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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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audioaggregator: Improve log messages
Make the level of log messages saner and improve some.
This commit is contained in:
parent
9deb334251
commit
f0cbeb4140
1 changed files with 10 additions and 10 deletions
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@ -829,7 +829,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
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if (discont) {
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if (discont) {
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/* Have discont, need resync */
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/* Have discont, need resync */
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if (pad->priv->next_offset != -1)
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if (pad->priv->next_offset != -1)
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GST_INFO_OBJECT (pad, "Have discont. Expected %"
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GST_DEBUG_OBJECT (pad, "Have discont. Expected %"
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G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
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G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
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pad->priv->next_offset, start_offset);
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pad->priv->next_offset, start_offset);
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pad->priv->output_offset = -1;
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pad->priv->output_offset = -1;
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@ -995,7 +995,7 @@ gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg,
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if (pad->priv->position == pad->priv->size) {
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if (pad->priv->position == pad->priv->size) {
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/* Buffer done, drop it */
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/* Buffer done, drop it */
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gst_buffer_replace (&pad->priv->buffer, NULL);
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gst_buffer_replace (&pad->priv->buffer, NULL);
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GST_DEBUG_OBJECT (pad, "Finished mixing buffer, waiting for next");
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GST_LOG_OBJECT (pad, "Finished mixing buffer, waiting for next");
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return FALSE;
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return FALSE;
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}
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}
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@ -1194,8 +1194,8 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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if (timeout) {
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if (timeout) {
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if (pad->priv->output_offset < next_offset) {
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if (pad->priv->output_offset < next_offset) {
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gint64 diff = next_offset - pad->priv->output_offset;
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gint64 diff = next_offset - pad->priv->output_offset;
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GST_LOG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT " frames (%"
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GST_DEBUG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT
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GST_TIME_FORMAT ")", diff,
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" frames (%" GST_TIME_FORMAT ")", diff,
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GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
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GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
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GST_AUDIO_INFO_RATE (&aagg->info))));
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GST_AUDIO_INFO_RATE (&aagg->info))));
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}
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}
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@ -1241,7 +1241,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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pad->priv->output_offset += diff;
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pad->priv->output_offset += diff;
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if (pad->priv->position == pad->priv->size) {
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if (pad->priv->position == pad->priv->size) {
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GST_LOG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
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GST_DEBUG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
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", dropping %" GST_PTR_FORMAT,
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", dropping %" GST_PTR_FORMAT,
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GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
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GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
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GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer);
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GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer);
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@ -1262,8 +1262,8 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer,
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drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer,
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outbuf);
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outbuf);
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if (pad->priv->output_offset >= next_offset) {
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if (pad->priv->output_offset >= next_offset) {
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GST_DEBUG_OBJECT (pad,
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GST_LOG_OBJECT (pad,
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"Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
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"Pad is at or after current offset: %" G_GUINT64_FORMAT " >= %"
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G_GINT64_FORMAT, pad->priv->output_offset, next_offset);
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G_GINT64_FORMAT, pad->priv->output_offset, next_offset);
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} else {
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} else {
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is_done = FALSE;
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is_done = FALSE;
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@ -1279,15 +1279,15 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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if (dropped) {
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if (dropped) {
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/* We dropped a buffer, retry */
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/* We dropped a buffer, retry */
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GST_INFO_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
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GST_LOG_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
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GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
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GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
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return GST_FLOW_OK;
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return GST_FLOW_OK;
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}
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}
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if (!is_done && !is_eos) {
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if (!is_done && !is_eos) {
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/* Get more buffers */
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/* Get more buffers */
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GST_INFO_OBJECT (aagg,
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GST_LOG_OBJECT (aagg,
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"We're not done yet for the current offset," " waiting for more data");
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"We're not done yet for the current offset, waiting for more data");
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GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
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GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
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return GST_FLOW_OK;
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return GST_FLOW_OK;
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}
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}
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