Commit ed814cba authored by Mersad Jelacic's avatar Mersad Jelacic Committed by Peter Kjellerstedt
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gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add...

gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...

Original commit message from CVS:
Patch by: Mersad Jelacic  <mersad at axis dot com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
possible to specify the sample size in bits. (#509637)
parent e8c69d59
2008-03-03 Peter Kjellerstedt <pkj@axis.com>
Patch by: Mersad Jelacic <mersad at axis dot com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
possible to specify the sample size in bits. (#509637)
2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
 
* tests/check/libs/mixer.c:
......@@ -45,12 +45,13 @@
* gst_base_rtp_audio_payload_set_frame_based() or
* gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the
* element's _init() function. Then, the child element must call either
* gst_base_rtp_audio_payload_set_frame_options() or
* gst_base_rtp_audio_payload_set_sample_options(). Since GstBaseRTPAudioPayload
* derives from GstBaseRTPPayload, the child element must set any variables or
* call/override any functions required by that base class. The child element
* does not need to override any other functions specific to
* GstBaseRTPAudioPayload.
* gst_base_rtp_audio_payload_set_frame_options(),
* gst_base_rtp_audio_payload_set_sample_options() or
* gst_base_rtp_audio_payload_set_samplebits_options. Since
* GstBaseRTPAudioPayload derives from GstBaseRTPPayload, the child element
* must set any variables or call/override any functions required by that base
* class. The child element does not need to override any other functions
* specific to GstBaseRTPAudioPayload.
* </para>
* </refsect2>
*/
......@@ -251,6 +252,29 @@ gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
{
g_return_if_fail (basertpaudiopayload != NULL);
/* sample_size is in bits internally */
basertpaudiopayload->sample_size = sample_size * 8;
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
}
/**
* gst_base_rtp_audio_payload_set_samplebits_options:
* @basertpaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bits.
*
* Sets the options for sample based audio codecs.
*
* Since: 0.10.18
*/
void
gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
* basertpaudiopayload, gint sample_size)
{
g_return_if_fail (basertpaudiopayload != NULL);
basertpaudiopayload->sample_size = sample_size;
if (basertpaudiopayload->priv->adapter) {
......@@ -435,7 +459,7 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
guint max_payload_len;
gboolean use_adapter = FALSE;
guint sample_size;
guint fragment_size;
ret = GST_FLOW_OK;
......@@ -446,12 +470,17 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
sample_size = basertpaudiopayload->sample_size;
/* sample_size is in bits and is converted into multiple bytes */
fragment_size = basertpaudiopayload->sample_size;
while ((fragment_size % 8) != 0)
fragment_size += fragment_size;
fragment_size /= 8;
/* max number of bytes based on given ptime */
if (basepayload->max_ptime != -1) {
maxptime_octets = basepayload->max_ptime * basepayload->clock_rate /
(sample_size * GST_SECOND);
maxptime_octets = 8 * basepayload->max_ptime * basepayload->clock_rate /
(basertpaudiopayload->sample_size * GST_SECOND);
}
max_payload_len = MIN (
......@@ -463,10 +492,10 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
/* min number of bytes based on a given ptime, has to be a multiple
of sample rate */
minptime_octets = basepayload->min_ptime * basepayload->clock_rate /
(sample_size * GST_SECOND);
minptime_octets = 8 * basepayload->min_ptime * basepayload->clock_rate /
(basertpaudiopayload->sample_size * GST_SECOND);
min_payload_len = MAX (minptime_octets, sample_size);
min_payload_len = MAX (minptime_octets, fragment_size);
if (min_payload_len > max_payload_len) {
min_payload_len = max_payload_len;
......@@ -506,7 +535,7 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
gfloat num, datarate;
payload_len =
MIN (max_payload_len, (available / sample_size) * sample_size);
MIN (max_payload_len, (available / fragment_size) * fragment_size);
if (use_adapter) {
data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
......@@ -516,11 +545,11 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
basertpaudiopayload->base_ts);
num = payload_len;
datarate = (sample_size * basepayload->clock_rate);
num = payload_len * 8;
datarate = (basertpaudiopayload->sample_size * basepayload->clock_rate);
basertpaudiopayload->base_ts +=
/* payload_len (bytes) * nsecs/sec / datarate (bytes*sec) */
/* payload_len (bits) * nsecs/sec / datarate (bits*sec) */
gst_gdouble_to_guint64 (num / datarate * GST_SECOND);
GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT,
GST_TIME_ARGS (basertpaudiopayload->base_ts));
......
......@@ -84,6 +84,10 @@ void
gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint sample_size);
void
gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint sample_size);
GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp);
......
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