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audioresample: Fix up test_live_switch
Actually check that we get back all samples, which we didn't before because no draining was happening. Also remove commented out 0.10 code and related comments. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/670>
This commit is contained in:
Sebastian Dröge
GStreamer Merge Bot
parent
44cd1c7a65
commit
71c937b565
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@ -443,106 +443,49 @@ GST_START_TEST (test_shutdown)
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GST_END_TEST;
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#if 0
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static GstFlowReturn
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live_switch_alloc_only_48000 (GstPad * pad, guint64 offset,
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guint size, GstCaps * caps, GstBuffer ** buf)
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{
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GstStructure *structure;
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gint rate;
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gint channels;
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GstCaps *desired;
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structure = gst_caps_get_structure (caps, 0);
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fail_unless (gst_structure_get_int (structure, "rate", &rate));
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fail_unless (gst_structure_get_int (structure, "channels", &channels));
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if (rate < 48000)
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return GST_FLOW_NOT_NEGOTIATED;
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desired = gst_caps_copy (caps);
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gst_caps_set_simple (desired, "rate", G_TYPE_INT, 48000, NULL);
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*buf = gst_buffer_new_and_alloc (channels * 48000);
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gst_buffer_set_caps (*buf, desired);
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gst_caps_unref (desired);
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return GST_FLOW_OK;
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}
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static GstCaps *
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live_switch_get_sink_caps (GstPad * pad)
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{
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GstCaps *result;
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result = gst_caps_make_writable (gst_pad_get_current_caps (pad));
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gst_caps_set_simple (result,
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"rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
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return result;
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}
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#endif
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static void
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live_switch_push (int rate, GstCaps * caps)
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live_switch_push (gint pts, gint rate, GstCaps * caps)
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{
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GstBuffer *inbuffer;
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GstCaps *desired;
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GList *l;
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desired = gst_caps_copy (caps);
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gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
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gst_pad_set_caps (mysrcpad, desired);
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#if 0
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fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
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GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
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#endif
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inbuffer = gst_buffer_new_and_alloc (rate * 4);
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gst_buffer_memset (inbuffer, 0, 0, rate * 4);
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inbuffer = gst_buffer_new_and_alloc (rate * 4 * 2);
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gst_buffer_memset (inbuffer, 0, 0, rate * 4 * 2);
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GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND;
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GST_BUFFER_OFFSET (inbuffer) = 0;
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GST_BUFFER_OFFSET_END (inbuffer) = rate - 1;
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
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/* ... but it ends up being collected on the global buffer list */
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fail_unless_equals_int (g_list_length (buffers), 1);
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for (l = buffers; l; l = l->next) {
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GstBuffer *buffer = GST_BUFFER (l->data);
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gst_buffer_unref (buffer);
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}
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g_list_free (buffers);
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buffers = NULL;
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gst_caps_unref (desired);
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}
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#if !GLIB_CHECK_VERSION(2,58,0)
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#define G_APPROX_VALUE(a, b, epsilon) \
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(((a) > (b) ? (a) - (b) : (b) - (a)) < (epsilon))
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#endif
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GST_START_TEST (test_live_switch)
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{
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GstElement *audioresample;
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GstEvent *newseg;
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GstCaps *caps;
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GstSegment segment;
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GList *l;
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guint i;
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audioresample =
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setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16));
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/* Let the sinkpad act like something that can only handle things of
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* rate 48000- and can only allocate buffers for that rate, but if someone
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* tries to get a buffer with a rate higher then 48000 tries to renegotiate
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* */
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//gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
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//gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
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gst_pad_use_fixed_caps (mysrcpad);
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caps = gst_pad_get_current_caps (mysrcpad);
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fail_unless (gst_caps_is_fixed (caps));
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@ -554,15 +497,72 @@ GST_START_TEST (test_live_switch)
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newseg = gst_event_new_segment (&segment);
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fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
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/* downstream can provide the requested rate, a buffer alloc will be passed
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* on */
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live_switch_push (48000, caps);
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/* downstream can accept the requested rate */
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live_switch_push (0, 48000, caps);
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/* Downstream can never accept this rate, buffer alloc isn't passed on */
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live_switch_push (40000, caps);
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/* buffer is directly passed through */
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fail_unless_equals_int (g_list_length (buffers), 1);
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/* Downstream can provide the requested rate but will re-negotiate */
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live_switch_push (50000, caps);
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/* Downstream can never accept this rate */
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live_switch_push (1, 40000, caps);
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/* one additional buffer is provided with the new sample rate */
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fail_unless_equals_int (g_list_length (buffers), 2);
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/* Downstream can never accept this rate */
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live_switch_push (2, 50000, caps);
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/* two additional buffers are provided. One is the drained remainder of
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* the previous sample rate, the second is the buffer with the new sample
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* rate */
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fail_unless_equals_int (g_list_length (buffers), 4);
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/* Send EOS to drain the remaining samples */
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
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fail_unless_equals_int (g_list_length (buffers), 5);
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/* Now test that each buffer has the expected samples. We simply check this
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* by checking whether the timestamps, durations and sizes are matching */
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for (l = buffers, i = 0; l; l = l->next, i++) {
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GstBuffer *buffer = GST_BUFFER (l->data);
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switch (i) {
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case 0:
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fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND);
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fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
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1 * GST_SECOND);
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fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2);
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break;
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case 1:
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fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND);
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fail_unless_equals_int (gst_buffer_get_size (buffer), 47961 * 4 * 2);
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break;
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case 2:
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fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
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GST_BUFFER_DURATION (buffer), 2 * GST_SECOND,
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GST_SECOND / 48000 + 1));
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fail_unless_equals_int (gst_buffer_get_size (buffer), 38 * 4 * 2);
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break;
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case 3:
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fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND);
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fail_unless_equals_int (gst_buffer_get_size (buffer), 47969 * 4 * 2);
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break;
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case 4:
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fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
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GST_BUFFER_DURATION (buffer), 3 * GST_SECOND,
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GST_SECOND / 48000 + 1));
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fail_unless_equals_int (gst_buffer_get_size (buffer), 30 * 4 * 2);
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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gst_buffer_unref (buffer);
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}
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g_list_free (buffers);
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buffers = NULL;
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cleanup_audioresample (audioresample);
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gst_caps_unref (caps);
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