static gboolean
gst_rtp_base_payload_audio_sink_event (GstRTPBasePayload * basep,
GstEvent * event)
{
GstRTPBaseAudioPayload *payload;
gboolean res = FALSE;
payload = GST_RTP_BASE_AUDIO_PAYLOAD (basep);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* flush remaining bytes in the adapter */
gst_rtp_base_audio_payload_flush (payload, -1, -1);
break;
case GST_EVENT_FLUSH_STOP:
gst_adapter_clear (payload->priv->adapter);
break;
default:
break;
}
/* let parent handle the remainder of the event */
res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (basep, event);
return res;
}
static GstStateChangeReturn
gst_rtp_base_payload_audio_change_state (GstElement * element,
GstStateChange transition)
{
GstRTPBaseAudioPayload *rtpbasepayload;
GstStateChangeReturn ret;
rtpbasepayload = GST_RTP_BASE_AUDIO_PAYLOAD (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtpbasepayload->priv->cached_mtu = -1;
rtpbasepayload->priv->last_rtptime = -1;
rtpbasepayload->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 (rtpbasepayload->priv->adapter);
break;
default:
break;
}
return ret;
}
static void
gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL16pay);
/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
static gboolean
gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpL16Pay *rtpL16pay;
gboolean res;
gchar *params;
GstAudioInfo *info;
const GstRTPChannelOrder *order;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
rtpL16pay = GST_RTP_L16_PAY (basepayload);
info = &rtpL16pay->info;
gst_audio_info_init (info);
if (!gst_audio_info_from_caps (info, caps))
goto invalid_caps;
order = gst_rtp_channels_get_by_pos (info->channels, info->position);
rtpL16pay->order = order;
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16",
info->rate);
params = g_strdup_printf ("%d", info->channels);
if (!order && info->channels > 2) {
GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", info->channels));
}
if (order && order->name) {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
info->channels, NULL);
}
g_free (params);
/* octet-per-sample is 2 * channels for L16 */
gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
2 * info->channels);
return res;
/* ERRORS */
invalid_caps:
{
GST_DEBUG_OBJECT (rtpL16pay, "invalid caps");
return FALSE;
}
}
static void
gst_rtp_base_audio_payload_finalize (GObject * object)
{
GstRTPBaseAudioPayload *payload;
payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
g_object_unref (payload->priv->adapter);
GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
}
static void
gst_rtp_bv_pay_init (GstRTPBVPay * rtpbvpay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbvpay);
rtpbvpay->mode = -1;
/* tell rtpbaseaudiopayload that this is a frame based codec */
gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
}
static void
gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg722pay);
GST_RTP_BASE_PAYLOAD (rtpg722pay)->pt = GST_RTP_PAYLOAD_G722;
/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
static void
gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmupay);
GST_RTP_BASE_PAYLOAD (rtppcmupay)->clock_rate = 8000;
/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
/* octet-per-sample is 1 for PCM */
gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, 1);
}
static void
gst_rtp_siren_pay_init (GstRTPSirenPay * rtpsirenpay)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpsirenpay);
rtpbaseaudiopayload = GST_RTP_BASE_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 */
rtpbasepayload->clock_rate = 16000;
/* tell rtpbaseaudiopayload that this is a frame based codec */
gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
}
static void
gst_rtp_base_audio_payload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRTPBaseAudioPayload *payload;
payload = GST_RTP_BASE_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 gboolean
gst_rtp_siren_pay_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
{
GstRTPSirenPay *rtpsirenpay;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
gint dct_length;
GstStructure *structure;
const char *payload_name;
rtpsirenpay = GST_RTP_SIREN_PAY (rtpbasepayload);
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
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_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "SIREN",
16000);
/* set options for this frame based audio codec */
gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, 20, 40);
return gst_rtp_base_payload_set_outcaps (rtpbasepayload, 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_bv_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
{
GstRTPBVPay *rtpbvpay;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
gint mode;
GstStructure *structure;
const char *payload_name;
rtpbvpay = GST_RTP_BV_PAY (rtpbasepayload);
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
structure = gst_caps_get_structure (caps, 0);
payload_name = gst_structure_get_name (structure);
if (g_ascii_strcasecmp ("audio/x-bv", payload_name))
goto wrong_caps;
if (!gst_structure_get_int (structure, "mode", &mode))
goto no_mode;
if (mode != 16 && mode != 32)
goto wrong_mode;
if (mode == 16) {
gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV16",
8000);
rtpbasepayload->clock_rate = 8000;
} else {
gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV32",
16000);
rtpbasepayload->clock_rate = 16000;
}
/* set options for this frame based audio codec */
gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
mode, mode == 16 ? 10 : 20);
if (mode != rtpbvpay->mode && rtpbvpay->mode != -1)
goto mode_changed;
rtpbvpay->mode = mode;
return TRUE;
/* ERRORS */
wrong_caps:
{
GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s",
payload_name);
return FALSE;
}
no_mode:
{
GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode");
return FALSE;
}
wrong_mode:
{
GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode);
return FALSE;
}
mode_changed:
{
GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! "
"Mode cannot change while streaming", rtpbvpay->mode, mode);
return FALSE;
}
}
static gboolean
gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpG722Pay *rtpg722pay;
GstStructure *structure;
gint rate, channels, clock_rate;
gboolean res;
gchar *params;
#if 0
GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
#endif
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_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;
/* FIXME: Do something with the channel positions */
#if 0
/* 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;
#endif
/* Clock rate is always 8000 Hz for G722 according to
* RFC 3551 although the sampling rate is 16000 Hz */
clock_rate = 8000;
gst_rtp_base_payload_set_options (basepayload, "audio",
basepayload->pt != GST_RTP_PAYLOAD_G722, "G722", clock_rate);
params = g_strdup_printf ("%d", channels);
#if 0
if (!order && channels > 2) {
GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
}
if (order && order->name) {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
#endif
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
#if 0
}
#endif
g_free (params);
#if 0
g_free (pos);
#endif
rtpg722pay->rate = rate;
rtpg722pay->channels = channels;
/* bits-per-sample is 4 * channels for G722, but as the RTP clock runs at
* half speed (8 instead of 16 khz), pretend it's 8 bits per sample
* channels. */
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
8 * 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;
}
}
static gboolean
gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload,
GstCaps * caps)
{
GstRTPILBCPay *rtpilbcpay;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
gboolean ret;
gint mode;
gchar *mode_str;
GstStructure *structure;
const char *payload_name;
rtpilbcpay = GST_RTP_ILBC_PAY (rtpbasepayload);
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
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_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "ILBC",
8000);
/* set options for this frame based audio codec */
gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
mode, mode == 30 ? 50 : 38);
mode_str = g_strdup_printf ("%d", mode);
ret =
gst_rtp_base_payload_set_outcaps (rtpbasepayload, "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;
}
}
