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Bug 431672 - gst_base_rtp_audio_payload_push() should take object of its own type as its first argument
gst_base_rtp_audio_payload_push() should take object of its own type as its f...
Status: RESOLVED FIXED
Product: GStreamer
Classification: Platform
Component: gst-plugins-base
git master
Other Linux
: Normal normal
: 0.10.13
Assigned To: GStreamer Maintainers
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2007-04-20 12:57 UTC by Zeeshan Ali
Modified: 2007-04-21 14:41 UTC
See Also:
GNOME target: ---
GNOME version: ---

Attachments
Patch to fix the problem (3.38 KB, patch)
2007-04-20 12:58 UTC, Zeeshan Ali 		committed Details | Review

Description Zeeshan Ali 2007-04-20 12:57:04 UTC 
GstBaseRTPAudioPayload API should take an object of it's own type (GstBaseRTPAudioPayload) as it's first argument rather than that of it's parent (GstBaseRTPPayload). Attaching the patch.
Comment 1 Zeeshan Ali 2007-04-20 12:58:19 UTC 
Created attachment 86687 [details] [review]
Patch to fix the problem
Comment 2 Tim-Philipp Müller 2007-04-20 13:11:09 UTC 
Absolutely, see bug #415001.
Comment 3 Tim-Philipp Müller 2007-04-21 14:41:40 UTC 
2007-04-21  Tim-Philipp Müller  <tim at centricular dot net>

        Patch by: Zeeshan Ali  <zeenix gmail com>

        * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
        (gst_base_rtp_audio_payload_handle_frame_based_buffer),
        (gst_base_rtp_audio_payload_handle_sample_based_buffer),
        (gst_base_rtp_audio_payload_push):
        * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
          The recently-added gst_base_rtp_audio_payload_push() should take an
          object of type GstBaseRTPAudioPayload as first argument (#431672).