diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c
index 7d139f0b22..367c70099a 100644
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c
@@ -33,9 +33,9 @@ gst_audio_buffer_unmap_internal (GstAudioBuffer * buffer, guint n_unmap)
     gst_buffer_unmap (buffer->buffer, &buffer->map_infos[i]);
   }
   if (buffer->planes != buffer->priv_planes_arr)
-    g_slice_free1 (buffer->n_planes * sizeof (gpointer), buffer->planes);
+    g_free (buffer->planes);
   if (buffer->map_infos != buffer->priv_map_infos_arr)
-    g_slice_free1 (buffer->n_planes * sizeof (GstMapInfo), buffer->map_infos);
+    g_free (buffer->map_infos);
 }
 
 /**
@@ -146,9 +146,8 @@ gst_audio_buffer_map (GstAudioBuffer * buffer, const GstAudioInfo * info,
     buffer->n_planes = GST_AUDIO_BUFFER_CHANNELS (buffer);
 
     if (G_UNLIKELY (buffer->n_planes > 8)) {
-      buffer->planes = g_slice_alloc (buffer->n_planes * sizeof (gpointer));
-      buffer->map_infos =
-          g_slice_alloc (buffer->n_planes * sizeof (GstMapInfo));
+      buffer->planes = g_new (gpointer, buffer->n_planes);
+      buffer->map_infos = g_new (GstMapInfo, buffer->n_planes);
     } else {
       buffer->planes = buffer->priv_planes_arr;
       buffer->map_infos = buffer->priv_map_infos_arr;
diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c
index 07c1148a9a..ab574a7045 100644
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c
@@ -95,7 +95,7 @@ gst_audio_channel_mixer_free (GstAudioChannelMixer * mix)
   g_free (mix->matrix_int);
   mix->matrix_int = NULL;
 
-  g_slice_free (GstAudioChannelMixer, mix);
+  g_free (mix);
 }
 
 /*
@@ -836,7 +836,7 @@ gst_audio_channel_mixer_new_with_matrix (GstAudioChannelMixerFlags flags,
   g_return_val_if_fail (in_channels > 0 && in_channels < 64, NULL);
   g_return_val_if_fail (out_channels > 0 && out_channels < 64, NULL);
 
-  mix = g_slice_new0 (GstAudioChannelMixer);
+  mix = g_new0 (GstAudioChannelMixer, 1);
   mix->in_channels = in_channels;
   mix->out_channels = out_channels;
 
diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c
index f1116d3917..13d8e5511f 100644
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c
@@ -197,7 +197,7 @@ audio_chain_new (AudioChain * prev, GstAudioConverter * convert)
 {
   AudioChain *chain;
 
-  chain = g_slice_new0 (AudioChain);
+  chain = g_new0 (AudioChain, 1);
   chain->prev = prev;
 
   if (convert->current_layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
@@ -229,7 +229,7 @@ audio_chain_free (AudioChain * chain)
   if (chain->make_func_notify)
     chain->make_func_notify (chain->make_func_data);
   g_free (chain->tmp);
-  g_slice_free (AudioChain, chain);
+  g_free (chain);
 }
 
 static gpointer *
@@ -1347,7 +1347,7 @@ gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo * in_info,
       && !opt_matrix)
     goto unpositioned;
 
-  convert = g_slice_new0 (GstAudioConverter);
+  convert = g_new0 (GstAudioConverter, 1);
 
   convert->flags = flags;
   convert->in = *in_info;
@@ -1481,7 +1481,7 @@ gst_audio_converter_free (GstAudioConverter * convert)
 
   gst_structure_free (convert->config);
 
-  g_slice_free (GstAudioConverter, convert);
+  g_free (convert);
 }
 
 /**
diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c
index a37e3cf025..13bbb4b979 100644
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c
@@ -61,7 +61,7 @@ ensure_debug_category (void)
 GstAudioInfo *
 gst_audio_info_copy (const GstAudioInfo * info)
 {
-  return g_slice_dup (GstAudioInfo, info);
+  return g_memdup2 (info, sizeof (GstAudioInfo));
 }
 
 /**
@@ -74,7 +74,7 @@ gst_audio_info_copy (const GstAudioInfo * info)
 void
 gst_audio_info_free (GstAudioInfo * info)
 {
-  g_slice_free (GstAudioInfo, info);
+  g_free (info);
 }
 
 G_DEFINE_BOXED_TYPE (GstAudioInfo, gst_audio_info,
@@ -93,7 +93,7 @@ gst_audio_info_new (void)
 {
   GstAudioInfo *info;
 
-  info = g_slice_new (GstAudioInfo);
+  info = g_new (GstAudioInfo, 1);
   gst_audio_info_init (info);
 
   return info;
diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c
index e4c33ba6a1..82f8b066e2 100644
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c
@@ -446,7 +446,7 @@ gst_audio_quantize_new (GstAudioDitherMethod dither,
   g_return_val_if_fail (format == GST_AUDIO_FORMAT_S32, NULL);
   g_return_val_if_fail (channels > 0, NULL);
 
-  quant = g_slice_new0 (GstAudioQuantize);
+  quant = g_new0 (GstAudioQuantize, 1);
   quant->dither = dither;
   quant->ns = ns;
   quant->flags = flags;
@@ -490,7 +490,7 @@ gst_audio_quantize_free (GstAudioQuantize * quant)
   g_free (quant->last_random);
   g_free (quant->dither_buf);
 
-  g_slice_free (GstAudioQuantize, quant);
+  g_free (quant);
 }
 
 /**
diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c
index a612d85ad2..fc80a2ddfc 100644
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c
@@ -1366,7 +1366,7 @@ gst_audio_resampler_new (GstAudioResamplerMethod method,
 
   audio_resampler_init ();
 
-  resampler = g_slice_new0 (GstAudioResampler);
+  resampler = g_new0 (GstAudioResampler, 1);
   resampler->method = method;
   resampler->flags = flags;
   resampler->format = format;
@@ -1634,7 +1634,7 @@ gst_audio_resampler_free (GstAudioResampler * resampler)
   g_free (resampler->sbuf);
   if (resampler->options)
     gst_structure_free (resampler->options);
-  g_slice_free (GstAudioResampler, resampler);
+  g_free (resampler);
 }
 
 /**
diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c
index 5605dc7b4c..21d1d82db2 100644
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c
@@ -327,7 +327,7 @@ gst_audio_meta_free (GstMeta * meta, GstBuffer * buffer)
   GstAudioMeta *ameta = (GstAudioMeta *) meta;
 
   if (ameta->offsets && ameta->offsets != ameta->priv_offsets_arr)
-    g_slice_free1 (ameta->info.channels * sizeof (gsize), ameta->offsets);
+    g_free (ameta->offsets);
 }
 
 static gboolean
@@ -413,7 +413,7 @@ gst_buffer_add_audio_meta (GstBuffer * buffer, const GstAudioInfo * info,
 #endif
 
     if (G_UNLIKELY (info->channels > 8))
-      meta->offsets = g_slice_alloc (info->channels * sizeof (gsize));
+      meta->offsets = g_new (gsize, info->channels);
     else
       meta->offsets = meta->priv_offsets_arr;
 
diff --git a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c
index ac71dd06a7..15de9c1d55 100644
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c
+++ b/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c
@@ -633,7 +633,7 @@ gst_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
 
   GST_INFO_OBJECT (buf, "Allocating an array for %d timestamps",
       spec->segtotal);
-  buf->timestamps = g_slice_alloc0 (sizeof (GstClockTime) * spec->segtotal);
+  buf->timestamps = g_new0 (GstClockTime, spec->segtotal);
   /* initialize array with invalid timestamps */
   for (i = 0; i < spec->segtotal; i++) {
     buf->timestamps[i] = GST_CLOCK_TIME_NONE;
@@ -727,7 +727,7 @@ gst_audio_ring_buffer_release (GstAudioRingBuffer * buf)
   if (G_LIKELY (buf->timestamps)) {
     GST_INFO_OBJECT (buf, "Freeing timestamp buffer, %d entries",
         buf->spec.segtotal);
-    g_slice_free1 (sizeof (GstClockTime) * buf->spec.segtotal, buf->timestamps);
+    g_free (buf->timestamps);
     buf->timestamps = NULL;
   }