mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-08-02 11:45:07 +00:00
Merge remote-tracking branch 'origin/master' into 0.11
Conflicts: ext/opus/gstopusdec.c ext/opus/gstopusparse.c gst-libs/gst/video/gstbasevideodecoder.c gst-libs/gst/video/gstbasevideodecoder.h
This commit is contained in:
Tim-Philipp Müller
commit
77361e2919
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@ -70,3 +70,19 @@ const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE},
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};
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const char *gst_opus_channel_names[] = {
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"mono",
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"front left",
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"front right",
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"rear center",
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"rear left",
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"rear right",
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"lfe",
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"front center",
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"front left of center",
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"front right of center",
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"side left",
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"side right",
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"none"
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};
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@ -27,6 +27,7 @@
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G_BEGIN_DECLS
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extern const GstAudioChannelPosition gst_opus_channel_positions[][8];
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extern const char *gst_opus_channel_names[];
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G_END_DECLS
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@ -219,7 +219,8 @@ gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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data = gst_buffer_map (buf, NULL, NULL, GST_MAP_READ);
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g_return_val_if_fail (dec->n_channels != data[9], GST_FLOW_ERROR);
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g_return_val_if_fail (dec->n_channels == 0
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|| dec->n_channels == data[9], GST_FLOW_ERROR);
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dec->n_channels = data[9];
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dec->pre_skip = GST_READ_UINT16_LE (data + 10);
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@ -288,6 +289,7 @@ gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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dec->sample_rate);
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if (pos) {
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GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
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gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
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}
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@ -187,11 +187,9 @@ static void
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gst_opus_enc_class_init (GstOpusEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioEncoderClass *base_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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base_class = (GstAudioEncoderClass *) klass;
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gobject_class->set_property = gst_opus_enc_set_property;
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@ -446,7 +444,8 @@ gst_opus_enc_setup_channel_mapping (GstOpusEnc * enc, const GstAudioInfo * info)
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GstAudioChannelPosition pos = GST_AUDIO_INFO_POSITION (info, n);
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int c;
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GST_DEBUG_OBJECT (enc, "Channel %d has position %d", n, pos);
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GST_DEBUG_OBJECT (enc, "Channel %d has position %d (%s)", n, pos,
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gst_opus_channel_names[pos]);
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for (c = 0; c < enc->n_channels; ++c) {
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if (gst_opus_channel_positions[enc->n_channels - 1][c] == pos) {
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GST_DEBUG_OBJECT (enc, "Found in Vorbis mapping as channel %d", c);
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@ -456,12 +455,13 @@ gst_opus_enc_setup_channel_mapping (GstOpusEnc * enc, const GstAudioInfo * info)
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if (c == enc->n_channels) {
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/* We did not find that position, so use undefined */
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GST_WARNING_OBJECT (enc,
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"Position %d not found in Vorbis mapping, using unknown mapping",
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pos);
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"Position %d (%s) not found in Vorbis mapping, using unknown mapping",
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pos, gst_opus_channel_positions[pos]);
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enc->channel_mapping_family = 255;
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return;
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}
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GST_DEBUG_OBJECT (enc, "Mapping output channel %d to %d", c, n);
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GST_DEBUG_OBJECT (enc, "Mapping output channel %d to %d (%s)", c, n,
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gst_opus_channel_names[pos]);
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enc->channel_mapping[c] = n;
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}
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GST_INFO_OBJECT (enc, "Permutation found, using Vorbis mapping");
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@ -512,13 +512,17 @@ gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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static gboolean
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gst_opus_enc_setup (GstOpusEnc * enc)
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{
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int error = OPUS_OK;
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int error = OPUS_OK, n;
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guint8 trivial_mapping[256];
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GST_DEBUG_OBJECT (enc, "setup");
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for (n = 0; n < 256; ++n)
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trivial_mapping[n] = n;
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enc->state =
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opus_multistream_encoder_create (enc->sample_rate, enc->n_channels,
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(enc->n_channels + 1) / 2, enc->n_channels / 2, enc->channel_mapping,
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enc->n_channels, 0, trivial_mapping,
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enc->audio_or_voip ? OPUS_APPLICATION_AUDIO : OPUS_APPLICATION_VOIP,
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&error);
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if (!enc->state || error != OPUS_OK)
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@ -44,8 +44,8 @@ gst_opus_enc_create_id_buffer (gint nchannels, gint sample_rate,
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gst_byte_writer_put_uint16_le (&bw, 0); /* output gain */
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gst_byte_writer_put_uint8 (&bw, channel_mapping_family);
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if (channel_mapping_family > 0) {
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gst_byte_writer_put_uint8 (&bw, (nchannels + 1) / 2);
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gst_byte_writer_put_uint8 (&bw, nchannels / 2);
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gst_byte_writer_put_uint8 (&bw, nchannels);
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gst_byte_writer_put_uint8 (&bw, 0);
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gst_byte_writer_put_data (&bw, channel_mapping, nchannels);
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}
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@ -286,20 +286,11 @@ gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
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if (!parse->header_sent) {
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GstCaps *caps;
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guint8 channels, channel_mapping_family, channel_mapping[256];
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data = gst_buffer_map (frame->buffer, &size, NULL, GST_MAP_READ);
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/* FIXME : Check available size ? */
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guint8 channels;
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/* Opus streams can decode to 1 or 2 channels, so use the header
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value if we have one, or 2 otherwise */
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if (is_idheader) {
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channels = data[9];
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channel_mapping_family = data[18];
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/* header probing will already have done the size check */
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memcpy (channel_mapping, data + 21, channels);
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gst_buffer_unmap (frame->buffer, data, size);
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gst_buffer_replace (&parse->id_header, frame->buffer);
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GST_DEBUG_OBJECT (parse, "Found ID header, keeping");
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return GST_BASE_PARSE_FLOW_DROPPED;
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@ -318,6 +309,7 @@ gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
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gst_opus_header_create_caps_from_headers (&caps, &parse->headers,
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parse->id_header, parse->comment_header);
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} else {
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guint8 channel_mapping_family, channel_mapping[256];
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GST_INFO_OBJECT (parse,
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"No headers, blindly setting up canonical stereo");
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channels = 2;
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@ -473,7 +473,7 @@ gst_vp8_dec_handle_frame (GstBaseVideoDecoder * decoder,
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if (deadline < 0) {
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GST_LOG_OBJECT (dec, "Skipping late frame (%f s past deadline)",
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(double) -deadline / GST_SECOND);
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gst_base_video_decoder_finish_frame (decoder, frame);
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gst_base_video_decoder_drop_frame (decoder, frame);
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} else {
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ret = gst_base_video_decoder_alloc_src_frame (decoder, frame);
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@ -954,6 +954,9 @@ gst_base_video_decoder_reset (GstBaseVideoDecoder * base_video_decoder,
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base_video_decoder->current_frame = NULL;
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}
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base_video_decoder->dropped = 0;
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base_video_decoder->processed = 0;
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GST_BASE_VIDEO_CODEC (base_video_decoder)->system_frame_number = 0;
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base_video_decoder->base_picture_number = 0;
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@ -1349,30 +1352,12 @@ gst_base_video_decoder_new_frame (GstBaseVideoDecoder * base_video_decoder)
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return frame;
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}
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/**
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* gst_base_video_decoder_finish_frame:
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* @base_video_decoder: a #GstBaseVideoDecoder
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* @frame: a decoded #GstVideoFrameState
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*
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* @frame should have a valid decoded data buffer, whose metadata fields
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* are then appropriately set according to frame data and pushed downstream.
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* If no output data is provided, @frame is considered skipped.
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* In any case, the frame is considered finished and released.
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*
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* Returns: a #GstFlowReturn resulting from sending data downstream
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*/
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GstFlowReturn
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gst_base_video_decoder_finish_frame (GstBaseVideoDecoder * base_video_decoder,
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GstVideoFrameState * frame)
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static void
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gst_base_video_decoder_prepare_finish_frame (GstBaseVideoDecoder *
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base_video_decoder, GstVideoFrameState * frame)
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{
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GstVideoState *state = &GST_BASE_VIDEO_CODEC (base_video_decoder)->state;
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GstBuffer *src_buffer;
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GstFlowReturn ret = GST_FLOW_OK;
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GList *l, *events = NULL;
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GST_LOG_OBJECT (base_video_decoder, "finish frame");
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GST_BASE_VIDEO_CODEC_STREAM_LOCK (base_video_decoder);
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#ifndef GST_DISABLE_GST_DEBUG
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GST_LOG_OBJECT (base_video_decoder,
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"n %d in %" G_GSIZE_FORMAT " out %" G_GSIZE_FORMAT,
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@ -1398,9 +1383,12 @@ gst_base_video_decoder_finish_frame (GstBaseVideoDecoder * base_video_decoder,
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break;
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}
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for (l = g_list_last (events); l; l = l->prev)
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for (l = g_list_last (events); l; l = l->prev) {
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GST_LOG_OBJECT (base_video_decoder, "pushing %s event",
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GST_EVENT_TYPE_NAME (l->data));
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gst_pad_push_event (GST_BASE_VIDEO_CODEC_SRC_PAD (base_video_decoder),
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l->data);
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}
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g_list_free (events);
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if (GST_CLOCK_TIME_IS_VALID (frame->presentation_timestamp)) {
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@ -1459,8 +1447,106 @@ gst_base_video_decoder_finish_frame (GstBaseVideoDecoder * base_video_decoder,
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}
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}
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base_video_decoder->last_timestamp = frame->presentation_timestamp;
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}
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/* no buffer data means this frame is skipped/dropped */
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static void
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gst_base_video_decoder_do_finish_frame (GstBaseVideoDecoder * dec,
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GstVideoFrameState * frame)
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{
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GST_BASE_VIDEO_CODEC (dec)->frames =
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g_list_remove (GST_BASE_VIDEO_CODEC (dec)->frames, frame);
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if (frame->src_buffer)
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gst_buffer_unref (frame->src_buffer);
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gst_base_video_codec_free_frame (frame);
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}
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/**
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* gst_base_video_decoder_drop_frame:
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* @dec: a #GstBaseVideoDecoder
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* @frame: the #GstVideoFrame to drop
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*
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* Similar to gst_base_video_decoder_finish_frame(), but drops @frame in any
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* case and posts a QoS message with the frame's details on the bus.
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* In any case, the frame is considered finished and released.
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*
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* Returns: a #GstFlowReturn, usually GST_FLOW_OK.
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*
|
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* Since: 0.10.23
|
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*/
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GstFlowReturn
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gst_base_video_decoder_drop_frame (GstBaseVideoDecoder * dec,
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GstVideoFrameState * frame)
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{
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GstClockTime stream_time, jitter, earliest_time, qostime, timestamp;
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GstSegment *segment;
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GstMessage *qos_msg;
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gdouble proportion;
|
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|
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GST_LOG_OBJECT (dec, "drop frame");
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GST_BASE_VIDEO_CODEC_STREAM_LOCK (dec);
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gst_base_video_decoder_prepare_finish_frame (dec, frame);
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GST_DEBUG_OBJECT (dec, "dropping frame %" GST_TIME_FORMAT,
|
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GST_TIME_ARGS (frame->presentation_timestamp));
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|
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dec->dropped++;
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/* post QoS message */
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timestamp = frame->presentation_timestamp;
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proportion = GST_BASE_VIDEO_CODEC (dec)->proportion;
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segment = &GST_BASE_VIDEO_CODEC (dec)->segment;
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stream_time =
|
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gst_segment_to_stream_time (segment, GST_FORMAT_TIME, timestamp);
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qostime = gst_segment_to_running_time (segment, GST_FORMAT_TIME, timestamp);
|
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earliest_time = GST_BASE_VIDEO_CODEC (dec)->earliest_time;
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jitter = GST_CLOCK_DIFF (qostime, earliest_time);
|
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qos_msg = gst_message_new_qos (GST_OBJECT_CAST (dec), FALSE,
|
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qostime, stream_time, timestamp, GST_CLOCK_TIME_NONE);
|
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gst_message_set_qos_values (qos_msg, jitter, proportion, 1000000);
|
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gst_message_set_qos_stats (qos_msg, GST_FORMAT_BUFFERS,
|
||||
dec->processed, dec->dropped);
|
||||
gst_element_post_message (GST_ELEMENT_CAST (dec), qos_msg);
|
||||
|
||||
/* now free the frame */
|
||||
gst_base_video_decoder_do_finish_frame (dec, frame);
|
||||
|
||||
GST_BASE_VIDEO_CODEC_STREAM_UNLOCK (dec);
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_base_video_decoder_finish_frame:
|
||||
* @base_video_decoder: a #GstBaseVideoDecoder
|
||||
* @frame: a decoded #GstVideoFrameState
|
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*
|
||||
* @frame should have a valid decoded data buffer, whose metadata fields
|
||||
* are then appropriately set according to frame data and pushed downstream.
|
||||
* If no output data is provided, @frame is considered skipped.
|
||||
* In any case, the frame is considered finished and released.
|
||||
*
|
||||
* Returns: a #GstFlowReturn resulting from sending data downstream
|
||||
*/
|
||||
GstFlowReturn
|
||||
gst_base_video_decoder_finish_frame (GstBaseVideoDecoder * base_video_decoder,
|
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GstVideoFrameState * frame)
|
||||
{
|
||||
GstVideoState *state = &GST_BASE_VIDEO_CODEC (base_video_decoder)->state;
|
||||
GstBuffer *src_buffer;
|
||||
GstFlowReturn ret = GST_FLOW_OK;
|
||||
|
||||
GST_LOG_OBJECT (base_video_decoder, "finish frame");
|
||||
GST_BASE_VIDEO_CODEC_STREAM_LOCK (base_video_decoder);
|
||||
|
||||
gst_base_video_decoder_prepare_finish_frame (base_video_decoder, frame);
|
||||
|
||||
base_video_decoder->processed++;
|
||||
|
||||
/* no buffer data means this frame is skipped */
|
||||
if (!frame->src_buffer) {
|
||||
GST_DEBUG_OBJECT (base_video_decoder, "skipping frame %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (frame->presentation_timestamp));
|
||||
|
|
@ -1566,9 +1652,8 @@ gst_base_video_decoder_finish_frame (GstBaseVideoDecoder * base_video_decoder,
|
|||
}
|
||||
|
||||
done:
|
||||
GST_BASE_VIDEO_CODEC (base_video_decoder)->frames =
|
||||
g_list_remove (GST_BASE_VIDEO_CODEC (base_video_decoder)->frames, frame);
|
||||
gst_base_video_codec_free_frame (frame);
|
||||
|
||||
gst_base_video_decoder_do_finish_frame (base_video_decoder, frame);
|
||||
|
||||
GST_BASE_VIDEO_CODEC_STREAM_UNLOCK (base_video_decoder);
|
||||
|
||||
|
|
|
|||
|
|
@ -188,6 +188,10 @@ struct _GstBaseVideoDecoder
|
|||
* GST_META_API_VIDEO_CROP */
|
||||
gboolean use_cropping;
|
||||
|
||||
/* qos messages: frames dropped/processed */
|
||||
guint dropped;
|
||||
guint processed;
|
||||
|
||||
/* FIXME before moving to base */
|
||||
void *padding[GST_PADDING_LARGE];
|
||||
};
|
||||
|
|
@ -273,6 +277,8 @@ GstClockTimeDiff gst_base_video_decoder_get_max_decode_time (
|
|||
GstVideoFrameState *frame);
|
||||
GstFlowReturn gst_base_video_decoder_finish_frame (GstBaseVideoDecoder *base_video_decoder,
|
||||
GstVideoFrameState *frame);
|
||||
GstFlowReturn gst_base_video_decoder_drop_frame (GstBaseVideoDecoder *dec,
|
||||
GstVideoFrameState *frame);
|
||||
|
||||
GType gst_base_video_decoder_get_type (void);
|
||||
|
||||
|
|
|
|||
|
|
@ -57,6 +57,12 @@ else
|
|||
check_voaacenc =
|
||||
endif
|
||||
|
||||
if USE_VOAMRWBENC
|
||||
check_voamrwbenc = elements/voamrwbenc
|
||||
else
|
||||
check_voamrwbenc =
|
||||
endif
|
||||
|
||||
if USE_EXIF
|
||||
check_jifmux = elements/jifmux
|
||||
else
|
||||
|
|
@ -162,6 +168,7 @@ check_PROGRAMS = \
|
|||
$(check_faac) \
|
||||
$(check_faad) \
|
||||
$(check_voaacenc) \
|
||||
$(check_voamrwbenc) \
|
||||
$(check_mpeg2enc) \
|
||||
$(check_mplex) \
|
||||
$(check_ofa) \
|
||||
|
|
|
|||
186
tests/check/elements/voamrwbenc.c
Normal file
186
tests/check/elements/voamrwbenc.c
Normal file
|
|
@ -0,0 +1,186 @@
|
|||
/* GStreamer
|
||||
*
|
||||
* unit test for voamrwbenc
|
||||
*
|
||||
* Copyright (C) <2011> Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
||||
*
|
||||
* 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., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#include <unistd.h>
|
||||
|
||||
#include <gst/check/gstcheck.h>
|
||||
|
||||
/* For ease of programming we use globals to keep refs for our floating
|
||||
* src and sink pads we create; otherwise we always have to do get_pad,
|
||||
* get_peer, and then remove references in every test function */
|
||||
static GstPad *mysrcpad, *mysinkpad;
|
||||
|
||||
#define AUDIO_CAPS_STRING "audio/x-raw-int, " \
|
||||
"rate = (int) 16000, " \
|
||||
"channels = (int) 1, " \
|
||||
"width = (int) 16, " \
|
||||
"depth = (int) 16, " \
|
||||
"signed = (boolean) true, " \
|
||||
"endianness = (int) BYTE_ORDER "
|
||||
|
||||
#define AMRWB_CAPS_STRING "audio/AMR-WB"
|
||||
|
||||
|
||||
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
|
||||
GST_PAD_SINK,
|
||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS (AMRWB_CAPS_STRING));
|
||||
|
||||
|
||||
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
|
||||
GST_PAD_SRC,
|
||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS (AUDIO_CAPS_STRING));
|
||||
|
||||
|
||||
static GstElement *
|
||||
setup_voamrwbenc (void)
|
||||
{
|
||||
GstElement *voamrwbenc;
|
||||
|
||||
GST_DEBUG ("setup_voamrwbenc");
|
||||
voamrwbenc = gst_check_setup_element ("voamrwbenc");
|
||||
/* ensure mode as expected */
|
||||
g_object_set (voamrwbenc, "band-mode", 0, NULL);
|
||||
mysrcpad = gst_check_setup_src_pad (voamrwbenc, &srctemplate, NULL);
|
||||
mysinkpad = gst_check_setup_sink_pad (voamrwbenc, &sinktemplate, NULL);
|
||||
gst_pad_set_active (mysrcpad, TRUE);
|
||||
gst_pad_set_active (mysinkpad, TRUE);
|
||||
|
||||
return voamrwbenc;
|
||||
}
|
||||
|
||||
static void
|
||||
cleanup_voamrwbenc (GstElement * voamrwbenc)
|
||||
{
|
||||
GST_DEBUG ("cleanup_aacenc");
|
||||
gst_element_set_state (voamrwbenc, GST_STATE_NULL);
|
||||
|
||||
gst_pad_set_active (mysrcpad, FALSE);
|
||||
gst_pad_set_active (mysinkpad, FALSE);
|
||||
gst_check_teardown_src_pad (voamrwbenc);
|
||||
gst_check_teardown_sink_pad (voamrwbenc);
|
||||
gst_check_teardown_element (voamrwbenc);
|
||||
}
|
||||
|
||||
static void
|
||||
do_test (void)
|
||||
{
|
||||
GstElement *voamrwbenc;
|
||||
GstBuffer *inbuffer, *outbuffer;
|
||||
GstCaps *caps;
|
||||
gint i, num_buffers;
|
||||
const gint nbuffers = 10;
|
||||
|
||||
voamrwbenc = setup_voamrwbenc ();
|
||||
fail_unless (gst_element_set_state (voamrwbenc,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
|
||||
/* corresponds to audio buffer mentioned in the caps */
|
||||
inbuffer = gst_buffer_new_and_alloc (320 * nbuffers * 2);
|
||||
/* makes valgrind's memcheck happier */
|
||||
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
|
||||
caps = gst_caps_from_string (AUDIO_CAPS_STRING);
|
||||
|
||||
gst_buffer_set_caps (inbuffer, caps);
|
||||
gst_caps_unref (caps);
|
||||
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
|
||||
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
||||
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
||||
|
||||
/* send eos to have all flushed if needed */
|
||||
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()) == TRUE);
|
||||
|
||||
num_buffers = g_list_length (buffers);
|
||||
fail_unless_equals_int (num_buffers, nbuffers);
|
||||
|
||||
/* clean up buffers */
|
||||
for (i = 0; i < num_buffers; ++i) {
|
||||
gint size;
|
||||
guint8 *data;
|
||||
GstClockTime time, dur;
|
||||
|
||||
outbuffer = GST_BUFFER (buffers->data);
|
||||
fail_if (outbuffer == NULL);
|
||||
|
||||
data = GST_BUFFER_DATA (outbuffer);
|
||||
size = GST_BUFFER_SIZE (outbuffer);
|
||||
|
||||
/* at least for mode 0 */
|
||||
fail_unless (size == 18);
|
||||
fail_unless ((data[0] & 0x83) == 0);
|
||||
fail_unless (((data[0] >> 3) & 0xF) == 0);
|
||||
|
||||
time = GST_BUFFER_TIMESTAMP (outbuffer);
|
||||
dur = GST_BUFFER_DURATION (outbuffer);
|
||||
fail_unless (time == 20 * GST_MSECOND * i);
|
||||
fail_unless (dur == 20 * GST_MSECOND);
|
||||
|
||||
buffers = g_list_remove (buffers, outbuffer);
|
||||
|
||||
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
|
||||
gst_buffer_unref (outbuffer);
|
||||
outbuffer = NULL;
|
||||
}
|
||||
|
||||
cleanup_voamrwbenc (voamrwbenc);
|
||||
g_list_free (buffers);
|
||||
buffers = NULL;
|
||||
}
|
||||
|
||||
GST_START_TEST (test_enc)
|
||||
{
|
||||
do_test ();
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
|
||||
static Suite *
|
||||
voamrwbenc_suite (void)
|
||||
{
|
||||
Suite *s = suite_create ("voamrwbenc");
|
||||
TCase *tc_chain = tcase_create ("general");
|
||||
|
||||
suite_add_tcase (s, tc_chain);
|
||||
tcase_add_test (tc_chain, test_enc);
|
||||
|
||||
return s;
|
||||
}
|
||||
|
||||
int
|
||||
main (int argc, char **argv)
|
||||
{
|
||||
int nf;
|
||||
|
||||
Suite *s = voamrwbenc_suite ();
|
||||
SRunner *sr = srunner_create (s);
|
||||
|
||||
gst_check_init (&argc, &argv);
|
||||
|
||||
srunner_run_all (sr, CK_NORMAL);
|
||||
nf = srunner_ntests_failed (sr);
|
||||
srunner_free (sr);
|
||||
|
||||
return nf;
|
||||
}
|
||||
Loading…
Reference in a new issue