diff --git a/gst/audiomixer/gstaudioaggregator.c b/gst/audiomixer/gstaudioaggregator.c index e609dda47e..9155b1baa7 100644 --- a/gst/audiomixer/gstaudioaggregator.c +++ b/gst/audiomixer/gstaudioaggregator.c @@ -892,7 +892,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg, pad->priv->output_offset = -1; GST_DEBUG_OBJECT (pad, "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %" - G_GUINT64_FORMAT, end_output_offset, aagg->priv->offset); + G_GINT64_FORMAT, end_output_offset, aagg->priv->offset); return FALSE; } @@ -922,7 +922,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg, pad->priv->output_offset = -1; GST_DEBUG_OBJECT (pad, "Buffer before segment or current position: %" G_GUINT64_FORMAT - " < %" G_GUINT64_FORMAT, end_output_offset, aagg->priv->offset); + " < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset); return FALSE; } } @@ -934,7 +934,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg, GST_DEBUG_OBJECT (pad, "Buffer resynced: Pad offset %" G_GUINT64_FORMAT - ", current audio aggregator offset %" G_GUINT64_FORMAT, + ", current audio aggregator offset %" G_GINT64_FORMAT, pad->priv->output_offset, aagg->priv->offset); } @@ -1139,7 +1139,8 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout) aagg->priv->offset = gst_util_uint64_scale (agg->segment.position - agg->segment.start, rate, GST_SECOND); - GST_DEBUG_OBJECT (aagg, "Starting at offset %lu", aagg->priv->offset); + GST_DEBUG_OBJECT (aagg, "Starting at offset %" G_GINT64_FORMAT, + aagg->priv->offset); } blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration, @@ -1172,7 +1173,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout) outbuf = aagg->priv->current_buffer; GST_LOG_OBJECT (agg, - "Starting to mix %u samples for offset %" G_GUINT64_FORMAT + "Starting to mix %u samples for offset %" G_GINT64_FORMAT " with timestamp %" GST_TIME_FORMAT, blocksize, aagg->priv->offset, GST_TIME_ARGS (agg->segment.position)); @@ -1263,7 +1264,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout) if (pad->priv->output_offset >= next_offset) { GST_DEBUG_OBJECT (pad, "Pad is after current offset: %" G_GUINT64_FORMAT " >= %" - G_GUINT64_FORMAT, pad->priv->output_offset, next_offset); + G_GINT64_FORMAT, pad->priv->output_offset, next_offset); } else { is_done = FALSE; } @@ -1314,7 +1315,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout) if (max_offset <= next_offset) { GST_DEBUG_OBJECT (aagg, "Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %" - G_GUINT64_FORMAT, max_offset, next_offset); + G_GINT64_FORMAT, max_offset, next_offset); next_offset = max_offset; next_timestamp = agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,