| /* GStreamer |
| * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| /** |
| * SECTION:element-rtpL16pay |
| * @see_also: rtpL16depay |
| * |
| * Payload raw audio into RTP packets according to RFC 3551. |
| * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt |
| * |
| * <refsect2> |
| * <title>Example pipeline</title> |
| * |[ |
| * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink |
| * ]| This example pipeline will payload raw audio. Refer to |
| * the rtpL16depay example to depayload and play the RTP stream. |
| * </refsect2> |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include "config.h" |
| #endif |
| |
| #include <string.h> |
| |
| #include <gst/audio/audio.h> |
| #include <gst/rtp/gstrtpbuffer.h> |
| |
| #include "gstrtpL16pay.h" |
| #include "gstrtpchannels.h" |
| |
| GST_DEBUG_CATEGORY_STATIC (rtpL16pay_debug); |
| #define GST_CAT_DEFAULT (rtpL16pay_debug) |
| |
| static GstStaticPadTemplate gst_rtp_L16_pay_sink_template = |
| GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-raw, " |
| "format = (string) S16BE, " |
| "layout = (string) interleaved, " |
| "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") |
| ); |
| |
| static GstStaticPadTemplate gst_rtp_L16_pay_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("application/x-rtp, " |
| "media = (string) \"audio\", " |
| "payload = (int) [ 96, 127 ], " |
| "clock-rate = (int) [ 1, MAX ], " |
| "encoding-name = (string) \"L16\", " |
| "channels = (int) [ 1, MAX ];" |
| "application/x-rtp, " |
| "media = (string) \"audio\", " |
| "encoding-name = (string) \"L16\", " |
| "payload = (int) " GST_RTP_PAYLOAD_L16_STEREO_STRING ", " |
| "clock-rate = (int) 44100;" |
| "application/x-rtp, " |
| "media = (string) \"audio\", " |
| "encoding-name = (string) \"L16\", " |
| "payload = (int) " GST_RTP_PAYLOAD_L16_MONO_STRING ", " |
| "clock-rate = (int) 44100") |
| ); |
| |
| static gboolean gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload, |
| GstCaps * caps); |
| static GstCaps *gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, |
| GstPad * pad, GstCaps * filter); |
| static GstFlowReturn |
| gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload, |
| GstBuffer * buffer); |
| |
| #define gst_rtp_L16_pay_parent_class parent_class |
| G_DEFINE_TYPE (GstRtpL16Pay, gst_rtp_L16_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD); |
| |
| static void |
| gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass) |
| { |
| GstElementClass *gstelement_class; |
| GstRTPBasePayloadClass *gstrtpbasepayload_class; |
| |
| gstelement_class = (GstElementClass *) klass; |
| gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; |
| |
| gstrtpbasepayload_class->set_caps = gst_rtp_L16_pay_setcaps; |
| gstrtpbasepayload_class->get_caps = gst_rtp_L16_pay_getcaps; |
| gstrtpbasepayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer; |
| |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_L16_pay_src_template); |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_L16_pay_sink_template); |
| |
| gst_element_class_set_static_metadata (gstelement_class, |
| "RTP audio payloader", "Codec/Payloader/Network/RTP", |
| "Payload-encode Raw audio into RTP packets (RFC 3551)", |
| "Wim Taymans <wim.taymans@gmail.com>"); |
| |
| GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0, |
| "L16 RTP Payloader"); |
| } |
| |
| 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 GstCaps * |
| gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad, |
| GstCaps * filter) |
| { |
| GstCaps *otherpadcaps; |
| GstCaps *caps; |
| |
| caps = gst_pad_get_pad_template_caps (pad); |
| |
| otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad); |
| if (otherpadcaps) { |
| if (!gst_caps_is_empty (otherpadcaps)) { |
| GstStructure *structure; |
| gint channels; |
| gint pt; |
| gint rate; |
| |
| structure = gst_caps_get_structure (otherpadcaps, 0); |
| caps = gst_caps_make_writable (caps); |
| |
| if (gst_structure_get_int (structure, "channels", &channels)) { |
| gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL); |
| } else if (gst_structure_get_int (structure, "payload", &pt)) { |
| if (pt == GST_RTP_PAYLOAD_L16_STEREO) |
| gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL); |
| else if (pt == GST_RTP_PAYLOAD_L16_MONO) |
| gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL); |
| } |
| |
| if (gst_structure_get_int (structure, "clock-rate", &rate)) { |
| gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL); |
| } else if (gst_structure_get_int (structure, "payload", &pt)) { |
| if (pt == GST_RTP_PAYLOAD_L16_STEREO || pt == GST_RTP_PAYLOAD_L16_MONO) |
| gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL); |
| } |
| |
| } |
| gst_caps_unref (otherpadcaps); |
| } |
| |
| if (filter) { |
| GstCaps *tcaps = caps; |
| |
| caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST); |
| gst_caps_unref (tcaps); |
| } |
| |
| return caps; |
| } |
| |
| static GstFlowReturn |
| gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload, |
| GstBuffer * buffer) |
| { |
| GstRtpL16Pay *rtpL16pay; |
| |
| rtpL16pay = GST_RTP_L16_PAY (basepayload); |
| buffer = gst_buffer_make_writable (buffer); |
| |
| if (rtpL16pay->order && |
| !gst_audio_buffer_reorder_channels (buffer, rtpL16pay->info.finfo->format, |
| rtpL16pay->info.channels, rtpL16pay->info.position, |
| rtpL16pay->order->pos)) { |
| return GST_FLOW_ERROR; |
| } |
| |
| return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload, |
| buffer); |
| } |
| |
| gboolean |
| gst_rtp_L16_pay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpL16pay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_L16_PAY); |
| } |