| /* GStreamer |
| * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu> |
| * Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br> |
| * Copyright (C) 2005 Nokia Corporation <kai.vehmanen@nokia.com> |
| * Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.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 "gstrtpg726pay.h" |
| |
| GST_DEBUG_CATEGORY_STATIC (rtpg726pay_debug); |
| #define GST_CAT_DEFAULT (rtpg726pay_debug) |
| |
| #define DEFAULT_FORCE_AAL2 TRUE |
| |
| enum |
| { |
| PROP_0, |
| PROP_FORCE_AAL2 |
| }; |
| |
| static GstStaticPadTemplate gst_rtp_g726_pay_sink_template = |
| GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-adpcm, " |
| "channels = (int) 1, " |
| "rate = (int) 8000, " |
| "bitrate = (int) { 16000, 24000, 32000, 40000 }, " |
| "layout = (string) \"g726\"") |
| ); |
| |
| static GstStaticPadTemplate gst_rtp_g726_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) 8000, " |
| "encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\", " |
| " \"AAL2-G726-16\", \"AAL2-G726-24\", \"AAL2-G726-32\", \"AAL2-G726-40\" } ") |
| ); |
| |
| static void gst_rtp_g726_pay_get_property (GObject * object, guint prop_id, |
| GValue * value, GParamSpec * pspec); |
| static void gst_rtp_g726_pay_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec); |
| |
| static gboolean gst_rtp_g726_pay_setcaps (GstRTPBasePayload * payload, |
| GstCaps * caps); |
| static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstRTPBasePayload * |
| payload, GstBuffer * buffer); |
| |
| #define gst_rtp_g726_pay_parent_class parent_class |
| G_DEFINE_TYPE (GstRtpG726Pay, gst_rtp_g726_pay, |
| GST_TYPE_RTP_BASE_AUDIO_PAYLOAD); |
| |
| static void |
| gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass) |
| { |
| GObjectClass *gobject_class; |
| GstElementClass *gstelement_class; |
| GstRTPBasePayloadClass *gstrtpbasepayload_class; |
| |
| gobject_class = (GObjectClass *) klass; |
| gstelement_class = (GstElementClass *) klass; |
| gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; |
| |
| gobject_class->set_property = gst_rtp_g726_pay_set_property; |
| gobject_class->get_property = gst_rtp_g726_pay_get_property; |
| |
| g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FORCE_AAL2, |
| g_param_spec_boolean ("force-aal2", "Force AAL2", |
| "Force AAL2 encoding for compatibility with bad depayloaders", |
| DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| gst_element_class_add_pad_template (gstelement_class, |
| gst_static_pad_template_get (&gst_rtp_g726_pay_sink_template)); |
| gst_element_class_add_pad_template (gstelement_class, |
| gst_static_pad_template_get (&gst_rtp_g726_pay_src_template)); |
| |
| gst_element_class_set_static_metadata (gstelement_class, |
| "RTP G.726 payloader", "Codec/Payloader/Network/RTP", |
| "Payload-encodes G.726 audio into a RTP packet", |
| "Axis Communications <dev-gstreamer@axis.com>"); |
| |
| gstrtpbasepayload_class->set_caps = gst_rtp_g726_pay_setcaps; |
| gstrtpbasepayload_class->handle_buffer = gst_rtp_g726_pay_handle_buffer; |
| |
| GST_DEBUG_CATEGORY_INIT (rtpg726pay_debug, "rtpg726pay", 0, |
| "G.726 RTP Payloader"); |
| } |
| |
| static void |
| gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay) |
| { |
| GstRTPBaseAudioPayload *rtpbaseaudiopayload; |
| |
| rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg726pay); |
| |
| GST_RTP_BASE_PAYLOAD (rtpg726pay)->clock_rate = 8000; |
| |
| rtpg726pay->force_aal2 = DEFAULT_FORCE_AAL2; |
| |
| /* sample based codec */ |
| gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); |
| } |
| |
| static gboolean |
| gst_rtp_g726_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) |
| { |
| gchar *encoding_name; |
| GstStructure *structure; |
| GstRTPBaseAudioPayload *rtpbaseaudiopayload; |
| GstRtpG726Pay *pay; |
| GstCaps *peercaps; |
| gboolean res; |
| |
| rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (payload); |
| pay = GST_RTP_G726_PAY (payload); |
| |
| structure = gst_caps_get_structure (caps, 0); |
| |
| if (!gst_structure_get_int (structure, "bitrate", &pay->bitrate)) |
| pay->bitrate = 32000; |
| |
| GST_DEBUG_OBJECT (payload, "using bitrate %d", pay->bitrate); |
| |
| pay->aal2 = FALSE; |
| |
| /* first see what we can do with the bitrate */ |
| switch (pay->bitrate) { |
| case 16000: |
| encoding_name = g_strdup ("G726-16"); |
| gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, |
| 2); |
| break; |
| case 24000: |
| encoding_name = g_strdup ("G726-24"); |
| gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, |
| 3); |
| break; |
| case 32000: |
| encoding_name = g_strdup ("G726-32"); |
| gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, |
| 4); |
| break; |
| case 40000: |
| encoding_name = g_strdup ("G726-40"); |
| gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, |
| 5); |
| break; |
| default: |
| goto invalid_bitrate; |
| } |
| |
| GST_DEBUG_OBJECT (payload, "selected base encoding %s", encoding_name); |
| |
| /* now see if we need to produce AAL2 or not */ |
| peercaps = gst_pad_peer_query_caps (payload->srcpad, NULL); |
| if (peercaps) { |
| GstCaps *filter, *intersect; |
| gchar *capsstr; |
| |
| GST_DEBUG_OBJECT (payload, "have peercaps %" GST_PTR_FORMAT, peercaps); |
| |
| capsstr = g_strdup_printf ("application/x-rtp, " |
| "media = (string) \"audio\", " |
| "clock-rate = (int) 8000, " |
| "encoding-name = (string) %s; " |
| "application/x-rtp, " |
| "media = (string) \"audio\", " |
| "clock-rate = (int) 8000, " |
| "encoding-name = (string) AAL2-%s", encoding_name, encoding_name); |
| filter = gst_caps_from_string (capsstr); |
| g_free (capsstr); |
| g_free (encoding_name); |
| |
| /* intersect to filter */ |
| intersect = gst_caps_intersect (peercaps, filter); |
| gst_caps_unref (peercaps); |
| gst_caps_unref (filter); |
| |
| GST_DEBUG_OBJECT (payload, "intersected to %" GST_PTR_FORMAT, intersect); |
| |
| if (!intersect) |
| goto no_format; |
| if (gst_caps_is_empty (intersect)) { |
| gst_caps_unref (intersect); |
| goto no_format; |
| } |
| |
| structure = gst_caps_get_structure (intersect, 0); |
| |
| /* now see what encoding name we settled on, we need to dup because the |
| * string goes away when we unref the intersection below. */ |
| encoding_name = |
| g_strdup (gst_structure_get_string (structure, "encoding-name")); |
| |
| /* if we managed to negotiate to AAL2, we definatly are going to do AAL2 |
| * encoding. Else we only encode AAL2 when explicitly set by the |
| * property. */ |
| if (g_str_has_prefix (encoding_name, "AAL2-")) |
| pay->aal2 = TRUE; |
| else |
| pay->aal2 = pay->force_aal2; |
| |
| GST_DEBUG_OBJECT (payload, "final encoding %s, AAL2 %d", encoding_name, |
| pay->aal2); |
| |
| gst_caps_unref (intersect); |
| } else { |
| /* downstream can do anything but we prefer the better supported non-AAL2 */ |
| pay->aal2 = pay->force_aal2; |
| GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2); |
| } |
| |
| gst_rtp_base_payload_set_options (payload, "audio", TRUE, encoding_name, |
| 8000); |
| res = gst_rtp_base_payload_set_outcaps (payload, NULL); |
| |
| g_free (encoding_name); |
| |
| return res; |
| |
| /* ERRORS */ |
| invalid_bitrate: |
| { |
| GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", pay->bitrate); |
| return FALSE; |
| } |
| no_format: |
| { |
| GST_ERROR_OBJECT (payload, "could not negotiate format"); |
| return FALSE; |
| } |
| } |
| |
| static GstFlowReturn |
| gst_rtp_g726_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer) |
| { |
| GstFlowReturn res; |
| GstRtpG726Pay *pay; |
| |
| pay = GST_RTP_G726_PAY (payload); |
| |
| if (!pay->aal2) { |
| GstMapInfo map; |
| guint8 *data, tmp; |
| gsize size; |
| |
| /* for non AAL2, we need to reshuffle the bytes, we can do this in-place |
| * when the buffer is writable. */ |
| buffer = gst_buffer_make_writable (buffer); |
| |
| gst_buffer_map (buffer, &map, GST_MAP_READWRITE); |
| data = map.data; |
| size = map.size; |
| |
| GST_LOG_OBJECT (pay, "packing %" G_GSIZE_FORMAT " bytes of data", map.size); |
| |
| /* we need to reshuffle the bytes, output is of the form: |
| * A B C D .. with the number of bits depending on the bitrate. */ |
| switch (pay->bitrate) { |
| case 16000: |
| { |
| /* 0 |
| * 0 1 2 3 4 5 6 7 |
| * +-+-+-+-+-+-+-+-+- |
| * |D D|C C|B B|A A| ... |
| * |0 1|0 1|0 1|0 1| |
| * +-+-+-+-+-+-+-+-+- |
| */ |
| while (size > 0) { |
| tmp = *data; |
| *data++ = ((tmp & 0xc0) >> 6) | |
| ((tmp & 0x30) >> 2) | ((tmp & 0x0c) << 2) | ((tmp & 0x03) << 6); |
| size--; |
| } |
| break; |
| } |
| case 24000: |
| { |
| /* 0 1 2 |
| * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- |
| * |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ... |
| * |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1| |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- |
| */ |
| while (size > 2) { |
| tmp = *data; |
| *data++ = ((tmp & 0xc0) >> 6) | |
| ((tmp & 0x38) >> 1) | ((tmp & 0x07) << 5); |
| tmp = *data; |
| *data++ = ((tmp & 0x80) >> 7) | |
| ((tmp & 0x70) >> 3) | ((tmp & 0x0e) << 4) | ((tmp & 0x01) << 7); |
| tmp = *data; |
| *data++ = ((tmp & 0xe0) >> 5) | |
| ((tmp & 0x1c) >> 2) | ((tmp & 0x03) << 6); |
| size -= 3; |
| } |
| break; |
| } |
| case 32000: |
| { |
| /* 0 1 |
| * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- |
| * |B B B B|A A A A|D D D D|C C C C| ... |
| * |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3| |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- |
| */ |
| while (size > 0) { |
| tmp = *data; |
| *data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4); |
| size--; |
| } |
| break; |
| } |
| case 40000: |
| { |
| /* 0 1 2 3 4 |
| * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- |
| * |B B B|A A A A A|D|C C C C C|B B|E E E E|D D D D|G G|F F F F F|E|H H H H H|G G G| |
| * |2 3 4|0 1 2 3 4|4|0 1 2 3 4|0 1|1 2 3 4|0 1 2 3|3 4|0 1 2 3 4|0|0 1 2 3 4|0 1 2| |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- |
| */ |
| while (size > 4) { |
| tmp = *data; |
| *data++ = ((tmp & 0xe0) >> 5) | ((tmp & 0x1f) << 3); |
| tmp = *data; |
| *data++ = ((tmp & 0x80) >> 7) | |
| ((tmp & 0x7c) >> 2) | ((tmp & 0x03) << 6); |
| tmp = *data; |
| *data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4); |
| tmp = *data; |
| *data++ = ((tmp & 0xc0) >> 6) | |
| ((tmp & 0x3e) << 2) | ((tmp & 0x01) << 7); |
| tmp = *data; |
| *data++ = ((tmp & 0xf8) >> 3) | ((tmp & 0x07) << 5); |
| size -= 5; |
| } |
| break; |
| } |
| } |
| gst_buffer_unmap (buffer, &map); |
| } |
| |
| res = |
| GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (payload, |
| buffer); |
| |
| return res; |
| } |
| |
| static void |
| gst_rtp_g726_pay_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec) |
| { |
| GstRtpG726Pay *rtpg726pay; |
| |
| rtpg726pay = GST_RTP_G726_PAY (object); |
| |
| switch (prop_id) { |
| case PROP_FORCE_AAL2: |
| rtpg726pay->force_aal2 = g_value_get_boolean (value); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_rtp_g726_pay_get_property (GObject * object, guint prop_id, |
| GValue * value, GParamSpec * pspec) |
| { |
| GstRtpG726Pay *rtpg726pay; |
| |
| rtpg726pay = GST_RTP_G726_PAY (object); |
| |
| switch (prop_id) { |
| case PROP_FORCE_AAL2: |
| g_value_set_boolean (value, rtpg726pay->force_aal2); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| gboolean |
| gst_rtp_g726_pay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpg726pay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_G726_PAY); |
| } |