omx: Improve debug output

1 files changed, 20 insertions, 0 deletions

diff --git a/omx/gstomxaudioenc.c b/omx/gstomxaudioenc.c
index d6160fe..185ad5a 100644
--- a/omx/gstomxaudioenc.c
+++ b/omx/gstomxaudioenc.c
@@ -292,6 +292,8 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
       goto caps_failed;
     }
 
+    GST_DEBUG_OBJECT (self, "Setting output caps: %" GST_PTR_FORMAT, caps);
+
     if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
       gst_caps_unref (caps);
       if (buf)
@@ -333,6 +335,7 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
       GstBuffer *codec_data;
       GstMapInfo map = GST_MAP_INFO_INIT;
 
+      GST_DEBUG_OBJECT (self, "Handling codec data");
       caps =
           gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD
               (self)));
@@ -359,6 +362,8 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
       GstBuffer *outbuf;
       guint n_samples;
 
+      GST_DEBUG_OBJECT (self, "Handling output data");
+
       n_samples =
           klass->get_num_samples (self, self->out_port,
           gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
@@ -391,6 +396,8 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
           outbuf, n_samples);
     }
 
+    GST_DEBUG_OBJECT (self, "Handled output data");
+
     if (is_eos || flow_ret == GST_FLOW_EOS) {
       g_mutex_lock (&self->drain_lock);
       if (self->draining) {
@@ -565,17 +572,23 @@ gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
    * format change happened we can just exit here.
    */
   if (needs_disable) {
+    GST_DEBUG_OBJECT (self, "Need to disable and drain encoder");
     gst_omx_audio_enc_drain (self);
 
     if (gst_omx_port_manual_reconfigure (self->in_port, TRUE) != OMX_ErrorNone)
       return FALSE;
     if (gst_omx_port_set_enabled (self->in_port, FALSE) != OMX_ErrorNone)
       return FALSE;
+
+    GST_DEBUG_OBJECT (self, "Encoder drained and disabled");
   }
 
   port_def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+  GST_DEBUG_OBJECT (self, "Setting inport port definition");
   if (!gst_omx_port_update_port_definition (self->in_port, &port_def))
     return FALSE;
+
+  GST_DEBUG_OBJECT (self, "Setting outport port definition");
   if (!gst_omx_port_update_port_definition (self->out_port, NULL))
     return FALSE;
 
@@ -632,6 +645,7 @@ gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
     pcm_param.eChannelMapping[i] = pos;
   }
 
+  GST_DEBUG_OBJECT (self, "Setting PCM parameters");
   err =
       gst_omx_component_set_parameter (self->component, OMX_IndexParamAudioPcm,
       &pcm_param);
@@ -648,6 +662,7 @@ gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
     }
   }
 
+  GST_DEBUG_OBJECT (self, "Enabling component");
   if (needs_disable) {
     if (gst_omx_port_set_enabled (self->in_port, TRUE) != OMX_ErrorNone)
       return FALSE;
@@ -689,6 +704,7 @@ gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
   }
 
   /* Start the srcpad loop again */
+  GST_DEBUG_OBJECT (self, "Starting task again");
   self->downstream_flow_ret = GST_FLOW_OK;
   gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self),
       (GstTaskFunction) gst_omx_audio_enc_loop, encoder, NULL);
@@ -797,6 +813,8 @@ gst_omx_audio_enc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
       goto full_buffer;
     }
 
+    GST_DEBUG_OBJECT (self, "Handling frame at offset %d", offset);
+
     /* Copy the buffer content in chunks of size as requested
      * by the port */
     buf->omx_buf->nFilledLen =
@@ -828,6 +846,8 @@ gst_omx_audio_enc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
     gst_omx_port_release_buffer (self->in_port, buf);
   }
 
+  GST_DEBUG_OBJECT (self, "Passed frame to component");
+
   return self->downstream_flow_ret;
 
 full_buffer: