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Diffstat (limited to 'tests/check/elements/audiointerleave.c')
| -rw-r--r-- | tests/check/elements/audiointerleave.c | 1128 |
1 files changed, 1128 insertions, 0 deletions
diff --git a/tests/check/elements/audiointerleave.c b/tests/check/elements/audiointerleave.c new file mode 100644 index 000000000..71348f459 --- /dev/null +++ b/tests/check/elements/audiointerleave.c @@ -0,0 +1,1128 @@ +/* GStreamer unit tests for the audiointerleave element + * Copyright (C) 2007 Tim-Philipp Müller <tim centricular net> + * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org> + * + * 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., 51 Franklin St, Fifth Floor, + * Boston, MA 02110-1301, USA. + */ + +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#ifdef HAVE_VALGRIND +# include <valgrind/valgrind.h> +#endif + +#include <gst/check/gstcheck.h> +#include <gst/audio/audio.h> +#include <gst/audio/audio-enumtypes.h> + +#include <gst/check/gstharness.h> + +static void +gst_check_setup_events_audiointerleave (GstPad * srcpad, GstElement * element, + GstCaps * caps, GstFormat format, const gchar * stream_id) +{ + GstSegment segment; + + gst_segment_init (&segment, format); + + fail_unless (gst_pad_push_event (srcpad, + gst_event_new_stream_start (stream_id))); + if (caps) + fail_unless (gst_pad_push_event (srcpad, gst_event_new_caps (caps))); + fail_unless (gst_pad_push_event (srcpad, gst_event_new_segment (&segment))); +} + +GST_START_TEST (test_create_and_unref) +{ + GstElement *interleave; + + interleave = gst_element_factory_make ("audiointerleave", NULL); + fail_unless (interleave != NULL); + + gst_element_set_state (interleave, GST_STATE_NULL); + gst_object_unref (interleave); +} + +GST_END_TEST; + +GST_START_TEST (test_request_pads) +{ + GstElement *interleave; + GstPad *pad1, *pad2; + + interleave = gst_element_factory_make ("audiointerleave", NULL); + fail_unless (interleave != NULL); + + pad1 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (pad1 != NULL); + fail_unless_equals_string (GST_OBJECT_NAME (pad1), "sink_0"); + + pad2 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (pad2 != NULL); + fail_unless_equals_string (GST_OBJECT_NAME (pad2), "sink_1"); + + gst_element_release_request_pad (interleave, pad2); + gst_object_unref (pad2); + gst_element_release_request_pad (interleave, pad1); + gst_object_unref (pad1); + + gst_element_set_state (interleave, GST_STATE_NULL); + gst_object_unref (interleave); +} + +GST_END_TEST; + +static GstPad **mysrcpads, *mysinkpad; +static GstBus *bus; +static GstElement *interleave; +static GMutex data_mutex; +static GCond data_cond; +static gint have_data; +static gfloat input[2]; + +static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " GST_AUDIO_NE (F32) ", " + "channels = (int) 2, layout = (string) {interleaved, non-interleaved}, rate = (int) 48000")); + +static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " GST_AUDIO_NE (F32) ", " + "channels = (int) 1, layout = (string) interleaved, rate = (int) 48000")); + +#define CAPS_48khz \ + "audio/x-raw, " \ + "format = (string) " GST_AUDIO_NE (F32) ", " \ + "channels = (int) 1, layout = (string) non-interleaved," \ + "rate = (int) 48000" + +static GstFlowReturn +interleave_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer) +{ + GstMapInfo map; + gfloat *outdata; + gint i; + + fail_unless (GST_IS_BUFFER (buffer)); + fail_unless (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)); + gst_buffer_map (buffer, &map, GST_MAP_READ); + outdata = (gfloat *) map.data; + fail_unless (outdata != NULL); + +#ifdef HAVE_VALGRIND + if (!(RUNNING_ON_VALGRIND)) +#endif + for (i = 0; i < map.size / sizeof (float); i += 2) { + fail_unless_equals_float (outdata[i], input[0]); + fail_unless_equals_float (outdata[i + 1], input[1]); + } + + g_mutex_lock (&data_mutex); + have_data += map.size; + g_cond_signal (&data_cond); + g_mutex_unlock (&data_mutex); + + gst_buffer_unmap (buffer, &map); + gst_buffer_unref (buffer); + + + return GST_FLOW_OK; +} + +GST_START_TEST (test_audiointerleave_2ch) +{ + GstElement *queue; + GstPad *sink0, *sink1, *src, *tmp; + GstCaps *caps; + gint i; + GstBuffer *inbuf; + gfloat *indata; + GstMapInfo map; + + mysrcpads = g_new0 (GstPad *, 2); + + have_data = 0; + + interleave = gst_element_factory_make ("audiointerleave", NULL); + fail_unless (interleave != NULL); + + g_object_set (interleave, "latency", GST_SECOND / 4, NULL); + + queue = gst_element_factory_make ("queue", "queue"); + fail_unless (queue != NULL); + + sink0 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sink0 != NULL); + fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0"); + + sink1 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sink1 != NULL); + fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1"); + + mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0"); + fail_unless (mysrcpads[0] != NULL); + + caps = gst_caps_from_string (CAPS_48khz); + gst_pad_set_active (mysrcpads[0], TRUE); + gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps, + GST_FORMAT_TIME, "0"); + gst_pad_use_fixed_caps (mysrcpads[0]); + + mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1"); + fail_unless (mysrcpads[1] != NULL); + + gst_pad_set_active (mysrcpads[1], TRUE); + gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps, + GST_FORMAT_TIME, "1"); + gst_pad_use_fixed_caps (mysrcpads[1]); + + tmp = gst_element_get_static_pad (queue, "sink"); + fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + tmp = gst_element_get_static_pad (queue, "src"); + fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK); + + mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); + fail_unless (mysinkpad != NULL); + gst_pad_set_chain_function (mysinkpad, interleave_chain_func); + gst_pad_set_active (mysinkpad, TRUE); + + src = gst_element_get_static_pad (interleave, "src"); + fail_unless (src != NULL); + fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK); + gst_object_unref (src); + + bus = gst_bus_new (); + gst_element_set_bus (interleave, bus); + + fail_unless (gst_element_set_state (interleave, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); + fail_unless (gst_element_set_state (queue, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); + + input[0] = -1.0; + inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); + //GST_BUFFER_PTS (inbuf) = 0; + gst_buffer_map (inbuf, &map, GST_MAP_WRITE); + indata = (gfloat *) map.data; + for (i = 0; i < 48000; i++) + indata[i] = -1.0; + gst_buffer_unmap (inbuf, &map); + fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK); + + input[1] = 1.0; + inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); + //GST_BUFFER_PTS (inbuf) = 0; + gst_buffer_map (inbuf, &map, GST_MAP_WRITE); + indata = (gfloat *) map.data; + for (i = 0; i < 48000; i++) + indata[i] = 1.0; + gst_buffer_unmap (inbuf, &map); + fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK); + + inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); + //GST_BUFFER_PTS (inbuf) = GST_SECOND; + gst_buffer_map (inbuf, &map, GST_MAP_WRITE); + indata = (gfloat *) map.data; + for (i = 0; i < 48000; i++) + indata[i] = -1.0; + gst_buffer_unmap (inbuf, &map); + fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK); + + inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); + //GST_BUFFER_PTS (inbuf) = GST_SECOND; + gst_buffer_map (inbuf, &map, GST_MAP_WRITE); + indata = (gfloat *) map.data; + for (i = 0; i < 48000; i++) + indata[i] = 1.0; + gst_buffer_unmap (inbuf, &map); + fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK); + + g_mutex_lock (&data_mutex); + while (have_data < 48000 * 2 * 2 * sizeof (float)) + g_cond_wait (&data_cond, &data_mutex); + g_mutex_unlock (&data_mutex); + + gst_bus_set_flushing (bus, TRUE); + gst_element_set_state (interleave, GST_STATE_NULL); + gst_element_set_state (queue, GST_STATE_NULL); + + gst_object_unref (mysrcpads[0]); + gst_object_unref (mysrcpads[1]); + gst_object_unref (mysinkpad); + + gst_element_release_request_pad (interleave, sink0); + gst_object_unref (sink0); + gst_element_release_request_pad (interleave, sink1); + gst_object_unref (sink1); + + gst_object_unref (interleave); + gst_object_unref (queue); + gst_object_unref (bus); + gst_caps_unref (caps); + + g_free (mysrcpads); +} + +GST_END_TEST; + +GST_START_TEST (test_audiointerleave_2ch_1eos) +{ + GstElement *queue; + GstPad *sink0, *sink1, *src, *tmp; + GstCaps *caps; + gint i; + GstBuffer *inbuf; + gfloat *indata; + GstMapInfo map; + + mysrcpads = g_new0 (GstPad *, 2); + + have_data = 0; + + interleave = gst_element_factory_make ("audiointerleave", NULL); + fail_unless (interleave != NULL); + + g_object_set (interleave, "latency", GST_SECOND / 4, NULL); + + queue = gst_element_factory_make ("queue", "queue"); + fail_unless (queue != NULL); + + sink0 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sink0 != NULL); + fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0"); + + sink1 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sink1 != NULL); + fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1"); + + mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0"); + fail_unless (mysrcpads[0] != NULL); + + caps = gst_caps_from_string (CAPS_48khz); + gst_pad_set_active (mysrcpads[0], TRUE); + gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps, + GST_FORMAT_TIME, "0"); + gst_pad_use_fixed_caps (mysrcpads[0]); + + mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1"); + fail_unless (mysrcpads[1] != NULL); + + gst_pad_set_active (mysrcpads[1], TRUE); + gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps, + GST_FORMAT_TIME, "1"); + gst_pad_use_fixed_caps (mysrcpads[1]); + + tmp = gst_element_get_static_pad (queue, "sink"); + fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + tmp = gst_element_get_static_pad (queue, "src"); + fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK); + + mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); + fail_unless (mysinkpad != NULL); + gst_pad_set_chain_function (mysinkpad, interleave_chain_func); + gst_pad_set_active (mysinkpad, TRUE); + + src = gst_element_get_static_pad (interleave, "src"); + fail_unless (src != NULL); + fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK); + gst_object_unref (src); + + bus = gst_bus_new (); + gst_element_set_bus (interleave, bus); + + fail_unless (gst_element_set_state (interleave, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); + fail_unless (gst_element_set_state (queue, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); + + input[0] = -1.0; + inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); + GST_BUFFER_PTS (inbuf) = 0; + gst_buffer_map (inbuf, &map, GST_MAP_WRITE); + indata = (gfloat *) map.data; + for (i = 0; i < 48000; i++) + indata[i] = -1.0; + gst_buffer_unmap (inbuf, &map); + fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK); + + input[1] = 1.0; + inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); + GST_BUFFER_PTS (inbuf) = 0; + gst_buffer_map (inbuf, &map, GST_MAP_WRITE); + indata = (gfloat *) map.data; + for (i = 0; i < 48000; i++) + indata[i] = 1.0; + gst_buffer_unmap (inbuf, &map); + fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK); + + g_mutex_lock (&data_mutex); + /* 48000 samples per buffer * 2 sources * 2 buffers */ + while (have_data != 48000 * 2 * sizeof (float)) + g_cond_wait (&data_cond, &data_mutex); + g_mutex_unlock (&data_mutex); + + input[0] = 0.0; + gst_pad_push_event (mysrcpads[0], gst_event_new_eos ()); + + input[1] = 1.0; + inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); + GST_BUFFER_PTS (inbuf) = GST_SECOND; + gst_buffer_map (inbuf, &map, GST_MAP_WRITE); + indata = (gfloat *) map.data; + for (i = 0; i < 48000; i++) + indata[i] = 1.0; + gst_buffer_unmap (inbuf, &map); + fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK); + + g_mutex_lock (&data_mutex); + /* 48000 samples per buffer * 2 sources * 2 buffers */ + while (have_data != 48000 * 2 * 2 * sizeof (float)) + g_cond_wait (&data_cond, &data_mutex); + g_mutex_unlock (&data_mutex); + + gst_bus_set_flushing (bus, TRUE); + gst_element_set_state (interleave, GST_STATE_NULL); + gst_element_set_state (queue, GST_STATE_NULL); + + gst_object_unref (mysrcpads[0]); + gst_object_unref (mysrcpads[1]); + gst_object_unref (mysinkpad); + + gst_element_release_request_pad (interleave, sink0); + gst_object_unref (sink0); + gst_element_release_request_pad (interleave, sink1); + gst_object_unref (sink1); + + gst_object_unref (interleave); + gst_object_unref (queue); + gst_object_unref (bus); + gst_caps_unref (caps); + + g_free (mysrcpads); +} + +GST_END_TEST; + +static void +src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad, + gboolean interleaved, gpointer user_data) +{ + gint n = GPOINTER_TO_INT (user_data); + gfloat *data; + gint i, num_samples; + GstCaps *caps; + guint64 mask; + GstAudioChannelPosition pos; + GstMapInfo map; + + fail_unless (gst_buffer_is_writable (buffer)); + + switch (n) { + case 0: + case 1: + case 2: + pos = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; + break; + case 3: + pos = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; + break; + default: + pos = GST_AUDIO_CHANNEL_POSITION_INVALID; + break; + } + + mask = G_GUINT64_CONSTANT (1) << pos; + + caps = gst_caps_new_simple ("audio/x-raw", + "format", G_TYPE_STRING, GST_AUDIO_NE (F32), + "channels", G_TYPE_INT, 1, + "layout", G_TYPE_STRING, interleaved ? "interleaved" : "non-interleaved", + "channel-mask", GST_TYPE_BITMASK, mask, "rate", G_TYPE_INT, 48000, NULL); + + gst_pad_set_caps (pad, caps); + gst_caps_unref (caps); + + fail_unless (gst_buffer_map (buffer, &map, GST_MAP_WRITE)); + fail_unless (map.size % sizeof (gfloat) == 0); + + fail_unless (map.size > 480); + + num_samples = map.size / sizeof (gfloat); + data = (gfloat *) map.data; + + for (i = 0; i < num_samples; i++) + data[i] = (n % 2 == 0) ? -1.0 : 1.0; + + gst_buffer_unmap (buffer, &map); +} + +static void +src_handoff_float32_audiointerleaved (GstElement * element, GstBuffer * buffer, + GstPad * pad, gpointer user_data) +{ + src_handoff_float32 (element, buffer, pad, TRUE, user_data); +} + +static void +src_handoff_float32_non_audiointerleaved (GstElement * element, + GstBuffer * buffer, GstPad * pad, gpointer user_data) +{ + src_handoff_float32 (element, buffer, pad, FALSE, user_data); +} + +static void +sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad, + gpointer user_data) +{ + gint i; + GstMapInfo map; + gfloat *data; + GstCaps *caps, *ccaps; + gint n = GPOINTER_TO_INT (user_data); + guint64 mask; + + fail_unless (GST_IS_BUFFER (buffer)); + gst_buffer_map (buffer, &map, GST_MAP_READ); + data = (gfloat *) map.data; + + /* Give a little leeway for rounding errors */ + fail_unless (gst_util_uint64_scale (map.size, GST_SECOND, + 48000 * 2 * sizeof (gfloat)) <= GST_BUFFER_DURATION (buffer) + 1 || + gst_util_uint64_scale (map.size, GST_SECOND, + 48000 * 2 * sizeof (gfloat)) >= GST_BUFFER_DURATION (buffer) - 1); + + if (n == 0 || n == 3) { + GstAudioChannelPosition pos[2] = + { GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE }; + gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask); + } else if (n == 1) { + GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, + GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT + }; + gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask); + } else if (n == 2) { + GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, + GST_AUDIO_CHANNEL_POSITION_REAR_CENTER + }; + gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask); + } else { + g_assert_not_reached (); + } + + if (pad) { + caps = gst_caps_new_simple ("audio/x-raw", + "format", G_TYPE_STRING, GST_AUDIO_NE (F32), + "channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000, + "layout", G_TYPE_STRING, "interleaved", + "channel-mask", GST_TYPE_BITMASK, mask, NULL); + + ccaps = gst_pad_get_current_caps (pad); + fail_unless (gst_caps_is_equal (caps, ccaps)); + gst_caps_unref (ccaps); + gst_caps_unref (caps); + } +#ifdef HAVE_VALGRIND + if (!(RUNNING_ON_VALGRIND)) +#endif + for (i = 0; i < map.size / sizeof (float); i += 2) { + fail_unless_equals_float (data[i], -1.0); + if (n != 3) + fail_unless_equals_float (data[i + 1], 1.0); + } + have_data += map.size; + + gst_buffer_unmap (buffer, &map); + +} + +static void +test_audiointerleave_2ch_pipeline (gboolean interleaved) +{ + GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink; + GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2; + GstMessage *msg; + void *src_handoff_float32 = + interleaved ? &src_handoff_float32_audiointerleaved : + &src_handoff_float32_non_audiointerleaved; + + have_data = 0; + + pipeline = (GstElement *) gst_pipeline_new ("pipeline"); + fail_unless (pipeline != NULL); + + src1 = gst_element_factory_make ("fakesrc", "src1"); + fail_unless (src1 != NULL); + g_object_set (src1, "num-buffers", 4, NULL); + g_object_set (src1, "sizetype", 2, + "sizemax", (int) 48000 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_object_set (src1, "signal-handoffs", TRUE, NULL); + g_object_set (src1, "format", GST_FORMAT_TIME, NULL); + g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32), + GINT_TO_POINTER (0)); + gst_bin_add (GST_BIN (pipeline), src1); + + src2 = gst_element_factory_make ("fakesrc", "src2"); + fail_unless (src2 != NULL); + g_object_set (src2, "num-buffers", 4, NULL); + g_object_set (src2, "sizetype", 2, + "sizemax", (int) 48000 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_object_set (src2, "signal-handoffs", TRUE, NULL); + g_object_set (src2, "format", GST_FORMAT_TIME, NULL); + g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32), + GINT_TO_POINTER (1)); + gst_bin_add (GST_BIN (pipeline), src2); + + queue = gst_element_factory_make ("queue", "queue"); + fail_unless (queue != NULL); + gst_bin_add (GST_BIN (pipeline), queue); + + interleave = gst_element_factory_make ("audiointerleave", "audiointerleave"); + fail_unless (interleave != NULL); + gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave)); + + sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sinkpad0 != NULL); + tmp = gst_element_get_static_pad (src1, "src"); + fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sinkpad1 != NULL); + tmp = gst_element_get_static_pad (src2, "src"); + tmp2 = gst_element_get_static_pad (queue, "sink"); + fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + gst_object_unref (tmp2); + tmp = gst_element_get_static_pad (queue, "src"); + fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + sink = gst_element_factory_make ("fakesink", "sink"); + fail_unless (sink != NULL); + g_object_set (sink, "signal-handoffs", TRUE, NULL); + g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32), + GINT_TO_POINTER (0)); + gst_bin_add (GST_BIN (pipeline), sink); + tmp = gst_element_get_static_pad (interleave, "src"); + tmp2 = gst_element_get_static_pad (sink, "sink"); + fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + gst_object_unref (tmp2); + + gst_element_set_state (pipeline, GST_STATE_PLAYING); + + msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); + gst_message_unref (msg); + + /* 48000 samples per buffer * 2 sources * 4 buffers */ + fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat)); + + gst_element_set_state (pipeline, GST_STATE_NULL); + gst_element_release_request_pad (interleave, sinkpad0); + gst_object_unref (sinkpad0); + gst_element_release_request_pad (interleave, sinkpad1); + gst_object_unref (sinkpad1); + gst_object_unref (interleave); + gst_object_unref (pipeline); +} + +GST_START_TEST (test_audiointerleave_2ch_pipeline_audiointerleaved) +{ + test_audiointerleave_2ch_pipeline (TRUE); +} + +GST_END_TEST; + +GST_START_TEST (test_audiointerleave_2ch_pipeline_non_audiointerleaved) +{ + test_audiointerleave_2ch_pipeline (FALSE); +} + +GST_END_TEST; + +GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos) +{ + GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink; + GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2; + GstMessage *msg; + + have_data = 0; + + pipeline = (GstElement *) gst_pipeline_new ("pipeline"); + fail_unless (pipeline != NULL); + + src1 = gst_element_factory_make ("fakesrc", "src1"); + fail_unless (src1 != NULL); + g_object_set (src1, "num-buffers", 4, NULL); + g_object_set (src1, "sizetype", 2, + "sizemax", (int) 48000 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_object_set (src1, "signal-handoffs", TRUE, NULL); + g_object_set (src1, "format", GST_FORMAT_TIME, NULL); + g_signal_connect (src1, "handoff", + G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2)); + gst_bin_add (GST_BIN (pipeline), src1); + + src2 = gst_element_factory_make ("fakesrc", "src2"); + fail_unless (src2 != NULL); + g_object_set (src2, "num-buffers", 4, NULL); + g_object_set (src2, "sizetype", 2, + "sizemax", (int) 48000 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_object_set (src2, "signal-handoffs", TRUE, NULL); + g_object_set (src2, "format", GST_FORMAT_TIME, NULL); + g_signal_connect (src2, "handoff", + G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (3)); + gst_bin_add (GST_BIN (pipeline), src2); + + queue = gst_element_factory_make ("queue", "queue"); + fail_unless (queue != NULL); + gst_bin_add (GST_BIN (pipeline), queue); + + interleave = gst_element_factory_make ("audiointerleave", "audiointerleave"); + fail_unless (interleave != NULL); + g_object_set (interleave, "channel-positions-from-input", TRUE, NULL); + gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave)); + + sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sinkpad0 != NULL); + tmp = gst_element_get_static_pad (src1, "src"); + fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sinkpad1 != NULL); + tmp = gst_element_get_static_pad (src2, "src"); + tmp2 = gst_element_get_static_pad (queue, "sink"); + fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + gst_object_unref (tmp2); + tmp = gst_element_get_static_pad (queue, "src"); + fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + sink = gst_element_factory_make ("fakesink", "sink"); + fail_unless (sink != NULL); + g_object_set (sink, "signal-handoffs", TRUE, NULL); + g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32), + GINT_TO_POINTER (1)); + gst_bin_add (GST_BIN (pipeline), sink); + tmp = gst_element_get_static_pad (interleave, "src"); + tmp2 = gst_element_get_static_pad (sink, "sink"); + fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + gst_object_unref (tmp2); + + gst_element_set_state (pipeline, GST_STATE_PLAYING); + + msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); + gst_message_unref (msg); + + /* 48000 samples per buffer * 2 sources * 4 buffers */ + fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat)); + + gst_element_set_state (pipeline, GST_STATE_NULL); + gst_element_release_request_pad (interleave, sinkpad0); + gst_object_unref (sinkpad0); + gst_element_release_request_pad (interleave, sinkpad1); + gst_object_unref (sinkpad1); + gst_object_unref (interleave); + gst_object_unref (pipeline); +} + +GST_END_TEST; + +GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos) +{ + GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink; + GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2; + GstMessage *msg; + GValueArray *arr; + GValue val = { 0, }; + + have_data = 0; + + pipeline = (GstElement *) gst_pipeline_new ("pipeline"); + fail_unless (pipeline != NULL); + + src1 = gst_element_factory_make ("fakesrc", "src1"); + fail_unless (src1 != NULL); + g_object_set (src1, "num-buffers", 4, NULL); + g_object_set (src1, "signal-handoffs", TRUE, NULL); + g_object_set (src1, "sizetype", 2, + "sizemax", (int) 48000 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_object_set (src1, "format", GST_FORMAT_TIME, NULL); + g_signal_connect (src1, "handoff", + G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0)); + gst_bin_add (GST_BIN (pipeline), src1); + + src2 = gst_element_factory_make ("fakesrc", "src2"); + fail_unless (src2 != NULL); + g_object_set (src2, "num-buffers", 4, NULL); + g_object_set (src2, "signal-handoffs", TRUE, NULL); + g_object_set (src2, "sizetype", 2, + "sizemax", (int) 48000 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_object_set (src2, "format", GST_FORMAT_TIME, NULL); + g_signal_connect (src2, "handoff", + G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1)); + gst_bin_add (GST_BIN (pipeline), src2); + + queue = gst_element_factory_make ("queue", "queue"); + fail_unless (queue != NULL); + gst_bin_add (GST_BIN (pipeline), queue); + + interleave = gst_element_factory_make ("audiointerleave", "audiointerleave"); + fail_unless (interleave != NULL); + g_object_set (interleave, "channel-positions-from-input", FALSE, NULL); + arr = g_value_array_new (2); + + g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION); + g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER); + g_value_array_append (arr, &val); + g_value_reset (&val); + g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER); + g_value_array_append (arr, &val); + g_value_unset (&val); + + g_object_set (interleave, "channel-positions", arr, NULL); + g_value_array_free (arr); + gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave)); + + sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sinkpad0 != NULL); + tmp = gst_element_get_static_pad (src1, "src"); + fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sinkpad1 != NULL); + tmp = gst_element_get_static_pad (src2, "src"); + tmp2 = gst_element_get_static_pad (queue, "sink"); + fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + gst_object_unref (tmp2); + tmp = gst_element_get_static_pad (queue, "src"); + fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + sink = gst_element_factory_make ("fakesink", "sink"); + fail_unless (sink != NULL); + g_object_set (sink, "signal-handoffs", TRUE, NULL); + g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32), + GINT_TO_POINTER (2)); + gst_bin_add (GST_BIN (pipeline), sink); + tmp = gst_element_get_static_pad (interleave, "src"); + tmp2 = gst_element_get_static_pad (sink, "sink"); + fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + gst_object_unref (tmp2); + + gst_element_set_state (pipeline, GST_STATE_PLAYING); + + msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); + gst_message_unref (msg); + + /* 48000 samples per buffer * 2 sources * 4 buffers */ + fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat)); + + gst_element_set_state (pipeline, GST_STATE_NULL); + gst_element_release_request_pad (interleave, sinkpad0); + gst_object_unref (sinkpad0); + gst_element_release_request_pad (interleave, sinkpad1); + gst_object_unref (sinkpad1); + gst_object_unref (interleave); + gst_object_unref (pipeline); +} + +GST_END_TEST; + +GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos) +{ + GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink; + GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2; + GstMessage *msg; + + have_data = 0; + + pipeline = (GstElement *) gst_pipeline_new ("pipeline"); + fail_unless (pipeline != NULL); + + src1 = gst_element_factory_make ("fakesrc", "src1"); + fail_unless (src1 != NULL); + g_object_set (src1, "num-buffers", 4, NULL); + g_object_set (src1, "signal-handoffs", TRUE, NULL); + g_object_set (src1, "sizetype", 2, + "sizemax", (int) 48000 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_object_set (src1, "format", GST_FORMAT_TIME, NULL); + g_signal_connect (src1, "handoff", + G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0)); + gst_bin_add (GST_BIN (pipeline), src1); + + src2 = gst_element_factory_make ("fakesrc", "src2"); + fail_unless (src2 != NULL); + g_object_set (src2, "num-buffers", 4, NULL); + g_object_set (src2, "signal-handoffs", TRUE, NULL); + g_object_set (src2, "sizetype", 2, + "sizemax", (int) 48000 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_object_set (src2, "format", GST_FORMAT_TIME, NULL); + g_signal_connect (src2, "handoff", + G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1)); + gst_bin_add (GST_BIN (pipeline), src2); + + queue = gst_element_factory_make ("queue", "queue"); + fail_unless (queue != NULL); + gst_bin_add (GST_BIN (pipeline), queue); + + interleave = gst_element_factory_make ("audiointerleave", "audiointerleave"); + fail_unless (interleave != NULL); + g_object_set (interleave, "channel-positions-from-input", FALSE, NULL); + gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave)); + + sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sinkpad0 != NULL); + tmp = gst_element_get_static_pad (src1, "src"); + fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u"); + fail_unless (sinkpad1 != NULL); + tmp = gst_element_get_static_pad (src2, "src"); + tmp2 = gst_element_get_static_pad (queue, "sink"); + fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + gst_object_unref (tmp2); + tmp = gst_element_get_static_pad (queue, "src"); + fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + + sink = gst_element_factory_make ("fakesink", "sink"); + fail_unless (sink != NULL); + g_object_set (sink, "signal-handoffs", TRUE, NULL); + g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32), + GINT_TO_POINTER (0)); + gst_bin_add (GST_BIN (pipeline), sink); + tmp = gst_element_get_static_pad (interleave, "src"); + tmp2 = gst_element_get_static_pad (sink, "sink"); + fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); + gst_object_unref (tmp); + gst_object_unref (tmp2); + + gst_element_set_state (pipeline, GST_STATE_PLAYING); + + msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); + gst_message_unref (msg); + + /* 48000 samples per buffer * 2 sources * 4 buffers */ + fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat)); + + gst_element_set_state (pipeline, GST_STATE_NULL); + gst_element_release_request_pad (interleave, sinkpad0); + gst_object_unref (sinkpad0); + gst_element_release_request_pad (interleave, sinkpad1); + gst_object_unref (sinkpad1); + gst_object_unref (interleave); + gst_object_unref (pipeline); +} + +GST_END_TEST; + +static void +forward_check_event (GstHarness * h, GstHarness * hsrc, GstEventType type) +{ + GstEvent *e; + + e = gst_harness_pull_event (hsrc); + fail_unless (GST_EVENT_TYPE (e) == type); + gst_harness_push_event (h, e); +} + +GST_START_TEST (test_audiointerleave_2ch_smallbuf) +{ + GstElement *audiointerleave; + GstHarness *hsrc; + GstHarness *h; + GstHarness *h2; + GstBuffer *buffer; + gint i; + GstEvent *ev; + GstCaps *ecaps, *caps; + + audiointerleave = gst_element_factory_make ("audiointerleave", NULL); + + g_object_set (audiointerleave, "latency", GST_SECOND / 2, + "output-buffer-duration", GST_SECOND / 4, NULL); + + h = gst_harness_new_with_element (audiointerleave, "sink_0", "src"); + gst_harness_use_testclock (h); + + h2 = gst_harness_new_with_element (audiointerleave, "sink_1", NULL); + gst_harness_set_src_caps_str (h2, "audio/x-raw, " + "format=" GST_AUDIO_NE (F32) ", channels=(int)1," + " layout=interleaved, rate=48000, channel-mask=(bitmask)8"); + + hsrc = gst_harness_new ("fakesrc"); + gst_harness_use_testclock (hsrc); + g_object_set (hsrc->element, + "is-live", TRUE, + "sync", TRUE, + "signal-handoffs", TRUE, + "format", GST_FORMAT_TIME, + "sizetype", 2, + "sizemax", (int) 480 * sizeof (gfloat), + "datarate", (int) 48000 * sizeof (gfloat), NULL); + g_signal_connect (hsrc->element, "handoff", + G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2)); + gst_harness_play (hsrc); + + gst_harness_crank_single_clock_wait (hsrc); + forward_check_event (h, hsrc, GST_EVENT_STREAM_START); + forward_check_event (h, hsrc, GST_EVENT_CAPS); + forward_check_event (h, hsrc, GST_EVENT_SEGMENT); + gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */ + + for (i = 0; i < 24; i++) { + gst_harness_crank_single_clock_wait (hsrc); + forward_check_event (h, hsrc, GST_EVENT_CAPS); + gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */ + } + + gst_harness_crank_single_clock_wait (h); + + + gst_event_unref (gst_harness_pull_event (h)); /* stream-start */ + ev = gst_harness_pull_event (h); /* caps */ + fail_unless_equals_int (GST_EVENT_CAPS, GST_EVENT_TYPE (ev)); + + caps = gst_caps_new_simple ("audio/x-raw", + "format", G_TYPE_STRING, GST_AUDIO_NE (F32), + "channels", G_TYPE_INT, 2, + "layout", G_TYPE_STRING, "interleaved", + "rate", G_TYPE_INT, 48000, "channel-mask", GST_TYPE_BITMASK, + (guint64) 0x9, NULL); + + gst_event_parse_caps (ev, &ecaps); + gst_check_caps_equal (ecaps, caps); + gst_caps_unref (caps); + gst_event_unref (ev); + + /* eat the caps processing */ + gst_harness_crank_single_clock_wait (h); + for (i = 0; i < 23; i++) + gst_harness_crank_single_clock_wait (h); + fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK + (h->element)), 750 * GST_MSECOND); + + buffer = gst_harness_pull (h); + sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); + gst_buffer_unref (buffer); + fail_unless_equals_int (gst_harness_buffers_received (h), 1); + + for (i = 0; i < 50; i++) { + gst_harness_crank_single_clock_wait (hsrc); + forward_check_event (h, hsrc, GST_EVENT_CAPS); + gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */ + } + for (i = 0; i < 25; i++) + gst_harness_crank_single_clock_wait (h); + fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK + (h->element)), 1000 * GST_MSECOND); + buffer = gst_harness_pull (h); + sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); + gst_buffer_unref (buffer); + fail_unless_equals_int (gst_harness_buffers_received (h), 2); + + for (i = 0; i < 25; i++) { + gst_harness_crank_single_clock_wait (hsrc); + forward_check_event (h, hsrc, GST_EVENT_CAPS); + gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */ + } + for (i = 0; i < 25; i++) + gst_harness_crank_single_clock_wait (h); + fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK + (h->element)), 1250 * GST_MSECOND); + buffer = gst_harness_pull (h); + sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); + gst_buffer_unref (buffer); + fail_unless_equals_int (gst_harness_buffers_received (h), 3); + + gst_harness_push_event (h, gst_event_new_eos ()); + + for (i = 0; i < 25; i++) + gst_harness_crank_single_clock_wait (h); + fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK + (h->element)), 1500 * GST_MSECOND); + buffer = gst_harness_pull (h); + sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); + gst_buffer_unref (buffer); + + fail_unless_equals_int (gst_harness_buffers_received (h), 4); + + gst_harness_teardown (h2); + gst_harness_teardown (h); + gst_harness_teardown (hsrc); + gst_object_unref (audiointerleave); +} + +GST_END_TEST; + +static Suite * +audiointerleave_suite (void) +{ + Suite *s = suite_create ("audiointerleave"); + TCase *tc_chain = tcase_create ("general"); + + suite_add_tcase (s, tc_chain); + tcase_set_timeout (tc_chain, 180); + tcase_add_test (tc_chain, test_create_and_unref); + tcase_add_test (tc_chain, test_request_pads); + tcase_add_test (tc_chain, test_audiointerleave_2ch); + tcase_add_test (tc_chain, test_audiointerleave_2ch_1eos); + tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_audiointerleaved); + tcase_add_test (tc_chain, + test_audiointerleave_2ch_pipeline_non_audiointerleaved); + tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_input_chanpos); + tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_custom_chanpos); + tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_no_chanpos); + tcase_add_test (tc_chain, test_audiointerleave_2ch_smallbuf); + + return s; +} + +GST_CHECK_MAIN (audiointerleave); |