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gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...

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Original commit message from CVS:
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).
This commit is contained in:
Zeeshan Ali 2007-04-21 14:40:45 +00:00 committed by Tim-Philipp Müller
parent 97cff37e11
commit 80ebb9eb42
3 changed files with 24 additions and 7 deletions
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12
ChangeLog
View file

@ -1,3 +1,15 @@
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).
2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
* gst/audioresample/gstaudioresample.c:

17
gst-libs/gst/rtp/gstbasertpaudiopayload.c
View file

@ -414,7 +414,7 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
/* this will check against max_ptime and max_mtu */
if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
GST_BUFFER_SIZE (buffer) <= max_payload_len) {
ret = gst_base_rtp_audio_payload_push (basepayload,
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
GST_BUFFER_TIMESTAMP (buffer));
gst_buffer_unref (buffer);
@ -437,7 +437,8 @@ gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
}
ret = gst_base_rtp_audio_payload_push (basepayload, data, payload_len,
ret =
gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
basertpaudiopayload->base_ts);
ts_inc = (payload_len * frame_duration) / frame_size;
@ -540,7 +541,7 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
/* this will check against max_ptime and max_mtu */
if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
GST_BUFFER_SIZE (buffer) <= max_payload_len) {
ret = gst_base_rtp_audio_payload_push (basepayload,
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
GST_BUFFER_TIMESTAMP (buffer));
gst_buffer_unref (buffer);
@ -562,7 +563,8 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
}
ret = gst_base_rtp_audio_payload_push (basepayload, data, payload_len,
ret =
gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
basertpaudiopayload->base_ts);
num = payload_len;
@ -612,14 +614,17 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
* Returns: a #GstFlowReturn
*/
GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload,
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
{
GstBaseRTPPayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
GST_DEBUG_OBJECT (basepayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create buffer to hold the payload */

2
gst-libs/gst/rtp/gstbasertpaudiopayload.h
View file

@ -85,7 +85,7 @@ gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint sample_size);
GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload,
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp);
GstAdapter*