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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-03-30 12:49:40 +00:00
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
This commit is contained in:
Wim Taymans
Wim Taymans
parent
cdb8c718bb
commit
2ee7f58416
2 changed files with 17 additions and 139 deletions
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@ -74,47 +74,16 @@ static GstStaticPadTemplate gst_rtp_L16_pay_src_template =
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"clock-rate = (int) 44100")
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);
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static void gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass);
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static void gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass);
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static void gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay);
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static void gst_rtp_L16_pay_finalize (GObject * object);
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static gboolean gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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static GstCaps *gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload,
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GstPad * pad);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtp_L16_pay_get_type (void)
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{
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static GType rtpL16pay_type = 0;
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if (!rtpL16pay_type) {
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static const GTypeInfo rtpL16pay_info = {
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sizeof (GstRtpL16PayClass),
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(GBaseInitFunc) gst_rtp_L16_pay_base_init,
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NULL,
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(GClassInitFunc) gst_rtp_L16_pay_class_init,
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NULL,
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NULL,
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sizeof (GstRtpL16Pay),
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0,
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(GInstanceInitFunc) gst_rtp_L16_pay_init,
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};
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rtpL16pay_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpL16Pay",
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&rtpL16pay_info, 0);
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}
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return rtpL16pay_type;
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}
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GST_BOILERPLATE (GstRtpL16Pay, gst_rtp_L16_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass)
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gst_rtp_L16_pay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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@ -129,41 +98,26 @@ gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass)
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static void
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gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_rtp_L16_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_L16_pay_setcaps;
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gstbasertppayload_class->get_caps = gst_rtp_L16_pay_getcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0,
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"L16 RTP Payloader");
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}
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static void
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gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay)
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gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay, GstRtpL16PayClass * klass)
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{
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rtpL16pay->adapter = gst_adapter_new ();
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}
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GstBaseRTPAudioPayload *basertpaudiopayload;
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static void
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gst_rtp_L16_pay_finalize (GObject * object)
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{
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GstRtpL16Pay *rtpL16pay;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpL16pay);
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rtpL16pay = GST_RTP_L16_PAY (object);
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g_object_unref (rtpL16pay->adapter);
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rtpL16pay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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/* tell basertpaudiopayload that this is a sample based codec */
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gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
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}
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static gboolean
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@ -176,7 +130,9 @@ gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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gchar *params;
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GstAudioChannelPosition *pos;
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const GstRTPChannelOrder *order;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
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rtpL16pay = GST_RTP_L16_PAY (basepayload);
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structure = gst_caps_get_structure (caps, 0);
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@ -219,6 +175,10 @@ gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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rtpL16pay->rate = rate;
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rtpL16pay->channels = channels;
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/* octet-per-sample is 2 * channels for L16 */
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gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload,
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2 * rtpL16pay->channels);
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return res;
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/* ERRORS */
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@ -234,84 +194,6 @@ no_channels:
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}
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}
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static GstFlowReturn
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gst_rtp_L16_pay_flush (GstRtpL16Pay * rtpL16pay, guint len)
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{
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GstBuffer *outbuf;
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guint8 *payload;
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GstFlowReturn ret;
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guint samples;
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GstClockTime duration;
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/* calculate the amount of samples and round down the length */
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samples = len / (2 * rtpL16pay->channels);
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len = samples * (2 * rtpL16pay->channels);
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/* now alloc output buffer */
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outbuf = gst_rtp_buffer_new_allocate (len, 0, 0);
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/* get payload, this is now writable */
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payload = gst_rtp_buffer_get_payload (outbuf);
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/* copy and flush data out of adapter into the RTP payload */
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gst_adapter_copy (rtpL16pay->adapter, payload, 0, len);
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gst_adapter_flush (rtpL16pay->adapter, len);
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duration = gst_util_uint64_scale_int (samples, GST_SECOND, rtpL16pay->rate);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpL16pay->first_ts;
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GST_BUFFER_DURATION (outbuf) = duration;
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/* increase count (in ts) of data pushed to basertppayload */
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if (GST_CLOCK_TIME_IS_VALID (rtpL16pay->first_ts))
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rtpL16pay->first_ts += duration;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpL16pay), outbuf);
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return ret;
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}
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static GstFlowReturn
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gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpL16Pay *rtpL16pay;
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GstFlowReturn ret = GST_FLOW_OK;
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guint payload_len;
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GstClockTime timestamp;
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guint mtu, avail;
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rtpL16pay = GST_RTP_L16_PAY (basepayload);
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mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpL16pay);
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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if (GST_BUFFER_IS_DISCONT (buffer))
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gst_adapter_clear (rtpL16pay->adapter);
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avail = gst_adapter_available (rtpL16pay->adapter);
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if (avail == 0) {
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rtpL16pay->first_ts = timestamp;
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}
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/* push buffer in adapter */
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gst_adapter_push (rtpL16pay->adapter, buffer);
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/* get payload len for MTU */
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payload_len = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
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/* flush complete MTU while we have enough data in the adapter */
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while (avail >= payload_len) {
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/* flush payload_len bytes */
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ret = gst_rtp_L16_pay_flush (rtpL16pay, payload_len);
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if (ret != GST_FLOW_OK)
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break;
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avail = gst_adapter_available (rtpL16pay->adapter);
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}
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return ret;
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}
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static GstCaps *
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gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
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{
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@ -21,8 +21,7 @@
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#define __GST_RTP_L16_PAY_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstbasertppayload.h>
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#include <gst/base/gstadapter.h>
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#include <gst/rtp/gstbasertpaudiopayload.h>
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G_BEGIN_DECLS
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@ -42,10 +41,7 @@ typedef struct _GstRtpL16PayClass GstRtpL16PayClass;
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struct _GstRtpL16Pay
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{
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GstBaseRTPPayload payload;
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GstAdapter *adapter;
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GstClockTime first_ts;
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GstBaseRTPAudioPayload payload;
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gint rate;
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gint channels;
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@ -53,7 +49,7 @@ struct _GstRtpL16Pay
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struct _GstRtpL16PayClass
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{
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GstBaseRTPPayloadClass parent_class;
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GstBaseRTPAudioPayloadClass parent_class;
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};
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gboolean gst_rtp_L16_pay_plugin_init (GstPlugin * plugin);
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