+2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * gst/rtp/gstrtppcmapay.c:
+ * gst/rtp/gstrtppcmapay.h:
+ * gst/rtp/gstrtppcmupay.c:
+ * gst/rtp/gstrtppcmupay.h:
+ Ported mulaw and alaw payloaders to use new base class
+
2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
* po/af.po:
static gboolean gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
-static GstFlowReturn gst_rtp_pcma_pay_handle_buffer (GstBaseRTPPayload *
- payload, GstBuffer * buffer);
-static void gst_rtp_pcma_pay_finalize (GObject * object);
-GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPPayload,
- GST_TYPE_BASE_RTP_PAYLOAD);
-
-/* The lower limit for number of octet to put in one packet
- * (clock-rate=8000, octet-per-sample=1). The default 80 is equal
- * to to 10msec (see RFC3551) */
-#define GST_RTP_PCMA_MIN_PTIME_OCTETS 80
+GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPAudioPayload,
+ GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_pcma_pay_base_init (gpointer klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
- gobject_class->finalize = gst_rtp_pcma_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_pcma_pay_setcaps;
- gstbasertppayload_class->handle_buffer = gst_rtp_pcma_pay_handle_buffer;
}
static void
gst_rtp_pcma_pay_init (GstRtpPmcaPay * rtppcmapay, GstRtpPmcaPayClass * klass)
{
- rtppcmapay->adapter = gst_adapter_new ();
- GST_BASE_RTP_PAYLOAD (rtppcmapay)->clock_rate = 8000;
-}
+ GstBaseRTPAudioPayload *basertpaudiopayload;
-static void
-gst_rtp_pcma_pay_finalize (GObject * object)
-{
- GstRtpPmcaPay *rtppcmapay;
+ basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmapay);
- rtppcmapay = GST_RTP_PCMA_PAY (object);
+ GST_BASE_RTP_PAYLOAD (rtppcmapay)->clock_rate = 8000;
- g_object_unref (rtppcmapay->adapter);
- rtppcmapay->adapter = NULL;
+ /* tell basertpaudiopayload that this is a sample based codec */
+ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ /* octet-per-sample is 1 for PCM */
+ gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
}
static gboolean
return TRUE;
}
-static GstFlowReturn
-gst_rtp_pcma_pay_flush (GstRtpPmcaPay * rtppcmapay, guint32 clock_rate)
-{
- guint avail;
- GstBuffer *outbuf;
- GstFlowReturn ret;
- guint maxptime_octets = G_MAXUINT;
- guint minptime_octets = GST_RTP_PCMA_MIN_PTIME_OCTETS;
-
- if (GST_BASE_RTP_PAYLOAD (rtppcmapay)->max_ptime > 0) {
- /* calculate octet count with:
- maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
- maxptime_octets =
- gst_util_uint64_scale_int (GST_BASE_RTP_PAYLOAD (rtppcmapay)->max_ptime,
- clock_rate, GST_SECOND);
- }
-
- /* the data available in the adapter is either smaller
- * than the MTU or bigger. In the case it is smaller, the complete
- * adapter contents can be put in one packet. */
- avail = gst_adapter_available (rtppcmapay->adapter);
-
- ret = GST_FLOW_OK;
-
- while (avail >= minptime_octets) {
- guint8 *payload;
- guint8 *data;
- guint payload_len;
- guint packet_len;
-
- /* fill one MTU or all available bytes */
- payload_len =
- MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmapay), maxptime_octets),
- avail);
-
- /* this will be the total lenght of the packet */
- packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
-
- /* create buffer to hold the payload */
- outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
-
- /* copy payload */
- gst_rtp_buffer_set_payload_type (outbuf,
- GST_BASE_RTP_PAYLOAD_PT (rtppcmapay));
- payload = gst_rtp_buffer_get_payload (outbuf);
- data = (guint8 *) gst_adapter_peek (rtppcmapay->adapter, payload_len);
- memcpy (payload, data, payload_len);
- gst_adapter_flush (rtppcmapay->adapter, payload_len);
-
- avail -= payload_len;
-
- GST_BUFFER_TIMESTAMP (outbuf) = rtppcmapay->first_ts;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmapay), outbuf);
-
- /* increase count (in ts) of data pushed to basertppayload */
- rtppcmapay->first_ts +=
- gst_util_uint64_scale_int (payload_len, GST_SECOND, clock_rate);
-
- /* store amount of unpushed data (in ts) */
- rtppcmapay->duration =
- gst_util_uint64_scale_int (avail, GST_SECOND, clock_rate);
- }
-
- return ret;
-}
-
-static GstFlowReturn
-gst_rtp_pcma_pay_handle_buffer (GstBaseRTPPayload * basepayload,
- GstBuffer * buffer)
-{
- GstRtpPmcaPay *rtppcmapay;
- guint size, packet_len, avail;
- GstFlowReturn ret;
- GstClockTime duration;
- guint32 clock_rate;
-
- rtppcmapay = GST_RTP_PCMA_PAY (basepayload);
-
- clock_rate = basepayload->clock_rate;
-
- size = GST_BUFFER_SIZE (buffer);
- duration = gst_util_uint64_scale_int (size, GST_SECOND, clock_rate);
-
- avail = gst_adapter_available (rtppcmapay->adapter);
- if (avail == 0) {
- rtppcmapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
- rtppcmapay->duration = 0;
- }
-
- /* get packet length of data and see if we exceeded MTU. */
- packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
-
- /* if this buffer is going to overflow the packet, flush what we
- * have. */
- if (gst_basertppayload_is_filled (basepayload,
- packet_len, rtppcmapay->duration + duration)) {
- ret = gst_rtp_pcma_pay_flush (rtppcmapay, clock_rate);
- /* note: first_ts and duration updated in ...pay_flush() */
- } else {
- ret = GST_FLOW_OK;
- }
-
- gst_adapter_push (rtppcmapay->adapter, buffer);
- rtppcmapay->duration += duration;
-
- return ret;
-}
-
gboolean
gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin)
{
#define __GST_RTP_PCMA_PAY_H__
#include <gst/gst.h>
-#include <gst/rtp/gstbasertppayload.h>
-#include <gst/base/gstadapter.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
struct _GstRtpPmcaPay
{
- GstBaseRTPPayload payload;
- GstAdapter *adapter;
-
- GstClockTime first_ts;
- GstClockTime duration;
+ GstBaseRTPAudioPayload audiopayload;
};
struct _GstRtpPmcaPayClass
{
- GstBaseRTPPayloadClass parent_class;
+ GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin);
static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
-static GstFlowReturn gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload *
- payload, GstBuffer * buffer);
-static void gst_rtp_pcmu_pay_finalize (GObject * object);
-GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPPayload,
- GST_TYPE_BASE_RTP_PAYLOAD);
-
-/* The lower limit for number of octet to put in one packet
- * (clock-rate=8000, octet-per-sample=1). The default 80 is equal
- * to to 10msec (see RFC3551) */
-#define GST_RTP_PCMU_MIN_PTIME_OCTETS 80
+GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPAudioPayload,
+ GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_pcmu_pay_base_init (gpointer klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
- gobject_class->finalize = gst_rtp_pcmu_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
- gstbasertppayload_class->handle_buffer = gst_rtp_pcmu_pay_handle_buffer;
}
static void
gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
{
- rtppcmupay->adapter = gst_adapter_new ();
- GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
-}
+ GstBaseRTPAudioPayload *basertpaudiopayload;
-static void
-gst_rtp_pcmu_pay_finalize (GObject * object)
-{
- GstRtpPcmuPay *rtppcmupay;
+ basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmupay);
- rtppcmupay = GST_RTP_PCMU_PAY (object);
+ GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
- g_object_unref (rtppcmupay->adapter);
- rtppcmupay->adapter = NULL;
+ /* tell basertpaudiopayload that this is a sample based codec */
+ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ /* octet-per-sample is 1 for PCM */
+ gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
}
static gboolean
return TRUE;
}
-static GstFlowReturn
-gst_rtp_pcmu_pay_flush (GstRtpPcmuPay * rtppcmupay, guint32 clock_rate)
-{
- guint avail;
- GstBuffer *outbuf;
- GstFlowReturn ret;
- guint maxptime_octets = G_MAXUINT;
- guint minptime_octets = GST_RTP_PCMU_MIN_PTIME_OCTETS;
-
- if (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime > 0) {
- /* calculate octet count with:
- maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
- maxptime_octets =
- gst_util_uint64_scale_int (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime,
- clock_rate, GST_SECOND);
- }
-
- /* the data available in the adapter is either smaller
- * than the MTU or bigger. In the case it is smaller, the complete
- * adapter contents can be put in one packet. */
- avail = gst_adapter_available (rtppcmupay->adapter);
-
- ret = GST_FLOW_OK;
-
- while (avail >= minptime_octets) {
- guint8 *payload;
- guint8 *data;
- guint payload_len;
- guint packet_len;
-
- /* fill one MTU or all available bytes */
- payload_len =
- MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmupay), maxptime_octets),
- avail);
-
- /* this will be the total lenght of the packet */
- packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
-
- /* create buffer to hold the payload */
- outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
-
- /* copy payload */
- gst_rtp_buffer_set_payload_type (outbuf,
- GST_BASE_RTP_PAYLOAD_PT (rtppcmupay));
- payload = gst_rtp_buffer_get_payload (outbuf);
- data = (guint8 *) gst_adapter_peek (rtppcmupay->adapter, payload_len);
- memcpy (payload, data, payload_len);
- gst_adapter_flush (rtppcmupay->adapter, payload_len);
-
- avail -= payload_len;
-
- GST_BUFFER_TIMESTAMP (outbuf) = rtppcmupay->first_ts;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmupay), outbuf);
-
- /* increase count (in ts) of data pushed to basertppayload */
- rtppcmupay->first_ts +=
- gst_util_uint64_scale_int (payload_len, GST_SECOND, clock_rate);
-
- /* store amount of unpushed data (in ts) */
- rtppcmupay->duration =
- gst_util_uint64_scale_int (avail, GST_SECOND, clock_rate);
- }
-
- return ret;
-}
-
-static GstFlowReturn
-gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload * basepayload,
- GstBuffer * buffer)
-{
- GstRtpPcmuPay *rtppcmupay;
- guint size, packet_len, avail;
- GstFlowReturn ret;
- GstClockTime duration;
- guint32 clock_rate;
-
- rtppcmupay = GST_RTP_PCMU_PAY (basepayload);
-
- clock_rate = basepayload->clock_rate;
-
- size = GST_BUFFER_SIZE (buffer);
- duration = gst_util_uint64_scale_int (size, GST_SECOND, clock_rate);
-
- avail = gst_adapter_available (rtppcmupay->adapter);
- if (avail == 0) {
- rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
- rtppcmupay->duration = 0;
- }
-
- /* get packet length of data and see if we exceeded MTU. */
- packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
-
- /* if this buffer is going to overflow the packet, flush what we
- * have. */
- if (gst_basertppayload_is_filled (basepayload,
- packet_len, rtppcmupay->duration + duration)) {
- /* note: first_ts and duration updated in ...pay_flush() */
- ret = gst_rtp_pcmu_pay_flush (rtppcmupay, clock_rate);
- } else {
- ret = GST_FLOW_OK;
- }
-
- gst_adapter_push (rtppcmupay->adapter, buffer);
- rtppcmupay->duration += duration;
-
- return ret;
-}
-
gboolean
gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
{
#define __GST_RTP_PCMU_PAY_H__
#include <gst/gst.h>
-#include <gst/rtp/gstbasertppayload.h>
-#include <gst/base/gstadapter.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
struct _GstRtpPcmuPay
{
- GstBaseRTPPayload payload;
- GstAdapter *adapter;
-
- GstClockTime first_ts;
- GstClockTime duration;
+ GstBaseRTPAudioPayload audiopayload;
};
struct _GstRtpPcmuPayClass
{
- GstBaseRTPPayloadClass parent_class;
+ GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin);
projects / platform / upstream / gstreamer.git / commitdiff