[gst-cvs] gst-plugins-good: rtpL16pay: convert to baseaudiopayload
Wim Taymans
wtay at kemper.freedesktop.org
Tue Dec 22 15:39:53 PST 2009
Module: gst-plugins-good
Branch: master
Commit: 2ee7f58416375496c6738119465240f57c0ff9d6
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=2ee7f58416375496c6738119465240f57c0ff9d6
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date: Wed Dec 23 00:38:05 2009 +0100
rtpL16pay: convert to baseaudiopayload
Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes
a bunch of problems that were already solved in the base class.
Fixes #853367
---
gst/rtp/gstrtpL16pay.c | 146 +++++-------------------------------------------
gst/rtp/gstrtpL16pay.h | 10 +--
2 files changed, 17 insertions(+), 139 deletions(-)
diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c
index 927b3db..1af2337 100644
--- a/gst/rtp/gstrtpL16pay.c
+++ b/gst/rtp/gstrtpL16pay.c
@@ -74,47 +74,16 @@ static GstStaticPadTemplate gst_rtp_L16_pay_src_template =
"clock-rate = (int) 44100")
);
-static void gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass);
-static void gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass);
-static void gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay);
-static void gst_rtp_L16_pay_finalize (GObject * object);
-
static gboolean gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
-static GstFlowReturn gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * pad,
- GstBuffer * buffer);
static GstCaps *gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload,
GstPad * pad);
-static GstBaseRTPPayloadClass *parent_class = NULL;
-
-static GType
-gst_rtp_L16_pay_get_type (void)
-{
- static GType rtpL16pay_type = 0;
-
- if (!rtpL16pay_type) {
- static const GTypeInfo rtpL16pay_info = {
- sizeof (GstRtpL16PayClass),
- (GBaseInitFunc) gst_rtp_L16_pay_base_init,
- NULL,
- (GClassInitFunc) gst_rtp_L16_pay_class_init,
- NULL,
- NULL,
- sizeof (GstRtpL16Pay),
- 0,
- (GInstanceInitFunc) gst_rtp_L16_pay_init,
- };
-
- rtpL16pay_type =
- g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpL16Pay",
- &rtpL16pay_info, 0);
- }
- return rtpL16pay_type;
-}
+GST_BOILERPLATE (GstRtpL16Pay, gst_rtp_L16_pay, GstBaseRTPAudioPayload,
+ GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
-gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass)
+gst_rtp_L16_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
@@ -129,41 +98,26 @@ gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass)
static void
gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
{
- GObjectClass *gobject_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
- gobject_class = (GObjectClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
- parent_class = g_type_class_peek_parent (klass);
-
- gobject_class->finalize = gst_rtp_L16_pay_finalize;
-
gstbasertppayload_class->set_caps = gst_rtp_L16_pay_setcaps;
gstbasertppayload_class->get_caps = gst_rtp_L16_pay_getcaps;
- gstbasertppayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0,
"L16 RTP Payloader");
}
static void
-gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay)
+gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay, GstRtpL16PayClass * klass)
{
- rtpL16pay->adapter = gst_adapter_new ();
-}
+ GstBaseRTPAudioPayload *basertpaudiopayload;
-static void
-gst_rtp_L16_pay_finalize (GObject * object)
-{
- GstRtpL16Pay *rtpL16pay;
+ basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpL16pay);
- rtpL16pay = GST_RTP_L16_PAY (object);
-
- g_object_unref (rtpL16pay->adapter);
- rtpL16pay->adapter = NULL;
-
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ /* tell basertpaudiopayload that this is a sample based codec */
+ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
}
static gboolean
@@ -176,7 +130,9 @@ gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
gchar *params;
GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
+ GstBaseRTPAudioPayload *basertpaudiopayload;
+ basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
rtpL16pay = GST_RTP_L16_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
@@ -219,6 +175,10 @@ gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
rtpL16pay->rate = rate;
rtpL16pay->channels = channels;
+ /* octet-per-sample is 2 * channels for L16 */
+ gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload,
+ 2 * rtpL16pay->channels);
+
return res;
/* ERRORS */
@@ -234,84 +194,6 @@ no_channels:
}
}
-static GstFlowReturn
-gst_rtp_L16_pay_flush (GstRtpL16Pay * rtpL16pay, guint len)
-{
- GstBuffer *outbuf;
- guint8 *payload;
- GstFlowReturn ret;
- guint samples;
- GstClockTime duration;
-
- /* calculate the amount of samples and round down the length */
- samples = len / (2 * rtpL16pay->channels);
- len = samples * (2 * rtpL16pay->channels);
-
- /* now alloc output buffer */
- outbuf = gst_rtp_buffer_new_allocate (len, 0, 0);
-
- /* get payload, this is now writable */
- payload = gst_rtp_buffer_get_payload (outbuf);
-
- /* copy and flush data out of adapter into the RTP payload */
- gst_adapter_copy (rtpL16pay->adapter, payload, 0, len);
- gst_adapter_flush (rtpL16pay->adapter, len);
-
- duration = gst_util_uint64_scale_int (samples, GST_SECOND, rtpL16pay->rate);
-
- GST_BUFFER_TIMESTAMP (outbuf) = rtpL16pay->first_ts;
- GST_BUFFER_DURATION (outbuf) = duration;
-
- /* increase count (in ts) of data pushed to basertppayload */
- if (GST_CLOCK_TIME_IS_VALID (rtpL16pay->first_ts))
- rtpL16pay->first_ts += duration;
-
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpL16pay), outbuf);
-
- return ret;
-}
-
-static GstFlowReturn
-gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * basepayload,
- GstBuffer * buffer)
-{
- GstRtpL16Pay *rtpL16pay;
- GstFlowReturn ret = GST_FLOW_OK;
- guint payload_len;
- GstClockTime timestamp;
- guint mtu, avail;
-
- rtpL16pay = GST_RTP_L16_PAY (basepayload);
- mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpL16pay);
-
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
-
- if (GST_BUFFER_IS_DISCONT (buffer))
- gst_adapter_clear (rtpL16pay->adapter);
-
- avail = gst_adapter_available (rtpL16pay->adapter);
- if (avail == 0) {
- rtpL16pay->first_ts = timestamp;
- }
-
- /* push buffer in adapter */
- gst_adapter_push (rtpL16pay->adapter, buffer);
-
- /* get payload len for MTU */
- payload_len = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
-
- /* flush complete MTU while we have enough data in the adapter */
- while (avail >= payload_len) {
- /* flush payload_len bytes */
- ret = gst_rtp_L16_pay_flush (rtpL16pay, payload_len);
- if (ret != GST_FLOW_OK)
- break;
-
- avail = gst_adapter_available (rtpL16pay->adapter);
- }
- return ret;
-}
-
static GstCaps *
gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
{
diff --git a/gst/rtp/gstrtpL16pay.h b/gst/rtp/gstrtpL16pay.h
index a1b79b9..8c30cee 100644
--- a/gst/rtp/gstrtpL16pay.h
+++ b/gst/rtp/gstrtpL16pay.h
@@ -21,8 +21,7 @@
#define __GST_RTP_L16_PAY_H__
#include <gst/gst.h>
-#include <gst/rtp/gstbasertppayload.h>
-#include <gst/base/gstadapter.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
@@ -42,10 +41,7 @@ typedef struct _GstRtpL16PayClass GstRtpL16PayClass;
struct _GstRtpL16Pay
{
- GstBaseRTPPayload payload;
-
- GstAdapter *adapter;
- GstClockTime first_ts;
+ GstBaseRTPAudioPayload payload;
gint rate;
gint channels;
@@ -53,7 +49,7 @@ struct _GstRtpL16Pay
struct _GstRtpL16PayClass
{
- GstBaseRTPPayloadClass parent_class;
+ GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_L16_pay_plugin_init (GstPlugin * plugin);
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