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rtpg729pay: avoid basertppayload perfect-rtptime mode
G729 packets may only occur intermittently (e.g. cn packets), and as such do not allow for perfect-rtptime calculating rtp times based on frame or byte count. In particular, do not use rtp audio base payloader as base class, but rather base payloader directly.
This commit is contained in:
Mark Nauwelaerts
parent
6405df0c50
commit
f1fe0e7157
2 changed files with 112 additions and 23 deletions
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@ -48,6 +48,9 @@ gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
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static GstFlowReturn
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gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
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static GstStateChangeReturn
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gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition);
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static GstStaticPadTemplate gst_rtp_g729_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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@ -71,8 +74,8 @@ static GstStaticPadTemplate gst_rtp_g729_pay_src_template =
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"clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
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);
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GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_g729_pay_base_init (gpointer klass)
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@ -92,11 +95,27 @@ gst_rtp_g729_pay_base_init (gpointer klass)
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"G.729 RTP Payloader");
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}
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static void
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gst_rtp_g729_pay_finalize (GObject * object)
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{
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GstRTPG729Pay *pay = GST_RTP_G729_PAY (object);
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g_object_unref (pay->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
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gobject_class->finalize = gst_rtp_g729_pay_finalize;
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gstelement_class->change_state = gst_rtp_g729_pay_change_state;
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payload_class->set_caps = gst_rtp_g729_pay_set_caps;
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payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
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}
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@ -105,15 +124,18 @@ static void
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gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass)
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{
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GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
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GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
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payload->pt = GST_RTP_PAYLOAD_G729;
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gst_basertppayload_set_options (payload, "audio", FALSE, "G729", 8000);
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gst_base_rtp_audio_payload_set_frame_based (audiopayload);
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gst_base_rtp_audio_payload_set_frame_options (audiopayload,
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G729_FRAME_DURATION_MS, G729_FRAME_SIZE);
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pay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_g729_pay_reset (GstRTPG729Pay * pay)
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{
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gst_adapter_clear (pay->adapter);
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pay->discont = FALSE;
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}
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static gboolean
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@ -135,12 +157,49 @@ gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
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return res;
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}
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static GstFlowReturn
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gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay,
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const guint8 * data, guint payload_len, GstClockTime timestamp,
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GstClockTime duration)
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{
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GstBaseRTPPayload *basepayload;
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GstBuffer *outbuf;
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guint8 *payload;
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GstFlowReturn ret;
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basepayload = GST_BASE_RTP_PAYLOAD (rtpg729pay);
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GST_DEBUG_OBJECT (rtpg729pay, "Pushing %d bytes ts %" GST_TIME_FORMAT,
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payload_len, GST_TIME_ARGS (timestamp));
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* copy payload */
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payload = gst_rtp_buffer_get_payload (outbuf);
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memcpy (payload, data, payload_len);
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/* set metadata */
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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GST_BUFFER_DURATION (outbuf) = duration;
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if (G_UNLIKELY (rtpg729pay->discont)) {
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GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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gst_rtp_buffer_set_marker (outbuf, TRUE);
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rtpg729pay->discont = FALSE;
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}
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ret = gst_basertppayload_push (basepayload, outbuf);
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return ret;
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}
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static GstFlowReturn
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gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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GstBaseRTPAudioPayload *basertpaudiopayload =
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GST_BASE_RTP_AUDIO_PAYLOAD (payload);
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GstRTPG729Pay *rtpg729pay = GST_RTP_G729_PAY (payload);
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GstAdapter *adapter = NULL;
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guint payload_len;
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guint available;
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@ -164,7 +223,7 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
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(int) (ptime_ms / G729_FRAME_DURATION_MS);
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if (maxptime_octets < G729_FRAME_SIZE) {
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GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %" G_GINT64_FORMAT
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GST_WARNING_OBJECT (payload, "Given ptime %" G_GINT64_FORMAT
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" is smaller than minimum %d ns, overwriting to minimum",
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payload->max_ptime, G729_FRAME_DURATION_MS);
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maxptime_octets = G729_FRAME_SIZE;
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@ -174,7 +233,8 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
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max_payload_len = MIN (
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/* MTU max */
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(int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
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(basertpaudiopayload), 0, 0) / G729_FRAME_SIZE) * G729_FRAME_SIZE,
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(payload), 0, 0) / G729_FRAME_SIZE)
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* G729_FRAME_SIZE,
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/* ptime max */
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maxptime_octets);
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@ -207,25 +267,26 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
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min_payload_len = max_payload_len = ptime_in_bytes;
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}
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GST_LOG_OBJECT (basertpaudiopayload,
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GST_LOG_OBJECT (payload,
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"Calculated min_payload_len %u and max_payload_len %u",
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min_payload_len, max_payload_len);
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adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
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if (GST_BUFFER_IS_DISCONT (buf))
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rtpg729pay->discont = TRUE;
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adapter = rtpg729pay->adapter;
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/* let's reset the base timestamp when the adapter is empty */
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if (gst_adapter_available (adapter) == 0)
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basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
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rtpg729pay->next_ts = GST_BUFFER_TIMESTAMP (buf);
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if (gst_adapter_available (adapter) == 0 &&
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GST_BUFFER_SIZE (buf) >= min_payload_len &&
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GST_BUFFER_SIZE (buf) <= max_payload_len) {
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ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
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ret = gst_rtp_g729_pay_push (rtpg729pay,
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GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
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GST_BUFFER_TIMESTAMP (buf));
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GST_BUFFER_TIMESTAMP (buf), GST_BUFFER_DURATION (buf));
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gst_buffer_unref (buf);
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g_object_unref (adapter);
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return ret;
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}
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@ -236,7 +297,7 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
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/* this loop will push all available buffers till the last frame */
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while (available >= min_payload_len ||
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available % G729_FRAME_SIZE == G729B_CN_FRAME_SIZE) {
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guint num;
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GstClockTime duration;
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/* We send as much as we can */
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if (available <= max_payload_len) {
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@ -246,17 +307,16 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
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(available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
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}
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ret = gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
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basertpaudiopayload->base_ts);
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duration = (payload_len / G729_FRAME_SIZE) * G729_FRAME_DURATION;
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rtpg729pay->next_ts += duration;
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num = payload_len / G729_FRAME_SIZE;
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basertpaudiopayload->base_ts += G729_FRAME_DURATION * num;
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ret = gst_rtp_g729_pay_push (rtpg729pay,
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gst_adapter_take (adapter, payload_len), payload_len,
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rtpg729pay->next_ts, duration);
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available = gst_adapter_available (adapter);
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}
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g_object_unref (adapter);
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return ret;
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/* ERRORS */
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@ -272,6 +332,31 @@ invalid_size:
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}
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}
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static GstStateChangeReturn
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gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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/* handle upwards state changes here */
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switch (transition) {
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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/* handle downwards state changes */
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_rtp_g729_pay_reset (GST_RTP_G729_PAY (element));
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break;
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default:
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break;
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}
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return ret;
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}
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gboolean
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gst_rtp_g729_pay_plugin_init (GstPlugin * plugin)
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{
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@ -43,6 +43,10 @@ typedef struct _GstRTPG729PayClass GstRTPG729PayClass;
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struct _GstRTPG729Pay
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{
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GstBaseRTPAudioPayload audiopayload;
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GstAdapter *adapter;
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GstClockTime next_ts;
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gboolean discont;
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};
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struct _GstRTPG729PayClass
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