-raw-int, " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"signed = (bool) TRUE"
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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,
|
||||
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);
|
||||
Loading…
Reference in a new issue