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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
rtpg723pay: rewrite payloader
Handle all 3 packet sizes according to RFC 3551. Totally untested, we don't have a G723 encoder. Fixes #605882
This commit is contained in:
Wim Taymans
parent
48615d5e98
commit
d6d06630e8
2 changed files with 181 additions and 160 deletions
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@ -19,12 +19,6 @@
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* Boston, MA 02111-1307, USA.
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*/
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/*
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* This payloader assumes that the data will ALWAYS come as zero or more
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* 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
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* Any other buffer format won't work
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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@ -38,23 +32,18 @@
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#define GST_RTP_PAYLOAD_G723 4
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#define GST_RTP_PAYLOAD_G723_STRING "4"
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/* According to RFC 3551, works only with G723 encoded with 6.3 kb/s high-rate */
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#define G723_FRAME_SIZE 24
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#define G723B_SID_FRAME_SIZE 4
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#define G723_FRAME_DURATION (30 * GST_MSECOND)
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#define G723_FRAME_DURATION_MS (30)
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static gboolean
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gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
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static GstFlowReturn
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gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
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static gboolean gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buf);
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static const GstElementDetails gst_rtp_g723_pay_details =
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GST_ELEMENT_DETAILS ("RTP G.723 payloader",
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"Codec/Payloader/Network",
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"Packetize 6.3kb/s G.723 audio into RTP packets",
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"Tiago Katcipis <tiago.katcipis@digitro.com.br>");
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"Packetize G.723 audio into RTP packets",
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"Wim Taymans <wim.taymans@gmail.com>");
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static GstStaticPadTemplate gst_rtp_g723_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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@ -79,11 +68,15 @@ static GstStaticPadTemplate gst_rtp_g723_pay_src_template =
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"clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
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);
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static void
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gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass);
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static void gst_rtp_g723_pay_init (GstRTPG723Pay * pay,
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GstRTPG723PayClass * klass);
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static void gst_rtp_g723_pay_finalize (GObject * object);
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GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static GstStateChangeReturn gst_rtp_g723_pay_change_state (GstElement * element,
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GstStateChange transition);
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GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_g723_pay_base_init (gpointer klass)
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@ -100,7 +93,17 @@ gst_rtp_g723_pay_base_init (gpointer klass)
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static void
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gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
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{
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GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *payload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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payload_class = (GstBaseRTPPayloadClass *) klass;
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gobject_class->finalize = gst_rtp_g723_pay_finalize;
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gstelement_class->change_state = gst_rtp_g723_pay_change_state;
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payload_class->set_caps = gst_rtp_g723_pay_set_caps;
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payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
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@ -110,17 +113,27 @@ static void
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gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * 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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pay->adapter = gst_adapter_new ();
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payload->pt = GST_RTP_PAYLOAD_G723;
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gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 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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G723_FRAME_DURATION_MS, G723_FRAME_SIZE);
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}
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static void
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gst_rtp_g723_pay_finalize (GObject * object)
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{
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GstRTPG723Pay *pay;
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pay = GST_RTP_G723_PAY (object);
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g_object_unref (pay->adapter);
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pay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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@ -140,160 +153,163 @@ gst_rtp_g723_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_g723_pay_flush (GstRTPG723Pay * pay)
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{
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GstBuffer *outbuf;
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GstFlowReturn ret;
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guint8 *payload;
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guint avail;
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avail = gst_adapter_available (pay->adapter);
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outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
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payload = gst_rtp_buffer_get_payload (outbuf);
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GST_BUFFER_TIMESTAMP (outbuf) = pay->timestamp;
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GST_BUFFER_DURATION (outbuf) = pay->duration;
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/* copy G723 data as payload */
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gst_adapter_copy (pay->adapter, payload, 0, avail);
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/* flush bytes from adapter */
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gst_adapter_flush (pay->adapter, avail);
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pay->timestamp = GST_CLOCK_TIME_NONE;
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pay->duration = 0;
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/* set discont and marker */
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if (pay->discont) {
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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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pay->discont = FALSE;
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}
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
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return ret;
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}
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/* 00 high-rate speech (6.3 kb/s) 24
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* 01 low-rate speech (5.3 kb/s) 20
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* 10 SID frame 4
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* 11 reserved 0 */
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static const guint size_tab[4] = {
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24, 20, 4, 0
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};
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static GstFlowReturn
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gst_rtp_g723_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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GstAdapter *adapter = NULL;
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guint payload_len;
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const guint8 *data = NULL;
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guint available;
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guint maxptime_octets = G_MAXUINT;
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guint minptime_octets = 0;
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guint min_payload_len;
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guint max_payload_len;
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gboolean use_adapter = FALSE;
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guint8 *data;
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guint size;
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guint8 HDR;
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GstRTPG723Pay *pay;
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GstClockTime packet_dur, timestamp;
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guint payload_len, packet_len;
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available = GST_BUFFER_SIZE (buf);
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pay = GST_RTP_G723_PAY (payload);
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if (available % G723_FRAME_SIZE != 0 &&
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available % G723_FRAME_SIZE != G723B_SID_FRAME_SIZE)
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size = GST_BUFFER_SIZE (buf);
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data = GST_BUFFER_DATA (buf);
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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if (GST_BUFFER_IS_DISCONT (buf)) {
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/* flush everything on discont */
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gst_adapter_clear (pay->adapter);
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pay->timestamp = GST_CLOCK_TIME_NONE;
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pay->duration = 0;
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pay->discont = TRUE;
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}
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/* should be one of these sizes */
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if (size != 4 && size != 20 && size != 24)
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goto invalid_size;
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/* max number of bytes based on given ptime, has to be multiple of
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* frame_duration */
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if (payload->max_ptime != -1) {
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guint ptime_ms = payload->max_ptime / 1000000;
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/* check size by looking at the header bits */
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HDR = data[0] & 0x3;
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if (size_tab[HDR] != size)
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goto wrong_size;
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maxptime_octets = G723_FRAME_SIZE *
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(int) (ptime_ms / G723_FRAME_DURATION_MS);
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/* calculate packet size and duration */
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payload_len = gst_adapter_available (pay->adapter) + size;
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packet_dur = pay->duration + G723_FRAME_DURATION;
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packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
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if (maxptime_octets < G723_FRAME_SIZE) {
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GST_WARNING_OBJECT (basertpaudiopayload, "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, G723_FRAME_DURATION_MS);
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maxptime_octets = G723_FRAME_SIZE;
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}
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if (gst_basertppayload_is_filled (payload, packet_len, packet_dur)) {
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/* size or duration would overflow the packet, flush the queued data */
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ret = gst_rtp_g723_pay_flush (pay);
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}
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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) / G723_FRAME_SIZE) * G723_FRAME_SIZE,
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/* ptime max */
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maxptime_octets);
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/* min number of bytes based on a given ptime, has to be a multiple
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of frame duration */
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{
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guint64 min_ptime;
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g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL);
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min_ptime = min_ptime / 1000000;
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minptime_octets = G723_FRAME_SIZE *
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(int) (min_ptime / G723_FRAME_DURATION_MS);
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/* update timestamp, we keep the timestamp for the first packet in the adapter
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* but are able to calculate it from next packets. */
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if (timestamp != GST_CLOCK_TIME_NONE && pay->timestamp == GST_CLOCK_TIME_NONE) {
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if (timestamp > pay->duration)
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pay->timestamp = timestamp - pay->duration;
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else
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pay->timestamp = 0;
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}
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min_payload_len = MAX (minptime_octets, G723_FRAME_SIZE);
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/* add packet to the queue */
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gst_adapter_push (pay->adapter, buf);
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pay->duration = packet_dur;
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if (min_payload_len > max_payload_len) {
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min_payload_len = max_payload_len;
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}
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GST_DEBUG_OBJECT (basertpaudiopayload,
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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 (adapter && gst_adapter_available (adapter)) {
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/* If there is always data in the adapter, we have to use it */
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gst_adapter_push (adapter, buf);
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available = gst_adapter_available (adapter);
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use_adapter = TRUE;
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} else {
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/* let's set the base timestamp */
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basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
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/* If buffer fits on an RTP packet, let's just push it through */
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/* this will check against max_ptime and max_mtu */
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if (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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GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
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GST_BUFFER_TIMESTAMP (buf));
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gst_buffer_unref (buf);
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return ret;
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}
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available = GST_BUFFER_SIZE (buf);
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data = (guint8 *) GST_BUFFER_DATA (buf);
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}
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/* as long as we have full frames */
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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 % G723_FRAME_SIZE == G723B_SID_FRAME_SIZE) {
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guint num;
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/* We send as much as we can */
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if (available <= max_payload_len) {
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payload_len = available;
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} else {
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payload_len = MIN (max_payload_len,
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(available / G723_FRAME_SIZE) * G723_FRAME_SIZE);
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}
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if (use_adapter) {
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data = gst_adapter_peek (adapter, payload_len);
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}
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ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
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payload_len, basertpaudiopayload->base_ts);
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num = payload_len / G723_FRAME_SIZE;
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basertpaudiopayload->base_ts += G723_FRAME_DURATION * num;
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if (use_adapter) {
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gst_adapter_flush (adapter, payload_len);
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available = gst_adapter_available (adapter);
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} else {
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available -= payload_len;
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data += payload_len;
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}
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}
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if (!use_adapter) {
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if (available != 0 && adapter) {
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GstBuffer *buf2;
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buf2 = gst_buffer_create_sub (buf,
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GST_BUFFER_SIZE (buf) - available, available);
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gst_adapter_push (adapter, buf2);
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} else {
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gst_buffer_unref (buf);
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}
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}
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if (adapter) {
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g_object_unref (adapter);
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/* check if we can flush now */
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if (pay->duration >= payload->min_ptime) {
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ret = gst_rtp_g723_pay_flush (pay);
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}
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return ret;
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/* ERRORS */
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/* WARNINGS */
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invalid_size:
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{
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GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
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GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
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("Invalid input buffer size"),
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("Invalid buffer size, should be a multiple of"
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" G723_FRAME_SIZE(24) with an optional G723B_SID_FRAME_SIZE(4)"
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" added to it, but it is %u", available));
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("Input size should be 4, 20 or 24, got %u", size));
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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return GST_FLOW_OK;
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}
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wrong_size:
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{
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GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
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("Wrong input buffer size"),
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("Expected input buffer size %u but got %u", size_tab[HDR], size));
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gst_buffer_unref (buf);
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return GST_FLOW_OK;
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}
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}
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static GstStateChangeReturn
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gst_rtp_g723_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstRTPG723Pay *pay;
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pay = GST_RTP_G723_PAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_adapter_clear (pay->adapter);
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pay->timestamp = GST_CLOCK_TIME_NONE;
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pay->duration = 0;
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pay->discont = TRUE;
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break;
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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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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_adapter_clear (pay->adapter);
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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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/*Plugin init functions*/
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@ -42,12 +42,17 @@ typedef struct _GstRTPG723PayClass GstRTPG723PayClass;
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struct _GstRTPG723Pay
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{
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GstBaseRTPAudioPayload audiopayload;
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GstBaseRTPPayload payload;
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GstAdapter *adapter;
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GstClockTime duration;
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GstClockTime timestamp;
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gboolean discont;
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};
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struct _GstRTPG723PayClass
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{
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GstBaseRTPAudioPayloadClass parent_class;
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GstBaseRTPPayloadClass parent_class;
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};
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gboolean gst_rtp_g723_pay_plugin_init (GstPlugin * plugin);
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