audiofx: Sync properties to the stream time

8 files changed, 104 insertions, 27 deletions

diff --git a/gst/audiofx/audioamplify.c b/gst/audiofx/audioamplify.c
index 4273626fa..fa1cc2e2c 100644
--- a/gst/audiofx/audioamplify.c
+++ b/gst/audiofx/audioamplify.c
@@ -479,11 +479,21 @@ static GstFlowReturn
 gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
 {
   GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
-  guint num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+  guint num_samples;
+  GstClockTime timestamp, stream_time;
+
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+  stream_time =
+      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
 
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
-    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (G_OBJECT (filter), stream_time);
+
+  num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
 
   if (gst_base_transform_is_passthrough (base) ||
       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
diff --git a/gst/audiofx/audiodynamic.c b/gst/audiofx/audiodynamic.c
index 240c270e7..dc096d4f7 100644
--- a/gst/audiofx/audiodynamic.c
+++ b/gst/audiofx/audiodynamic.c
@@ -695,11 +695,21 @@ static GstFlowReturn
 gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
 {
   GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
-  guint num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+  guint num_samples;
+  GstClockTime timestamp, stream_time;
+
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+  stream_time =
+      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
 
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
-    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (G_OBJECT (filter), stream_time);
+
+  num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
 
   if (gst_base_transform_is_passthrough (base) ||
       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
diff --git a/gst/audiofx/audioecho.c b/gst/audiofx/audioecho.c
index 885fb341d..cf4dab192 100644
--- a/gst/audiofx/audioecho.c
+++ b/gst/audiofx/audioecho.c
@@ -357,11 +357,21 @@ static GstFlowReturn
 gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
 {
   GstAudioEcho *self = GST_AUDIO_ECHO (base);
-  guint num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
+  guint num_samples;
+  GstClockTime timestamp, stream_time;
+
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+  stream_time =
+      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+  GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
 
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
-    gst_object_sync_values (G_OBJECT (self), GST_BUFFER_TIMESTAMP (buf));
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (G_OBJECT (self), stream_time);
+
+  num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
 
   if (self->buffer == NULL) {
     guint width, rate, channels;
diff --git a/gst/audiofx/audiofxbasefirfilter.c b/gst/audiofx/audiofxbasefirfilter.c
index 37fa639cf..ac1b14858 100644
--- a/gst/audiofx/audiofxbasefirfilter.c
+++ b/gst/audiofx/audiofxbasefirfilter.c
@@ -822,15 +822,24 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
   guint generated_samples;
   guint64 output_offset;
   gint64 diff = 0;
+  GstClockTime stream_time;
 
   timestamp = GST_BUFFER_TIMESTAMP (outbuf);
+
   if (!GST_CLOCK_TIME_IS_VALID (timestamp)
       && !GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
     GST_ERROR_OBJECT (self, "Invalid timestamp");
     return GST_FLOW_ERROR;
   }
 
-  gst_object_sync_values (G_OBJECT (self), timestamp);
+  stream_time =
+      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+  GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
+
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (G_OBJECT (self), stream_time);
 
   g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
   g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
diff --git a/gst/audiofx/audiofxbaseiirfilter.c b/gst/audiofx/audiofxbaseiirfilter.c
index 4571a2daf..0b0fc3464 100644
--- a/gst/audiofx/audiofxbaseiirfilter.c
+++ b/gst/audiofx/audiofxbaseiirfilter.c
@@ -355,11 +355,21 @@ gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base,
     GstBuffer * buf)
 {
   GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base);
-  guint num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+  guint num_samples;
+  GstClockTime timestamp, stream_time;
+
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+  stream_time =
+      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
 
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
-    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (G_OBJECT (filter), stream_time);
+
+  num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
 
   if (gst_base_transform_is_passthrough (base))
     return GST_FLOW_OK;
diff --git a/gst/audiofx/audioinvert.c b/gst/audiofx/audioinvert.c
index a6911697f..df43a0934 100644
--- a/gst/audiofx/audioinvert.c
+++ b/gst/audiofx/audioinvert.c
@@ -236,11 +236,21 @@ static GstFlowReturn
 gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
 {
   GstAudioInvert *filter = GST_AUDIO_INVERT (base);
-  guint num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+  guint num_samples;
+  GstClockTime timestamp, stream_time;
+
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+  stream_time =
+      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
 
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
-    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (G_OBJECT (filter), stream_time);
+
+  num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
 
   if (gst_base_transform_is_passthrough (base) ||
       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
diff --git a/gst/audiofx/audiokaraoke.c b/gst/audiofx/audiokaraoke.c
index fe34971e3..c39a79bc9 100644
--- a/gst/audiofx/audiokaraoke.c
+++ b/gst/audiofx/audiokaraoke.c
@@ -341,11 +341,21 @@ static GstFlowReturn
 gst_audio_karaoke_transform_ip (GstBaseTransform * base, GstBuffer * buf)
 {
   GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);
-  guint num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+  guint num_samples;
+  GstClockTime timestamp, stream_time;
+
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+  stream_time =
+      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
 
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
-    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (G_OBJECT (filter), stream_time);
+
+  num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
 
   if (gst_base_transform_is_passthrough (base) ||
       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
diff --git a/gst/audiofx/audiopanorama.c b/gst/audiofx/audiopanorama.c
index 3f57648dd..7bbbfef43 100644
--- a/gst/audiofx/audiopanorama.c
+++ b/gst/audiofx/audiopanorama.c
@@ -631,9 +631,17 @@ gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf,
 {
   GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
   guint num_samples = GST_BUFFER_SIZE (outbuf) / (2 * filter->width);
+  GstClockTime timestamp, stream_time;
 
-  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (outbuf)))
-    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (outbuf));
+  timestamp = GST_BUFFER_TIMESTAMP (inbuf);
+  stream_time =
+      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (timestamp));
+
+  if (GST_CLOCK_TIME_IS_VALID (stream_time))
+    gst_object_sync_values (G_OBJECT (filter), stream_time);
 
   if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
     GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);