/* GStreamer * * Copyright (C) 2020 Seungha Yang * * 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. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include GST_START_TEST (test_audioenc_drain) { GstHarness *h; GstAudioInfo info; GstBuffer *in_buf; gint i = 0; gint num_output = 0; GstFlowReturn ret; GstSegment segment; GstCaps *caps; gint samples_per_buffer = 1024; gint rate = 44100; gint size; GstClockTime duration; h = gst_harness_new ("avenc_aac"); fail_unless (h != NULL); gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_F32, rate, 1, NULL); caps = gst_audio_info_to_caps (&info); gst_harness_set_src_caps (h, gst_caps_copy (caps)); duration = gst_util_uint64_scale_int (samples_per_buffer, GST_SECOND, rate); size = samples_per_buffer * GST_AUDIO_INFO_BPF (&info); for (i = 0; i < 2; i++) { in_buf = gst_buffer_new_and_alloc (size); gst_buffer_memset (in_buf, 0, 0, size); /* small rounding error would be expected, but should be fine */ GST_BUFFER_PTS (in_buf) = i * duration; GST_BUFFER_DURATION (in_buf) = duration; ret = gst_harness_push (h, in_buf); fail_unless (ret == GST_FLOW_OK, "GstFlowReturn was %s", gst_flow_get_name (ret)); } gst_segment_init (&segment, GST_FORMAT_TIME); fail_unless (gst_segment_set_running_time (&segment, GST_FORMAT_TIME, 2 * duration)); /* Push new eos event to drain encoder */ fail_unless (gst_harness_push_event (h, gst_event_new_eos ())); /* And start new stream */ fail_unless (gst_harness_push_event (h, gst_event_new_stream_start ("new-stream-id"))); gst_harness_set_src_caps (h, caps); fail_unless (gst_harness_push_event (h, gst_event_new_segment (&segment))); in_buf = gst_buffer_new_and_alloc (size); GST_BUFFER_PTS (in_buf) = 2 * duration; GST_BUFFER_DURATION (in_buf) = duration; ret = gst_harness_push (h, in_buf); fail_unless (ret == GST_FLOW_OK, "GstFlowReturn was %s", gst_flow_get_name (ret)); /* Finish encoding and drain again */ fail_unless (gst_harness_push_event (h, gst_event_new_eos ())); do { GstBuffer *out_buf = NULL; out_buf = gst_harness_try_pull (h); if (out_buf) { num_output++; gst_buffer_unref (out_buf); continue; } break; } while (1); fail_unless (num_output >= 3); gst_harness_teardown (h); } GST_END_TEST; GST_START_TEST (test_audioenc_16_channels) { /* avaudenc used to have a bug for >8ch where a double-free attempt would occur, * crashing the whole process. Since >8ch encoding is quite rarely used, this test * is meant to detect any crashes that would indicate somebody broke that again */ GstHarness *h; GstAudioInfo info; GstBuffer *in_buf; GstCaps *caps; gint size; GstAudioChannelPosition position[16]; /* 16ch hexadecagonal layout */ guint64 channel_mask = 0x3137D37; h = gst_harness_new ("avenc_aac"); fail_unless (h != NULL); gst_audio_channel_positions_from_mask (16, channel_mask, position); gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_F32, 44100, 16, position); caps = gst_audio_info_to_caps (&info); gst_harness_set_src_caps (h, caps); size = 1024 * GST_AUDIO_INFO_BPF (&info); in_buf = gst_buffer_new_and_alloc (size); gst_buffer_memset (in_buf, 0, 0, size); GST_BUFFER_PTS (in_buf) = 0; GstFlowReturn ret = gst_harness_push (h, in_buf); fail_if (ret != GST_FLOW_OK); gst_harness_teardown (h); } GST_END_TEST; // Make sure we fix up any too-small memory alignment before feeding data // to FFmpeg. By default we use the malloc alignment, which might be 16, // but FFmpeg might be using SIMD operations that require a bigger alignment. GST_START_TEST (test_audioenc_alignment_fixup) { GstHarness *h; GstAudioInfo info; GstCaps *caps; h = gst_harness_new ("avenc_ac3"); fail_unless (h != NULL); gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_F32, 44100, 1, NULL); caps = gst_audio_info_to_caps (&info); gst_harness_set_src_caps (h, caps); fail_unless_equals_int (GST_AUDIO_INFO_BPF (&info), sizeof (float)); // AC-3 has 1536 samples per frame. Need to supply that many per buffer, // otherwise the audio encoder baseclass will realloc things via GstAdapter // and mess up our carefully curated audio buffer (mis)alignment. # define N_SAMPLES 1536 # define N_ALIGNMENTS 16 const gsize size = N_SAMPLES * sizeof (float); float *samples = g_new0 (float, (N_SAMPLES + N_ALIGNMENTS)); guint64 offset = 0; for (int i = 0; i < 100; ++i) { GstMemory *mem = gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, samples + (i % N_ALIGNMENTS), size, 0, size, NULL, NULL); GstBuffer *in_buf = gst_buffer_new (); gst_buffer_insert_memory (in_buf, 0, mem); GST_BUFFER_PTS (in_buf) = gst_util_uint64_scale (offset, GST_SECOND, 44100); GstFlowReturn ret = gst_harness_push (h, g_steal_pointer (&in_buf)); fail_unless_equals_int (ret, GST_FLOW_OK); offset += N_SAMPLES; } g_free (samples); gst_harness_teardown (h); } GST_END_TEST; static Suite * avaudenc_suite (void) { Suite *s = suite_create ("avaudenc"); TCase *tc_chain = tcase_create ("general"); suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_audioenc_drain); tcase_add_test (tc_chain, test_audioenc_16_channels); tcase_add_test (tc_chain, test_audioenc_alignment_fixup); return s; } GST_CHECK_MAIN (avaudenc)