| /* GStreamer |
| * Copyright (C) <2009> 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. |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include "config.h" |
| #endif |
| |
| #include <stdlib.h> |
| #include <string.h> |
| #include <gst/rtp/gstrtpbuffer.h> |
| #include <gst/audio/audio.h> |
| |
| #include "gstrtpceltpay.h" |
| #include "gstrtputils.h" |
| |
| GST_DEBUG_CATEGORY_STATIC (rtpceltpay_debug); |
| #define GST_CAT_DEFAULT (rtpceltpay_debug) |
| |
| static GstStaticPadTemplate gst_rtp_celt_pay_sink_template = |
| GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-celt, " |
| "rate = (int) [ 32000, 64000 ], " |
| "channels = (int) [1, 2], " "frame-size = (int) [ 64, 512 ]") |
| ); |
| |
| static GstStaticPadTemplate gst_rtp_celt_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) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " |
| "clock-rate = (int) [ 32000, 48000 ], " |
| "encoding-name = (string) \"CELT\"") |
| ); |
| |
| static void gst_rtp_celt_pay_finalize (GObject * object); |
| |
| static GstStateChangeReturn gst_rtp_celt_pay_change_state (GstElement * |
| element, GstStateChange transition); |
| |
| static gboolean gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload, |
| GstCaps * caps); |
| static GstCaps *gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload, |
| GstPad * pad, GstCaps * filter); |
| static GstFlowReturn gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * |
| payload, GstBuffer * buffer); |
| |
| #define gst_rtp_celt_pay_parent_class parent_class |
| G_DEFINE_TYPE (GstRtpCELTPay, gst_rtp_celt_pay, GST_TYPE_RTP_BASE_PAYLOAD); |
| |
| static void |
| gst_rtp_celt_pay_class_init (GstRtpCELTPayClass * klass) |
| { |
| GObjectClass *gobject_class; |
| GstElementClass *gstelement_class; |
| GstRTPBasePayloadClass *gstrtpbasepayload_class; |
| |
| GST_DEBUG_CATEGORY_INIT (rtpceltpay_debug, "rtpceltpay", 0, |
| "CELT RTP Payloader"); |
| |
| gobject_class = (GObjectClass *) klass; |
| gstelement_class = (GstElementClass *) klass; |
| gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; |
| |
| gobject_class->finalize = gst_rtp_celt_pay_finalize; |
| |
| gstelement_class->change_state = gst_rtp_celt_pay_change_state; |
| |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_celt_pay_sink_template); |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_celt_pay_src_template); |
| |
| gst_element_class_set_static_metadata (gstelement_class, "RTP CELT payloader", |
| "Codec/Payloader/Network/RTP", |
| "Payload-encodes CELT audio into a RTP packet", |
| "Wim Taymans <wim.taymans@gmail.com>"); |
| |
| gstrtpbasepayload_class->set_caps = gst_rtp_celt_pay_setcaps; |
| gstrtpbasepayload_class->get_caps = gst_rtp_celt_pay_getcaps; |
| gstrtpbasepayload_class->handle_buffer = gst_rtp_celt_pay_handle_buffer; |
| } |
| |
| static void |
| gst_rtp_celt_pay_init (GstRtpCELTPay * rtpceltpay) |
| { |
| rtpceltpay->queue = g_queue_new (); |
| } |
| |
| static void |
| gst_rtp_celt_pay_finalize (GObject * object) |
| { |
| GstRtpCELTPay *rtpceltpay; |
| |
| rtpceltpay = GST_RTP_CELT_PAY (object); |
| |
| g_queue_free (rtpceltpay->queue); |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| static void |
| gst_rtp_celt_pay_clear_queued (GstRtpCELTPay * rtpceltpay) |
| { |
| GstBuffer *buf; |
| |
| while ((buf = g_queue_pop_head (rtpceltpay->queue))) |
| gst_buffer_unref (buf); |
| |
| rtpceltpay->bytes = 0; |
| rtpceltpay->sbytes = 0; |
| rtpceltpay->qduration = 0; |
| } |
| |
| static void |
| gst_rtp_celt_pay_add_queued (GstRtpCELTPay * rtpceltpay, GstBuffer * buffer, |
| guint ssize, guint size, GstClockTime duration) |
| { |
| g_queue_push_tail (rtpceltpay->queue, buffer); |
| rtpceltpay->sbytes += ssize; |
| rtpceltpay->bytes += size; |
| /* only add durations when we have a valid previous duration */ |
| if (rtpceltpay->qduration != -1) { |
| if (duration != -1) |
| /* only add valid durations */ |
| rtpceltpay->qduration += duration; |
| else |
| /* if we add a buffer without valid duration, our total queued duration |
| * becomes unknown */ |
| rtpceltpay->qduration = -1; |
| } |
| } |
| |
| static gboolean |
| gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) |
| { |
| /* don't configure yet, we wait for the ident packet */ |
| return TRUE; |
| } |
| |
| |
| static GstCaps * |
| gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad, |
| GstCaps * filter) |
| { |
| GstCaps *otherpadcaps; |
| GstCaps *caps; |
| const gchar *params; |
| |
| caps = gst_pad_get_pad_template_caps (pad); |
| |
| otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad); |
| if (otherpadcaps) { |
| if (!gst_caps_is_empty (otherpadcaps)) { |
| GstStructure *ps; |
| GstStructure *s; |
| gint clock_rate = 0, frame_size = 0, channels = 1; |
| |
| caps = gst_caps_make_writable (caps); |
| |
| ps = gst_caps_get_structure (otherpadcaps, 0); |
| s = gst_caps_get_structure (caps, 0); |
| |
| if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) { |
| gst_structure_fixate_field_nearest_int (s, "rate", clock_rate); |
| } |
| |
| if ((params = gst_structure_get_string (ps, "frame-size"))) |
| frame_size = atoi (params); |
| if (frame_size) |
| gst_structure_set (s, "frame-size", G_TYPE_INT, frame_size, NULL); |
| |
| if ((params = gst_structure_get_string (ps, "encoding-params"))) { |
| channels = atoi (params); |
| gst_structure_fixate_field_nearest_int (s, "channels", channels); |
| } |
| |
| GST_DEBUG_OBJECT (payload, "clock-rate=%d frame-size=%d channels=%d", |
| clock_rate, frame_size, channels); |
| } |
| gst_caps_unref (otherpadcaps); |
| } |
| |
| if (filter) { |
| GstCaps *tmp; |
| |
| GST_DEBUG_OBJECT (payload, "Intersect %" GST_PTR_FORMAT " and filter %" |
| GST_PTR_FORMAT, caps, filter); |
| tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); |
| gst_caps_unref (caps); |
| caps = tmp; |
| } |
| |
| return caps; |
| } |
| |
| static gboolean |
| gst_rtp_celt_pay_parse_ident (GstRtpCELTPay * rtpceltpay, |
| const guint8 * data, guint size) |
| { |
| guint32 version, header_size, rate, nb_channels, frame_size, overlap; |
| guint32 bytes_per_packet; |
| GstRTPBasePayload *payload; |
| gchar *cstr, *fsstr; |
| gboolean res; |
| |
| /* we need the header string (8), the version string (20), the version |
| * and the header length. */ |
| if (size < 36) |
| goto too_small; |
| |
| if (!g_str_has_prefix ((const gchar *) data, "CELT ")) |
| goto wrong_header; |
| |
| /* skip header and version string */ |
| data += 28; |
| |
| version = GST_READ_UINT32_LE (data); |
| GST_DEBUG_OBJECT (rtpceltpay, "version %08x", version); |
| #if 0 |
| if (version != 1) |
| goto wrong_version; |
| #endif |
| |
| data += 4; |
| /* ensure sizes */ |
| header_size = GST_READ_UINT32_LE (data); |
| if (header_size < 56) |
| goto header_too_small; |
| |
| if (size < header_size) |
| goto payload_too_small; |
| |
| data += 4; |
| rate = GST_READ_UINT32_LE (data); |
| data += 4; |
| nb_channels = GST_READ_UINT32_LE (data); |
| data += 4; |
| frame_size = GST_READ_UINT32_LE (data); |
| data += 4; |
| overlap = GST_READ_UINT32_LE (data); |
| data += 4; |
| bytes_per_packet = GST_READ_UINT32_LE (data); |
| |
| GST_DEBUG_OBJECT (rtpceltpay, "rate %d, nb_channels %d, frame_size %d", |
| rate, nb_channels, frame_size); |
| GST_DEBUG_OBJECT (rtpceltpay, "overlap %d, bytes_per_packet %d", |
| overlap, bytes_per_packet); |
| |
| payload = GST_RTP_BASE_PAYLOAD (rtpceltpay); |
| |
| gst_rtp_base_payload_set_options (payload, "audio", FALSE, "CELT", rate); |
| cstr = g_strdup_printf ("%d", nb_channels); |
| fsstr = g_strdup_printf ("%d", frame_size); |
| res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params", |
| G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL); |
| g_free (cstr); |
| g_free (fsstr); |
| |
| return res; |
| |
| /* ERRORS */ |
| too_small: |
| { |
| GST_DEBUG_OBJECT (rtpceltpay, |
| "ident packet too small, need at least 32 bytes"); |
| return FALSE; |
| } |
| wrong_header: |
| { |
| GST_DEBUG_OBJECT (rtpceltpay, |
| "ident packet does not start with \"CELT \""); |
| return FALSE; |
| } |
| #if 0 |
| wrong_version: |
| { |
| GST_DEBUG_OBJECT (rtpceltpay, "can only handle version 1, have version %d", |
| version); |
| return FALSE; |
| } |
| #endif |
| header_too_small: |
| { |
| GST_DEBUG_OBJECT (rtpceltpay, |
| "header size too small, need at least 80 bytes, " "got only %d", |
| header_size); |
| return FALSE; |
| } |
| payload_too_small: |
| { |
| GST_DEBUG_OBJECT (rtpceltpay, |
| "payload too small, need at least %d bytes, got only %d", header_size, |
| size); |
| return FALSE; |
| } |
| } |
| |
| static GstFlowReturn |
| gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay) |
| { |
| GstFlowReturn ret; |
| GstBuffer *buf, *outbuf; |
| guint8 *payload, *spayload; |
| guint payload_len; |
| GstClockTime duration; |
| GstRTPBuffer rtp = { NULL, }; |
| |
| payload_len = rtpceltpay->bytes + rtpceltpay->sbytes; |
| duration = rtpceltpay->qduration; |
| |
| GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT, |
| payload_len, GST_TIME_ARGS (rtpceltpay->qduration)); |
| |
| /* get a big enough packet for the sizes + payloads */ |
| outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); |
| |
| GST_BUFFER_DURATION (outbuf) = duration; |
| |
| gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); |
| |
| /* point to the payload for size headers and data */ |
| spayload = gst_rtp_buffer_get_payload (&rtp); |
| payload = spayload + rtpceltpay->sbytes; |
| |
| while ((buf = g_queue_pop_head (rtpceltpay->queue))) { |
| guint size; |
| |
| /* copy first timestamp to output */ |
| if (GST_BUFFER_PTS (outbuf) == -1) |
| GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buf); |
| |
| /* write the size to the header */ |
| size = gst_buffer_get_size (buf); |
| while (size > 0xff) { |
| *spayload++ = 0xff; |
| size -= 0xff; |
| } |
| *spayload++ = size; |
| |
| /* copy payload */ |
| size = gst_buffer_get_size (buf); |
| gst_buffer_extract (buf, 0, payload, size); |
| payload += size; |
| |
| gst_rtp_copy_audio_meta (rtpceltpay, outbuf, buf); |
| |
| gst_buffer_unref (buf); |
| } |
| gst_rtp_buffer_unmap (&rtp); |
| |
| /* we consumed it all */ |
| rtpceltpay->bytes = 0; |
| rtpceltpay->sbytes = 0; |
| rtpceltpay->qduration = 0; |
| |
| ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf); |
| |
| return ret; |
| } |
| |
| static GstFlowReturn |
| gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * basepayload, |
| GstBuffer * buffer) |
| { |
| GstFlowReturn ret; |
| GstRtpCELTPay *rtpceltpay; |
| gsize payload_len; |
| GstMapInfo map; |
| GstClockTime duration, packet_dur; |
| guint i, ssize, packet_len; |
| |
| rtpceltpay = GST_RTP_CELT_PAY (basepayload); |
| |
| ret = GST_FLOW_OK; |
| |
| gst_buffer_map (buffer, &map, GST_MAP_READ); |
| |
| switch (rtpceltpay->packet) { |
| case 0: |
| /* ident packet. We need to parse the headers to construct the RTP |
| * properties. */ |
| if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, map.data, map.size)) |
| goto parse_error; |
| |
| goto cleanup; |
| case 1: |
| /* comment packet, we ignore it */ |
| goto cleanup; |
| default: |
| /* other packets go in the payload */ |
| break; |
| } |
| gst_buffer_unmap (buffer, &map); |
| |
| duration = GST_BUFFER_DURATION (buffer); |
| |
| GST_LOG_OBJECT (rtpceltpay, |
| "got buffer of duration %" GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT, |
| GST_TIME_ARGS (duration), map.size); |
| |
| /* calculate the size of the size field and the payload */ |
| ssize = 1; |
| for (i = map.size; i > 0xff; i -= 0xff) |
| ssize++; |
| |
| GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize); |
| |
| /* calculate what the new size and duration would be of the packet */ |
| payload_len = ssize + map.size + rtpceltpay->bytes + rtpceltpay->sbytes; |
| if (rtpceltpay->qduration != -1 && duration != -1) |
| packet_dur = rtpceltpay->qduration + duration; |
| else |
| packet_dur = 0; |
| |
| packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0); |
| |
| if (gst_rtp_base_payload_is_filled (basepayload, packet_len, packet_dur)) { |
| /* size or duration would overflow the packet, flush the queued data */ |
| ret = gst_rtp_celt_pay_flush_queued (rtpceltpay); |
| } |
| |
| /* queue the packet */ |
| gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, map.size, duration); |
| |
| done: |
| rtpceltpay->packet++; |
| |
| return ret; |
| |
| /* ERRORS */ |
| cleanup: |
| { |
| gst_buffer_unmap (buffer, &map); |
| goto done; |
| } |
| parse_error: |
| { |
| GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL), |
| ("Error parsing first identification packet.")); |
| gst_buffer_unmap (buffer, &map); |
| return GST_FLOW_ERROR; |
| } |
| } |
| |
| static GstStateChangeReturn |
| gst_rtp_celt_pay_change_state (GstElement * element, GstStateChange transition) |
| { |
| GstRtpCELTPay *rtpceltpay; |
| GstStateChangeReturn ret; |
| |
| rtpceltpay = GST_RTP_CELT_PAY (element); |
| |
| switch (transition) { |
| case GST_STATE_CHANGE_NULL_TO_READY: |
| break; |
| case GST_STATE_CHANGE_READY_TO_PAUSED: |
| rtpceltpay->packet = 0; |
| break; |
| default: |
| break; |
| } |
| |
| ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); |
| |
| switch (transition) { |
| case GST_STATE_CHANGE_PAUSED_TO_READY: |
| gst_rtp_celt_pay_clear_queued (rtpceltpay); |
| break; |
| case GST_STATE_CHANGE_READY_TO_NULL: |
| break; |
| default: |
| break; |
| } |
| return ret; |
| } |
| |
| gboolean |
| gst_rtp_celt_pay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpceltpay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_PAY); |
| } |