mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-03 09:12:19 +00:00
interaudio: Properly pass through caps from the sink to the source
Otherwise a magic capsfilter after the source is required with exactly the same caps as the input.
This commit is contained in:
parent
211a39e55e
commit
6f72e8ceab
5 changed files with 136 additions and 52 deletions
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@ -61,6 +61,8 @@ static void gst_inter_audio_sink_get_times (GstBaseSink * sink,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static gboolean gst_inter_audio_sink_start (GstBaseSink * sink);
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static gboolean gst_inter_audio_sink_stop (GstBaseSink * sink);
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static gboolean gst_inter_audio_sink_set_caps (GstBaseSink * sink,
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GstCaps * caps);
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static GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
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GstBuffer * buffer);
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@ -108,6 +110,7 @@ gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
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GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_times);
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base_sink_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_start);
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base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_stop);
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base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_set_caps);
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base_sink_class->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
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g_object_class_install_property (gobject_class, PROP_CHANNEL,
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@ -177,10 +180,10 @@ gst_inter_audio_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
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if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
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*end = *start + GST_BUFFER_DURATION (buffer);
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} else {
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if (interaudiosink->fps_n > 0) {
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if (interaudiosink->info.rate > 0) {
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*end = *start +
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gst_util_uint64_scale_int (GST_SECOND, interaudiosink->fps_d,
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interaudiosink->fps_n);
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gst_util_uint64_scale_int (gst_buffer_get_size (buffer), GST_SECOND,
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interaudiosink->info.rate * interaudiosink->info.bpf);
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}
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}
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}
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@ -194,6 +197,9 @@ gst_inter_audio_sink_start (GstBaseSink * sink)
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GST_DEBUG ("start");
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interaudiosink->surface = gst_inter_surface_get (interaudiosink->channel);
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g_mutex_lock (&interaudiosink->surface->mutex);
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memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
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g_mutex_unlock (&interaudiosink->surface->mutex);
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return TRUE;
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}
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@ -207,6 +213,7 @@ gst_inter_audio_sink_stop (GstBaseSink * sink)
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g_mutex_lock (&interaudiosink->surface->mutex);
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gst_adapter_clear (interaudiosink->surface->audio_adapter);
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memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
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g_mutex_unlock (&interaudiosink->surface->mutex);
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gst_inter_surface_unref (interaudiosink->surface);
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@ -215,22 +222,41 @@ gst_inter_audio_sink_stop (GstBaseSink * sink)
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return TRUE;
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}
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static gboolean
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gst_inter_audio_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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GstAudioInfo info;
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if (!gst_audio_info_from_caps (&info, caps)) {
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GST_ERROR_OBJECT (sink, "Failed to parse caps %" GST_PTR_FORMAT, caps);
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return FALSE;
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}
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g_mutex_lock (&interaudiosink->surface->mutex);
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interaudiosink->surface->audio_info = info;
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interaudiosink->info = info;
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g_mutex_unlock (&interaudiosink->surface->mutex);
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return TRUE;
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}
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static GstFlowReturn
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gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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int n;
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int n, bpf;
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GST_DEBUG ("render %" G_GSIZE_FORMAT, gst_buffer_get_size (buffer));
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bpf = interaudiosink->info.bpf;
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g_mutex_lock (&interaudiosink->surface->mutex);
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n = gst_adapter_available (interaudiosink->surface->audio_adapter) / 4;
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#define SIZE 1600
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if (n > (SIZE * 3)) {
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int n_chunks = (n / (SIZE / 2)) - 4;
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GST_WARNING ("flushing %d samples", n_chunks * 800);
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if (n > SIZE * 3) {
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GST_WARNING ("flushing %d samples", SIZE / 2);
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gst_adapter_flush (interaudiosink->surface->audio_adapter,
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n_chunks * (SIZE / 2) * 4);
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(SIZE / 2) * bpf);
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}
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gst_adapter_push (interaudiosink->surface->audio_adapter,
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gst_buffer_ref (buffer));
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@ -41,8 +41,7 @@ struct _GstInterAudioSink
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GstInterSurface *surface;
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char *channel;
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int fps_n;
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int fps_d;
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GstAudioInfo info;
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};
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struct _GstInterAudioSinkClass
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@ -57,6 +57,8 @@ static void gst_inter_audio_src_get_property (GObject * object,
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guint property_id, GValue * value, GParamSpec * pspec);
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static void gst_inter_audio_src_finalize (GObject * object);
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static GstCaps *gst_inter_audio_src_get_caps (GstBaseSrc * src,
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GstCaps * filter);
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static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps);
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static gboolean gst_inter_audio_src_start (GstBaseSrc * src);
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static gboolean gst_inter_audio_src_stop (GstBaseSrc * src);
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@ -86,6 +88,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
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/* class initialization */
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#define parent_class gst_inter_audio_src_parent_class
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G_DEFINE_TYPE (GstInterAudioSrc, gst_inter_audio_src, GST_TYPE_BASE_SRC);
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static void
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@ -110,6 +113,7 @@ gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass)
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gobject_class->set_property = gst_inter_audio_src_set_property;
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gobject_class->get_property = gst_inter_audio_src_get_property;
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gobject_class->finalize = gst_inter_audio_src_finalize;
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base_src_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_caps);
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base_src_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_set_caps);
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base_src_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_src_start);
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base_src_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_src_stop);
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@ -178,36 +182,51 @@ gst_inter_audio_src_finalize (GObject * object)
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G_OBJECT_CLASS (gst_inter_audio_src_parent_class)->finalize (object);
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}
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static GstCaps *
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gst_inter_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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GstCaps *caps;
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GST_DEBUG_OBJECT (interaudiosrc, "get_caps");
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if (!interaudiosrc->surface)
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return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter);
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g_mutex_lock (&interaudiosrc->surface->mutex);
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if (interaudiosrc->surface->audio_info.finfo) {
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caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
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if (filter) {
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GstCaps *tmp;
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tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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caps = tmp;
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}
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} else {
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caps = NULL;
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}
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g_mutex_unlock (&interaudiosrc->surface->mutex);
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if (caps)
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return caps;
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else
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return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter);
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}
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static gboolean
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gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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const GstStructure *structure;
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GstAudioInfo info;
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gboolean ret;
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int sample_rate;
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GST_DEBUG_OBJECT (interaudiosrc, "set_caps");
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "rate", &sample_rate)) {
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GST_ERROR_OBJECT (src, "Audio caps without rate");
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if (!gst_audio_info_from_caps (&interaudiosrc->info, caps)) {
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GST_ERROR_OBJECT (src, "Failed to parse caps %" GST_PTR_FORMAT, caps);
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return FALSE;
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}
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interaudiosrc->sample_rate = sample_rate;
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if (!gst_audio_info_from_caps (&info, caps)) {
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GST_ERROR_OBJECT (src, "Can't parse audio caps");
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return FALSE;
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}
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interaudiosrc->finfo = info.finfo;
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ret = gst_pad_set_caps (src->srcpad, caps);
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return ret;
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return gst_pad_set_caps (src->srcpad, caps);
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}
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static gboolean
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@ -218,6 +237,8 @@ gst_inter_audio_src_start (GstBaseSrc * src)
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GST_DEBUG_OBJECT (interaudiosrc, "start");
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interaudiosrc->surface = gst_inter_surface_get (interaudiosrc->channel);
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interaudiosrc->timestamp_offset = 0;
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interaudiosrc->n_samples = 0;
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return TRUE;
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}
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@ -231,7 +252,6 @@ gst_inter_audio_src_stop (GstBaseSrc * src)
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gst_inter_surface_unref (interaudiosrc->surface);
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interaudiosrc->surface = NULL;
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interaudiosrc->finfo = NULL;
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return TRUE;
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}
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@ -268,38 +288,68 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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GstBuffer ** buf)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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GstCaps *caps;
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GstBuffer *buffer;
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int n;
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int n, bpf;
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GST_DEBUG_OBJECT (interaudiosrc, "create");
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buffer = NULL;
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caps = NULL;
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g_mutex_lock (&interaudiosrc->surface->mutex);
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n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / 4;
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if (interaudiosrc->surface->audio_info.finfo) {
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if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info,
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&interaudiosrc->info)) {
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caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
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interaudiosrc->timestamp_offset =
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gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
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interaudiosrc->info.rate);
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interaudiosrc->n_samples = 0;
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}
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}
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bpf = interaudiosrc->surface->audio_info.bpf;
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n = bpf >
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0 ? gst_adapter_available (interaudiosrc->surface->audio_adapter) /
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bpf : 0;
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if (n > SIZE * 3) {
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GST_WARNING ("flushing %d samples", SIZE / 2);
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gst_adapter_flush (interaudiosrc->surface->audio_adapter, (SIZE / 2) * 4);
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gst_adapter_flush (interaudiosrc->surface->audio_adapter, (SIZE / 2) * bpf);
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n -= (SIZE / 2);
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}
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if (n > SIZE)
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n = SIZE;
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if (n > 0) {
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buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
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n * 4);
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n * bpf);
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} else {
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buffer = gst_buffer_new ();
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}
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g_mutex_unlock (&interaudiosrc->surface->mutex);
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if (caps) {
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gboolean ret = gst_base_src_set_caps (src, caps);
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gst_caps_unref (caps);
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if (!ret) {
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GST_ERROR_OBJECT (src, "Failed to set caps %" GST_PTR_FORMAT, caps);
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if (buffer)
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gst_buffer_unref (buffer);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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bpf = interaudiosrc->info.bpf;
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if (n < SIZE) {
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GstMapInfo map;
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GstMemory *mem;
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GST_WARNING ("creating %d samples of silence", SIZE - n);
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mem = gst_allocator_alloc (NULL, (SIZE - n) * 4, NULL);
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mem = gst_allocator_alloc (NULL, (SIZE - n) * bpf, NULL);
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if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
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gst_audio_format_fill_silence (interaudiosrc->finfo, map.data, map.size);
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gst_audio_format_fill_silence (interaudiosrc->info.finfo, map.data,
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map.size);
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gst_memory_unmap (mem, &map);
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}
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buffer = gst_buffer_make_writable (buffer);
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@ -311,12 +361,12 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
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GST_BUFFER_TIMESTAMP (buffer) =
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gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
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interaudiosrc->sample_rate);
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interaudiosrc->info.rate);
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GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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GST_BUFFER_DURATION (buffer) =
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gst_util_uint64_scale_int (interaudiosrc->n_samples + n, GST_SECOND,
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interaudiosrc->sample_rate) - GST_BUFFER_TIMESTAMP (buffer);
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interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer);
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GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
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GST_BUFFER_OFFSET_END (buffer) = -1;
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GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
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@ -333,6 +383,7 @@ gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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static gboolean
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gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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gboolean ret;
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GST_DEBUG_OBJECT (src, "query");
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@ -341,18 +392,26 @@ gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
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case GST_QUERY_LATENCY:{
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GstClockTime min_latency, max_latency;
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min_latency = 30 * gst_util_uint64_scale_int (GST_SECOND, SIZE, 48000);
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if (interaudiosrc->info.rate > 0) {
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/* FIXME: Where does the 30 come from? */
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max_latency = min_latency;
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min_latency =
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30 * gst_util_uint64_scale_int (GST_SECOND, SIZE,
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interaudiosrc->info.rate);
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GST_ERROR_OBJECT (src,
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"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
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GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
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max_latency = min_latency;
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gst_query_set_latency (query,
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gst_base_src_is_live (src), min_latency, max_latency);
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GST_ERROR_OBJECT (src,
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"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
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GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
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ret = TRUE;
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gst_query_set_latency (query,
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gst_base_src_is_live (src), min_latency, max_latency);
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ret = TRUE;
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} else {
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ret = FALSE;
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}
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break;
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}
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default:
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@ -375,6 +434,7 @@ gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps)
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_fixate_field_string (structure, "format", GST_AUDIO_NE (S16));
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gst_structure_fixate_field_nearest_int (structure, "channels", 2);
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gst_structure_fixate_field_nearest_int (structure, "rate", 48000);
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@ -43,9 +43,8 @@ struct _GstInterAudioSrc
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char *channel;
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guint64 n_samples;
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int sample_rate;
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const GstAudioFormatInfo *finfo;
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GstClockTime timestamp_offset;
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GstAudioInfo info;
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};
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struct _GstInterAudioSrcClass
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@ -21,6 +21,7 @@
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#define _GST_INTER_SURFACE_H_
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#include <gst/base/gstadapter.h>
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#include <gst/audio/audio.h>
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#include <gst/video/video.h>
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G_BEGIN_DECLS
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@ -39,8 +40,7 @@ struct _GstInterSurface
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int video_buffer_count;
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/* audio */
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int sample_rate;
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int n_channels;
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GstAudioInfo audio_info;
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GstBuffer *video_buffer;
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GstBuffer *sub_buffer;
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