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| author | George Kiagiadakis <george.kiagiadakis@collabora.com> | 2018-02-19 18:30:13 +0200 |
|---|---|---|
| committer | George Kiagiadakis <george.kiagiadakis@collabora.com> | 2018-08-03 13:20:12 +0300 |
| commit | d299c27892b9eff24df7107f979e93856c06672a (patch) | |
| tree | a1695dbe9acdb0da03a18106221a84ea2a061e4f /ext/webrtcdsp/gstwebrtcechoprobe.h | |
| parent | 0591bc934a227d1c269b406c365178f77a8434f0 (diff) | |
| download | gstreamer-plugins-bad-d299c27892b9eff24df7107f979e93856c06672a.tar.gz | |
webrtcdsp: add support for using F32/non-interleaved buffers
This is the native format that is in use by the webrtc audio processing
library internally, so this avoids internal {de,}interleaving and
format conversion (S16->F32 and back)
https://bugzilla.gnome.org/show_bug.cgi?id=793605
Diffstat (limited to 'ext/webrtcdsp/gstwebrtcechoprobe.h')
| -rw-r--r-- | ext/webrtcdsp/gstwebrtcechoprobe.h | 8 |
1 files changed, 7 insertions, 1 deletions
diff --git a/ext/webrtcdsp/gstwebrtcechoprobe.h b/ext/webrtcdsp/gstwebrtcechoprobe.h index 7c8a24022..a8a889907 100644 --- a/ext/webrtcdsp/gstwebrtcechoprobe.h +++ b/ext/webrtcdsp/gstwebrtcechoprobe.h @@ -28,6 +28,9 @@ #include <gst/base/gstbasetransform.h> #include <gst/audio/audio.h> +#define GST_USE_UNSTABLE_API +#include <gst/audio/gstplanaraudioadapter.h> + G_BEGIN_DECLS #define GST_TYPE_WEBRTC_ECHO_PROBE (gst_webrtc_echo_probe_get_type()) @@ -62,11 +65,14 @@ struct _GstWebrtcEchoProbe /* Protected by the lock */ GstAudioInfo info; guint period_size; + guint period_samples; GstClockTime latency; gint delay; + gboolean interleaved; GstSegment segment; GstAdapter *adapter; + GstPlanarAudioAdapter *padapter; /* Private */ gboolean acquired; @@ -82,7 +88,7 @@ GType gst_webrtc_echo_probe_get_type (void); GstWebrtcEchoProbe *gst_webrtc_acquire_echo_probe (const gchar * name); void gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe); gint gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, - GstClockTime rec_time, gpointer frame); + GstClockTime rec_time, gpointer frame, GstBuffer ** buf); G_END_DECLS #endif /* __GST_WEBRTC_ECHO_PROBE_H__ */ |