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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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voaacenc: Remove custom getcaps and just use the template
We know from the beginning the caps that are acceptable, no need for custom getcaps or manually generating caps.
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00b3832b54
commit
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1 changed files with 7 additions and 73 deletions
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@ -66,13 +66,19 @@ enum
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"88200, " \
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"96000"
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/* voaacenc only supports 1 or 2 channels */
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) [1, 2]")
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"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) 1;"
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"audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) 2, "
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"channel-mask=(bitmask)0x3")
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);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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@ -98,7 +104,6 @@ static gboolean gst_voaacenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_voaacenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc, GstCaps * filter);
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G_DEFINE_TYPE (GstVoAacEnc, gst_voaacenc, GST_TYPE_AUDIO_ENCODER);
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@ -150,7 +155,6 @@ gst_voaacenc_class_init (GstVoAacEncClass * klass)
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base_class->stop = GST_DEBUG_FUNCPTR (gst_voaacenc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_voaacenc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_voaacenc_handle_frame);
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base_class->getcaps = GST_DEBUG_FUNCPTR (gst_voaacenc_getcaps);
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g_object_class_install_property (object_class, PROP_BITRATE,
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g_param_spec_int ("bitrate",
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@ -240,76 +244,6 @@ static const GstAudioChannelPosition
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GST_AUDIO_CHANNEL_POSITION_LFE1}
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};
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static gpointer
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gst_voaacenc_generate_sink_caps (gpointer data)
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{
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GstCaps *caps;
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gint i, c;
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static const int rates[] = {
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8000, 11025, 12000, 16000, 22050, 24000,
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32000, 44100, 48000, 64000, 88200, 96000
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};
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GValue rates_arr = { 0, };
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GValue tmp = { 0, };
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GstStructure *s, *t;
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g_value_init (&rates_arr, GST_TYPE_LIST);
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g_value_init (&tmp, G_TYPE_INT);
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for (i = 0; i < G_N_ELEMENTS (rates); i++) {
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g_value_set_int (&tmp, rates[i]);
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gst_value_list_append_value (&rates_arr, &tmp);
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}
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g_value_unset (&tmp);
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s = gst_structure_new ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
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"layout", G_TYPE_STRING, "interleaved", NULL);
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gst_structure_set_value (s, "rate", &rates_arr);
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caps = gst_caps_new_empty ();
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for (i = 1; i <= 2 /* VOAAC_ENC_MAX_CHANNELS */ ; i++) {
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guint64 channel_mask = 0;
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t = gst_structure_copy (s);
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gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
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if (i > 1) {
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for (c = 0; c < i; c++)
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channel_mask |=
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G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
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gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask,
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NULL);
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}
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gst_caps_append_structure (caps, t);
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}
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gst_structure_free (s);
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g_value_unset (&rates_arr);
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GST_DEBUG ("generated sink caps: %" GST_PTR_FORMAT, caps);
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return caps;
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}
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static GstCaps *
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gst_voaacenc_get_sink_caps (void)
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{
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static GOnce g_once = G_ONCE_INIT;
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GstCaps *caps;
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g_once (&g_once, gst_voaacenc_generate_sink_caps, NULL);
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caps = g_once.retval;
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return caps;
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}
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static GstCaps *
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gst_voaacenc_getcaps (GstAudioEncoder * benc, GstCaps * filter)
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{
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return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps (),
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filter);
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}
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/* check downstream caps to configure format */
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static void
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gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
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