static gboolean gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event) { GstBaseRTPAudioPayload *payload; gboolean res = FALSE; payload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: /* flush remaining bytes in the adapter */ gst_base_rtp_audio_payload_flush (payload, -1, -1); break; case GST_EVENT_FLUSH_STOP: gst_adapter_clear (payload->priv->adapter); break; default: break; } gst_object_unref (payload); /* return FALSE to let parent handle the remainder of the event */ return res; }
static GstStateChangeReturn gst_base_rtp_payload_audio_change_state (GstElement * element, GstStateChange transition) { GstBaseRTPAudioPayload *basertppayload; GstStateChangeReturn ret; basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: basertppayload->priv->cached_mtu = -1; basertppayload->priv->last_rtptime = -1; basertppayload->priv->last_timestamp = -1; break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_adapter_clear (basertppayload->priv->adapter); break; default: break; } return ret; }
static void gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay, GstRtpG722PayClass * klass) { GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg722pay); /* tell basertpaudiopayload that this is a sample based codec */ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload); }
static void gst_base_rtp_audio_payload_finalize (GObject * object) { GstBaseRTPAudioPayload *payload; payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); g_object_unref (payload->priv->adapter); GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object)); }
static void gst_rtp_pcma_pay_init (GstRtpPmcaPay * rtppcmapay, GstRtpPmcaPayClass * klass) { GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmapay); GST_BASE_RTP_PAYLOAD (rtppcmapay)->clock_rate = 8000; /* tell basertpaudiopayload that this is a sample based codec */ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload); /* octet-per-sample is 1 for PCM */ gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1); }
static void gst_base_rtp_audio_payload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstBaseRTPAudioPayload *payload; payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); switch (prop_id) { case PROP_BUFFER_LIST: g_value_set_boolean (value, payload->priv->buffer_list); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } }
static void gst_rtp_siren_pay_init (GstRTPSirenPay * rtpsirenpay, GstRTPSirenPayClass * klass) { GstBaseRTPPayload *basertppayload; GstBaseRTPAudioPayload *basertpaudiopayload; basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay); basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay); /* we don't set the payload type, it should be set by the application using * the pt property or the default 96 will be used */ basertppayload->clock_rate = 16000; /* tell basertpaudiopayload that this is a frame based codec */ gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload); }
static gboolean gst_rtp_siren_pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps) { GstRTPSirenPay *rtpsirenpay; GstBaseRTPAudioPayload *basertpaudiopayload; gint dct_length; GstStructure *structure; const char *payload_name; rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload); basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload); structure = gst_caps_get_structure (caps, 0); gst_structure_get_int (structure, "dct-length", &dct_length); if (dct_length != 320) goto wrong_dct; payload_name = gst_structure_get_name (structure); if (g_ascii_strcasecmp ("audio/x-siren", payload_name)) goto wrong_caps; gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN", 16000); /* set options for this frame based audio codec */ gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40); return gst_basertppayload_set_outcaps (basertppayload, NULL); /* ERRORS */ wrong_dct: { GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d", dct_length); return FALSE; } wrong_caps: { GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s", payload_name); return FALSE; } }
static gboolean gst_rtp_ilbc_pay_sink_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps) { GstRTPILBCPay *rtpilbcpay; GstBaseRTPAudioPayload *basertpaudiopayload; gboolean ret; gint mode; gchar *mode_str; GstStructure *structure; const char *payload_name; rtpilbcpay = GST_RTP_ILBC_PAY (basertppayload); basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload); structure = gst_caps_get_structure (caps, 0); payload_name = gst_structure_get_name (structure); if (g_ascii_strcasecmp ("audio/x-iLBC", payload_name)) goto wrong_caps; if (!gst_structure_get_int (structure, "mode", &mode)) goto no_mode; if (mode != 20 && mode != 30) goto wrong_mode; gst_basertppayload_set_options (basertppayload, "audio", TRUE, "ILBC", 8000); /* set options for this frame based audio codec */ gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, mode, mode == 30 ? 50 : 38); mode_str = g_strdup_printf ("%d", mode); ret = gst_basertppayload_set_outcaps (basertppayload, "mode", G_TYPE_STRING, mode_str, NULL); g_free (mode_str); if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1) goto mode_changed; rtpilbcpay->mode = mode; return ret; /* ERRORS */ wrong_caps: { GST_ERROR_OBJECT (rtpilbcpay, "expected audio/x-iLBC, received %s", payload_name); return FALSE; } no_mode: { GST_ERROR_OBJECT (rtpilbcpay, "did not receive a mode"); return FALSE; } wrong_mode: { GST_ERROR_OBJECT (rtpilbcpay, "mode must be 20 or 30, received %d", mode); return FALSE; } mode_changed: { GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! " "Mode cannot change while streaming", rtpilbcpay->mode, mode); return FALSE; } }
static gboolean gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps) { GstRtpG722Pay *rtpg722pay; GstStructure *structure; gint rate, channels, clock_rate; gboolean res; gchar *params; GstAudioChannelPosition *pos; const GstRTPChannelOrder *order; GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); rtpg722pay = GST_RTP_G722_PAY (basepayload); structure = gst_caps_get_structure (caps, 0); /* first parse input caps */ if (!gst_structure_get_int (structure, "rate", &rate)) goto no_rate; if (!gst_structure_get_int (structure, "channels", &channels)) goto no_channels; /* get the channel order */ pos = gst_audio_get_channel_positions (structure); if (pos) order = gst_rtp_channels_get_by_pos (channels, pos); else order = NULL; /* Clock rate is always 8000 Hz for G722 according to * RFC 3551 although the sampling rate is 16000 Hz */ clock_rate = 8000; gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722", clock_rate); params = g_strdup_printf ("%d", channels); if (!order && channels > 2) { GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE, (NULL), ("Unknown channel order for %d channels", channels)); } if (order && order->name) { res = gst_basertppayload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, channels, "channel-order", G_TYPE_STRING, order->name, NULL); } else { res = gst_basertppayload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, channels, NULL); } g_free (params); g_free (pos); rtpg722pay->rate = rate; rtpg722pay->channels = channels; /* octet-per-sample is 1 * channels for G722 */ gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, 4 * rtpg722pay->channels); return res; /* ERRORS */ no_rate: { GST_DEBUG_OBJECT (rtpg722pay, "no rate given"); return FALSE; } no_channels: { GST_DEBUG_OBJECT (rtpg722pay, "no channels given"); return FALSE; } }