From 799757ccf1d03c33c75bc597cd5ef77741dcb6a7 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Fri, 3 Jun 2011 09:17:04 +0000 Subject: Imported upstream 4.91 --- audio/gstrtpsbcpay.c | 352 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 352 insertions(+) create mode 100644 audio/gstrtpsbcpay.c (limited to 'audio/gstrtpsbcpay.c') diff --git a/audio/gstrtpsbcpay.c b/audio/gstrtpsbcpay.c new file mode 100644 index 0000000..1159bfe --- /dev/null +++ b/audio/gstrtpsbcpay.c @@ -0,0 +1,352 @@ +/* + * + * BlueZ - Bluetooth protocol stack for Linux + * + * Copyright (C) 2004-2010 Marcel Holtmann + * + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser 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 +#endif + +#include "gstpragma.h" +#include "gstrtpsbcpay.h" +#include +#include + +#define RTP_SBC_PAYLOAD_HEADER_SIZE 1 +#define DEFAULT_MIN_FRAMES 0 +#define RTP_SBC_HEADER_TOTAL (12 + RTP_SBC_PAYLOAD_HEADER_SIZE) + +#if __BYTE_ORDER == __LITTLE_ENDIAN + +struct rtp_payload { + guint8 frame_count:4; + guint8 rfa0:1; + guint8 is_last_fragment:1; + guint8 is_first_fragment:1; + guint8 is_fragmented:1; +} __attribute__ ((packed)); + +#elif __BYTE_ORDER == __BIG_ENDIAN + +struct rtp_payload { + guint8 is_fragmented:1; + guint8 is_first_fragment:1; + guint8 is_last_fragment:1; + guint8 rfa0:1; + guint8 frame_count:4; +} __attribute__ ((packed)); + +#else +#error "Unknown byte order" +#endif + +enum { + PROP_0, + PROP_MIN_FRAMES +}; + +GST_DEBUG_CATEGORY_STATIC(gst_rtp_sbc_pay_debug); +#define GST_CAT_DEFAULT gst_rtp_sbc_pay_debug + +GST_BOILERPLATE(GstRtpSBCPay, gst_rtp_sbc_pay, GstBaseRTPPayload, + GST_TYPE_BASE_RTP_PAYLOAD); + +static const GstElementDetails gst_rtp_sbc_pay_details = + GST_ELEMENT_DETAILS("RTP packet payloader", + "Codec/Payloader/Network", + "Payload SBC audio as RTP packets", + "Thiago Sousa Santos " + ""); + +static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory = + GST_STATIC_PAD_TEMPLATE("sink", GST_PAD_SINK, GST_PAD_ALWAYS, + GST_STATIC_CAPS("audio/x-sbc, " + "rate = (int) { 16000, 32000, 44100, 48000 }, " + "channels = (int) [ 1, 2 ], " + "mode = (string) { \"mono\", \"dual\", \"stereo\", \"joint\" }, " + "blocks = (int) { 4, 8, 12, 16 }, " + "subbands = (int) { 4, 8 }, " + "allocation = (string) { \"snr\", \"loudness\" }, " + "bitpool = (int) [ 2, 64 ]") + ); + +static GstStaticPadTemplate gst_rtp_sbc_pay_src_factory = + 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) { 16000, 32000, 44100, 48000 }," + "encoding-name = (string) \"SBC\"") + ); + +static void gst_rtp_sbc_pay_set_property(GObject *object, guint prop_id, + const GValue *value, GParamSpec *pspec); +static void gst_rtp_sbc_pay_get_property(GObject *object, guint prop_id, + GValue *value, GParamSpec *pspec); + +static gint gst_rtp_sbc_pay_get_frame_len(gint subbands, gint channels, + gint blocks, gint bitpool, const gchar *channel_mode) +{ + gint len; + gint join; + + len = 4 + (4 * subbands * channels)/8; + + if (strcmp(channel_mode, "mono") == 0 || + strcmp(channel_mode, "dual") == 0) + len += ((blocks * channels * bitpool) + 7) / 8; + else { + join = strcmp(channel_mode, "joint") == 0 ? 1 : 0; + len += ((join * subbands + blocks * bitpool) + 7) / 8; + } + + return len; +} + +static gboolean gst_rtp_sbc_pay_set_caps(GstBaseRTPPayload *payload, + GstCaps *caps) +{ + GstRtpSBCPay *sbcpay; + gint rate, subbands, channels, blocks, bitpool; + gint frame_len; + const gchar *channel_mode; + GstStructure *structure; + + sbcpay = GST_RTP_SBC_PAY(payload); + + structure = gst_caps_get_structure(caps, 0); + if (!gst_structure_get_int(structure, "rate", &rate)) + return FALSE; + if (!gst_structure_get_int(structure, "channels", &channels)) + return FALSE; + if (!gst_structure_get_int(structure, "blocks", &blocks)) + return FALSE; + if (!gst_structure_get_int(structure, "bitpool", &bitpool)) + return FALSE; + if (!gst_structure_get_int(structure, "subbands", &subbands)) + return FALSE; + + channel_mode = gst_structure_get_string(structure, "mode"); + if (!channel_mode) + return FALSE; + + frame_len = gst_rtp_sbc_pay_get_frame_len(subbands, channels, blocks, + bitpool, channel_mode); + + sbcpay->frame_length = frame_len; + + gst_basertppayload_set_options(payload, "audio", TRUE, "SBC", rate); + + GST_DEBUG_OBJECT(payload, "calculated frame length: %d ", frame_len); + + return gst_basertppayload_set_outcaps(payload, NULL); +} + +static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay) +{ + guint available; + guint max_payload; + GstBuffer *outbuf; + guint8 *payload_data; + guint frame_count; + guint payload_length; + struct rtp_payload *payload; + + if (sbcpay->frame_length == 0) { + GST_ERROR_OBJECT(sbcpay, "Frame length is 0"); + return GST_FLOW_ERROR; + } + + available = gst_adapter_available(sbcpay->adapter); + + max_payload = gst_rtp_buffer_calc_payload_len( + GST_BASE_RTP_PAYLOAD_MTU(sbcpay) - RTP_SBC_PAYLOAD_HEADER_SIZE, + 0, 0); + + max_payload = MIN(max_payload, available); + frame_count = max_payload / sbcpay->frame_length; + payload_length = frame_count * sbcpay->frame_length; + if (payload_length == 0) /* Nothing to send */ + return GST_FLOW_OK; + + outbuf = gst_rtp_buffer_new_allocate(payload_length + + RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0); + + gst_rtp_buffer_set_payload_type(outbuf, + GST_BASE_RTP_PAYLOAD_PT(sbcpay)); + + payload_data = gst_rtp_buffer_get_payload(outbuf); + payload = (struct rtp_payload *) payload_data; + memset(payload, 0, sizeof(struct rtp_payload)); + payload->frame_count = frame_count; + + gst_adapter_copy(sbcpay->adapter, payload_data + + RTP_SBC_PAYLOAD_HEADER_SIZE, 0, payload_length); + gst_adapter_flush(sbcpay->adapter, payload_length); + + GST_BUFFER_TIMESTAMP(outbuf) = sbcpay->timestamp; + GST_DEBUG_OBJECT(sbcpay, "Pushing %d bytes", payload_length); + + return gst_basertppayload_push(GST_BASE_RTP_PAYLOAD(sbcpay), outbuf); +} + +static GstFlowReturn gst_rtp_sbc_pay_handle_buffer(GstBaseRTPPayload *payload, + GstBuffer *buffer) +{ + GstRtpSBCPay *sbcpay; + guint available; + + /* FIXME check for negotiation */ + + sbcpay = GST_RTP_SBC_PAY(payload); + sbcpay->timestamp = GST_BUFFER_TIMESTAMP(buffer); + + gst_adapter_push(sbcpay->adapter, buffer); + + available = gst_adapter_available(sbcpay->adapter); + if (available + RTP_SBC_HEADER_TOTAL >= + GST_BASE_RTP_PAYLOAD_MTU(sbcpay) || + (available > + (sbcpay->min_frames * sbcpay->frame_length))) + return gst_rtp_sbc_pay_flush_buffers(sbcpay); + + return GST_FLOW_OK; +} + +static gboolean gst_rtp_sbc_pay_handle_event(GstPad *pad, + GstEvent *event) +{ + GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY(GST_PAD_PARENT(pad)); + + switch (GST_EVENT_TYPE(event)) { + case GST_EVENT_EOS: + gst_rtp_sbc_pay_flush_buffers(sbcpay); + break; + default: + break; + } + + return FALSE; +} + +static void gst_rtp_sbc_pay_base_init(gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS(g_class); + + gst_element_class_add_pad_template(element_class, + gst_static_pad_template_get(&gst_rtp_sbc_pay_sink_factory)); + gst_element_class_add_pad_template(element_class, + gst_static_pad_template_get(&gst_rtp_sbc_pay_src_factory)); + + gst_element_class_set_details(element_class, &gst_rtp_sbc_pay_details); +} + +static void gst_rtp_sbc_pay_finalize(GObject *object) +{ + GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY(object); + g_object_unref(sbcpay->adapter); + + GST_CALL_PARENT(G_OBJECT_CLASS, finalize, (object)); +} + +static void gst_rtp_sbc_pay_class_init(GstRtpSBCPayClass *klass) +{ + GObjectClass *gobject_class; + GstBaseRTPPayloadClass *payload_class = + GST_BASE_RTP_PAYLOAD_CLASS(klass); + + gobject_class = G_OBJECT_CLASS(klass); + parent_class = g_type_class_peek_parent(klass); + + gobject_class->finalize = GST_DEBUG_FUNCPTR(gst_rtp_sbc_pay_finalize); + gobject_class->set_property = GST_DEBUG_FUNCPTR( + gst_rtp_sbc_pay_set_property); + gobject_class->get_property = GST_DEBUG_FUNCPTR( + gst_rtp_sbc_pay_get_property); + + payload_class->set_caps = GST_DEBUG_FUNCPTR(gst_rtp_sbc_pay_set_caps); + payload_class->handle_buffer = GST_DEBUG_FUNCPTR( + gst_rtp_sbc_pay_handle_buffer); + payload_class->handle_event = GST_DEBUG_FUNCPTR( + gst_rtp_sbc_pay_handle_event); + + /* properties */ + g_object_class_install_property(G_OBJECT_CLASS(klass), + PROP_MIN_FRAMES, + g_param_spec_int("min-frames", "minimum frame number", + "Minimum quantity of frames to send in one packet " + "(-1 for maximum allowed by the mtu)", + -1, G_MAXINT, DEFAULT_MIN_FRAMES, G_PARAM_READWRITE)); + + GST_DEBUG_CATEGORY_INIT(gst_rtp_sbc_pay_debug, "rtpsbcpay", 0, + "RTP SBC payloader"); +} + +static void gst_rtp_sbc_pay_set_property(GObject *object, guint prop_id, + const GValue *value, GParamSpec *pspec) +{ + GstRtpSBCPay *sbcpay; + + sbcpay = GST_RTP_SBC_PAY(object); + + switch (prop_id) { + case PROP_MIN_FRAMES: + sbcpay->min_frames = g_value_get_int(value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec); + break; + } +} + +static void gst_rtp_sbc_pay_get_property(GObject *object, guint prop_id, + GValue *value, GParamSpec *pspec) +{ + GstRtpSBCPay *sbcpay; + + sbcpay = GST_RTP_SBC_PAY(object); + + switch (prop_id) { + case PROP_MIN_FRAMES: + g_value_set_int(value, sbcpay->min_frames); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec); + break; + } +} + +static void gst_rtp_sbc_pay_init(GstRtpSBCPay *self, GstRtpSBCPayClass *klass) +{ + self->adapter = gst_adapter_new(); + self->frame_length = 0; + self->timestamp = 0; + + self->min_frames = DEFAULT_MIN_FRAMES; +} + +gboolean gst_rtp_sbc_pay_plugin_init(GstPlugin *plugin) +{ + return gst_element_register(plugin, "rtpsbcpay", GST_RANK_NONE, + GST_TYPE_RTP_SBC_PAY); +} + -- cgit v1.2.3