From: Arun Raghavan <git@xxxxxxxxxxxxxxxx>
This adds a GStreamer-based RTP implementation to replace our own. The
original implementation is retained for cases where it is not possible
to include GStreamer as a dependency.
The idea with this is to be able to start supporting more advanced RTP
features such as RTP, non-PCM audio, and potentially synchronised
playback.
---
configure.ac | 17 ++
src/Makefile.am | 16 +-
src/modules/rtp/module-rtp-recv.c | 2 +-
src/modules/rtp/module-rtp-send.c | 2 +-
src/modules/rtp/rtp-common.c | 97 ++++++++
src/modules/rtp/rtp-gstreamer.c | 475 ++++++++++++++++++++++++++++++++++++++
src/modules/rtp/rtp-native.c | 379 ++++++++++++++++++++++++++++++
src/modules/rtp/rtp.c | 451 ------------------------------------
src/modules/rtp/rtp.h | 4 +-
9 files changed, 984 insertions(+), 459 deletions(-)
create mode 100644 src/modules/rtp/rtp-common.c
create mode 100644 src/modules/rtp/rtp-gstreamer.c
create mode 100644 src/modules/rtp/rtp-native.c
delete mode 100644 src/modules/rtp/rtp.c
diff --git a/configure.ac b/configure.ac
index 8454e4c..6e440eb 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1301,6 +1301,22 @@ AC_SUBST(HAVE_SYSTEMD_JOURNAL)
AM_CONDITIONAL([HAVE_SYSTEMD_JOURNAL], [test "x$HAVE_SYSTEMD_JOURNAL" = x1])
AS_IF([test "x$HAVE_SYSTEMD_JOURNAL" = "x1"], AC_DEFINE([HAVE_SYSTEMD_JOURNAL], 1, [Have SYSTEMDJOURNAL?]))
+#### GStreamer-based RTP support (optional) ####
+
+AC_ARG_ENABLE([gstreamer],
+ AS_HELP_STRING([--disable-gstreamer],[Disable optional GStreamer-based RTP support]))
+
+AS_IF([test "x$enable_gstreamer" != "xno"],
+ [PKG_CHECK_MODULES(GSTREAMER, [ gstreamer-1.0 gstreamer-app-1.0 gstreamer-rtp-1.0 gio-2.0 ],
+ HAVE_GSTREAMER=1, HAVE_GSTREAMER=0)],
+ HAVE_GSTREAMER=0)
+
+AS_IF([test "x$enable_gstreamer" = "xyes" && test "x$HAVE_GSTREAMER" = "x0"],
+ [AC_MSG_ERROR([*** GStreamer 1.0 support not found])])
+
+AM_CONDITIONAL([HAVE_GSTREAMER], [test "x$HAVE_GSTREAMER" = x1])
+AS_IF([test "x$HAVE_GSTREAMER" = "x1"], AC_DEFINE([HAVE_GSTREAMER], 1, [Have GStreamer?]))
+
#### Build and Install man pages ####
AC_ARG_ENABLE([manpages],
@@ -1647,6 +1663,7 @@ echo "
Enable speex (resampler, AEC): ${ENABLE_SPEEX}
Enable soxr (resampler): ${ENABLE_SOXR}
Enable WebRTC echo canceller: ${ENABLE_WEBRTC}
+ Enable GStreamer-based RTP: $}HAVE_GSTREAMER}
Enable gcov coverage: ${ENABLE_GCOV}
Enable unit tests: ${ENABLE_TESTS}
Database
diff --git a/src/Makefile.am b/src/Makefile.am
index aa96999..85ac0da 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -1124,13 +1124,21 @@ libprotocol_esound_la_LIBADD = $(AM_LIBADD) libpulsecore- at PA_MAJORMINOR@.la libp
endif
librtp_la_SOURCES = \
- modules/rtp/rtp.c modules/rtp/rtp.h \
+ modules/rtp/rtp-common.c modules/rtp/rtp.h \
modules/rtp/sdp.c modules/rtp/sdp.h \
modules/rtp/sap.c modules/rtp/sap.h \
modules/rtp/rtsp_client.c modules/rtp/rtsp_client.h \
modules/rtp/headerlist.c modules/rtp/headerlist.h
+librtp_la_CFLAGS = $(AM_CFLAGS)
librtp_la_LDFLAGS = $(AM_LDFLAGS) $(AM_LIBLDFLAGS) -avoid-version
librtp_la_LIBADD = $(AM_LIBADD) libpulsecore- at PA_MAJORMINOR@.la libpulsecommon- at PA_MAJORMINOR@.la libpulse.la
+if HAVE_GSTREAMER
+librtp_la_SOURCES += modules/rtp/rtp-gstreamer.c
+librtp_la_CFLAGS += $(GSTREAMER_CFLAGS)
+librtp_la_LIBADD += $(GSTREAMER_LIBS)
+else
+librtp_la_SOURCES += modules/rtp/rtp-native.c
+endif
libraop_la_SOURCES = \
modules/raop/raop_client.c modules/raop/raop_client.h \
@@ -2067,12 +2075,12 @@ endif
module_rtp_send_la_SOURCES = modules/rtp/module-rtp-send.c
module_rtp_send_la_LDFLAGS = $(MODULE_LDFLAGS)
module_rtp_send_la_LIBADD = $(MODULE_LIBADD) librtp.la
-module_rtp_send_la_CFLAGS = $(AM_CFLAGS)
+module_rtp_send_la_CFLAGS = $(AM_CFLAGS) $(GSTREAMER_CFLAGS)
module_rtp_recv_la_SOURCES = modules/rtp/module-rtp-recv.c
module_rtp_recv_la_LDFLAGS = $(MODULE_LDFLAGS)
module_rtp_recv_la_LIBADD = $(MODULE_LIBADD) librtp.la
-module_rtp_recv_la_CFLAGS = $(AM_CFLAGS)
+module_rtp_recv_la_CFLAGS = $(AM_CFLAGS) $(GSTREAMER_CFLAGS)
# JACK
@@ -2185,7 +2193,7 @@ module_bluez5_device_la_CFLAGS = $(AM_CFLAGS) $(SBC_CFLAGS)
module_raop_sink_la_SOURCES = modules/raop/module-raop-sink.c
module_raop_sink_la_LDFLAGS = $(MODULE_LDFLAGS)
module_raop_sink_la_LIBADD = $(MODULE_LIBADD) librtp.la libraop.la
-module_raop_sink_la_CFLAGS = $(AM_CFLAGS) -I$(top_srcdir)/src/modules/rtp
+module_raop_sink_la_CFLAGS = $(AM_CFLAGS) -I$(top_srcdir)/src/modules/rtp $(GSTREAMER_CFLAGS)
module_raop_discover_la_SOURCES = modules/raop/module-raop-discover.c
module_raop_discover_la_LDFLAGS = $(MODULE_LDFLAGS)
diff --git a/src/modules/rtp/module-rtp-recv.c b/src/modules/rtp/module-rtp-recv.c
index 1ee9c91..7a74aa4 100644
--- a/src/modules/rtp/module-rtp-recv.c
+++ b/src/modules/rtp/module-rtp-recv.c
@@ -570,7 +570,7 @@ static struct session *session_new(struct userdata *u, const pa_sdp_info *sdp_in
pa_memblock_unref(silence.memblock);
- if (!(s->rtp_context = pa_rtp_context_new_recv(fd, sdp_info->payload, pa_frame_size(&s->sdp_info.sample_spec))))
+ if (!(s->rtp_context = pa_rtp_context_new_recv(fd, sdp_info->payload, &s->sdp_info.sample_spec)))
goto fail;
pa_hashmap_put(s->userdata->by_origin, s->sdp_info.origin, s);
diff --git a/src/modules/rtp/module-rtp-send.c b/src/modules/rtp/module-rtp-send.c
index 6110455..6797e5a 100644
--- a/src/modules/rtp/module-rtp-send.c
+++ b/src/modules/rtp/module-rtp-send.c
@@ -486,7 +486,7 @@ int pa__init(pa_module*m) {
pa_xfree(n);
- if (!(u->rtp_context = pa_rtp_context_new_send(fd, payload, mtu, pa_frame_size(&ss))))
+ if (!(u->rtp_context = pa_rtp_context_new_send(fd, payload, mtu, &ss)))
goto fail;
pa_sap_context_init_send(&u->sap_context, sap_fd, p);
diff --git a/src/modules/rtp/rtp-common.c b/src/modules/rtp/rtp-common.c
new file mode 100644
index 0000000..65e2c7a
--- /dev/null
+++ b/src/modules/rtp/rtp-common.c
@@ -0,0 +1,97 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2006 Lennart Poettering
+
+ PulseAudio 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.
+
+ PulseAudio 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
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "rtp.h"
+
+#include <pulsecore/core-util.h>
+
+uint8_t pa_rtp_payload_from_sample_spec(const pa_sample_spec *ss) {
+ pa_assert(ss);
+
+ if (ss->format == PA_SAMPLE_S16BE && ss->rate == 44100 && ss->channels == 2)
+ return 10;
+ if (ss->format == PA_SAMPLE_S16BE && ss->rate == 44100 && ss->channels == 1)
+ return 11;
+
+ return 127;
+}
+
+pa_sample_spec *pa_rtp_sample_spec_from_payload(uint8_t payload, pa_sample_spec *ss) {
+ pa_assert(ss);
+
+ switch (payload) {
+ case 10:
+ ss->channels = 2;
+ ss->format = PA_SAMPLE_S16BE;
+ ss->rate = 44100;
+ break;
+
+ case 11:
+ ss->channels = 1;
+ ss->format = PA_SAMPLE_S16BE;
+ ss->rate = 44100;
+ break;
+
+ default:
+ return NULL;
+ }
+
+ return ss;
+}
+
+pa_sample_spec *pa_rtp_sample_spec_fixup(pa_sample_spec * ss) {
+ pa_assert(ss);
+
+ if (!pa_rtp_sample_spec_valid(ss))
+ ss->format = PA_SAMPLE_S16BE;
+
+ pa_assert(pa_rtp_sample_spec_valid(ss));
+ return ss;
+}
+
+int pa_rtp_sample_spec_valid(const pa_sample_spec *ss) {
+ pa_assert(ss);
+
+ if (!pa_sample_spec_valid(ss))
+ return 0;
+
+ return ss->format == PA_SAMPLE_S16BE;
+}
+
+const char* pa_rtp_format_to_string(pa_sample_format_t f) {
+ switch (f) {
+ case PA_SAMPLE_S16BE:
+ return "L16";
+ default:
+ return NULL;
+ }
+}
+
+pa_sample_format_t pa_rtp_string_to_format(const char *s) {
+ pa_assert(s);
+
+ if (pa_streq(s, "L16"))
+ return PA_SAMPLE_S16BE;
+ else
+ return PA_SAMPLE_INVALID;
+}
diff --git a/src/modules/rtp/rtp-gstreamer.c b/src/modules/rtp/rtp-gstreamer.c
new file mode 100644
index 0000000..413d0e4
--- /dev/null
+++ b/src/modules/rtp/rtp-gstreamer.c
@@ -0,0 +1,475 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2016 Arun Raghavan <mail at arunraghavan.net>
+
+ PulseAudio 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.
+
+ PulseAudio 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
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/timeval.h>
+#include <pulsecore/fdsem.h>
+#include <pulsecore/core-rtclock.h>
+
+#include "rtp.h"
+
+#include <gio/gio.h>
+
+#include <gst/gst.h>
+#include <gst/app/gstappsrc.h>
+#include <gst/app/gstappsink.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#define MAKE_ELEMENT_NAMED(v, e, n) \
+ v = gst_element_factory_make(e, n); \
+ if (!v) { \
+ pa_log("Could not create %s element", e); \
+ goto fail; \
+ }
+
+#define MAKE_ELEMENT(v, e) MAKE_ELEMENT_NAMED((v), (e), NULL)
+
+typedef struct pa_rtp_context {
+ pa_fdsem *fdsem;
+ pa_sample_spec ss;
+
+ GstElement *pipeline;
+ GstElement *appsrc;
+ GstElement *appsink;
+
+ uint32_t last_timestamp;
+} pa_rtp_context;
+
+static GstCaps* caps_from_sample_spec(const pa_sample_spec *ss) {
+ if (ss->format != PA_SAMPLE_S16BE)
+ return NULL;
+
+ return gst_caps_new_simple("audio/x-raw",
+ "format", G_TYPE_STRING, "S16BE",
+ "rate", G_TYPE_INT, (int) ss->rate,
+ "channels", G_TYPE_INT, (int) ss->channels,
+ "layout", G_TYPE_STRING, "interleaved",
+ NULL);
+}
+static bool init_send_pipeline(pa_rtp_context *c, int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss) {
+ GstElement *appsrc = NULL, *pay = NULL, *capsf = NULL, *rtpbin = NULL, *sink = NULL;
+ GstCaps *caps;
+
+ MAKE_ELEMENT(appsrc, "appsrc");
+ MAKE_ELEMENT(pay, "rtpL16pay");
+ MAKE_ELEMENT(capsf, "capsfilter");
+ MAKE_ELEMENT(rtpbin, "rtpbin");
+ MAKE_ELEMENT(sink, "fdsink");
+
+ c->pipeline = gst_pipeline_new(NULL);
+
+ gst_bin_add_many(GST_BIN(c->pipeline), appsrc, pay, capsf, rtpbin, sink, NULL);
+
+ caps = caps_from_sample_spec(ss);
+ if (!caps) {
+ pa_log("Unsupported format to payload");
+ goto fail;
+ }
+
+ g_object_set(appsrc, "caps", caps, "is-live", TRUE, "blocksize", mtu, "format", 3 /* time */, NULL);
+ g_object_set(pay, "mtu", mtu, NULL);
+ g_object_set(sink, "fd", fd, "enable-last-sample", FALSE, NULL);
+
+ gst_caps_unref(caps);
+
+ /* Force the payload type that we want */
+ caps = gst_caps_new_simple("application/x-rtp", "payload", G_TYPE_INT, (int) payload, NULL);
+ g_object_set(capsf, "caps", caps, NULL);
+ gst_caps_unref(caps);
+
+ if (!gst_element_link(appsrc, pay) ||
+ !gst_element_link(pay, capsf) ||
+ !gst_element_link_pads(capsf, "src", rtpbin, "send_rtp_sink_0") ||
+ !gst_element_link_pads(rtpbin, "send_rtp_src_0", sink, "sink")) {
+
+ pa_log("Could not set up send pipeline");
+ goto fail;
+ }
+
+ if (gst_element_set_state(c->pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
+ pa_log("Could not start pipeline");
+ goto fail;
+ }
+
+ c->appsrc = gst_object_ref(appsrc);
+
+ return true;
+
+fail:
+ if (c->pipeline) {
+ gst_object_unref(c->pipeline);
+ } else {
+ /* These weren't yet added to pipeline, so we still have a ref */
+ if (appsrc)
+ gst_object_unref(appsrc);
+ if (pay)
+ gst_object_unref(pay);
+ if (capsf)
+ gst_object_unref(capsf);
+ if (rtpbin)
+ gst_object_unref(rtpbin);
+ if (sink)
+ gst_object_unref(sink);
+ }
+
+ return false;
+}
+
+pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss) {
+ pa_rtp_context *c = NULL;
+ GError *error = NULL;
+
+ pa_assert(fd >= 0);
+
+ c = pa_xnew0(pa_rtp_context, 1);
+
+ c->fdsem = pa_fdsem_new();
+ c->ss = *ss;
+
+ if (!gst_init_check(NULL, NULL, &error)) {
+ pa_log_error("Could not initialise GStreamer: %s", error->message);
+ g_error_free(error);
+ goto fail;
+ }
+
+ if (!init_send_pipeline(c, fd, payload, mtu, ss))
+ goto fail;
+
+ return c;
+
+fail:
+ pa_xfree(c);
+ return NULL;
+}
+
+static bool process_bus_messages(pa_rtp_context *c) {
+ GstBus *bus;
+ GstMessage *message;
+ bool ret = true;
+
+ bus = gst_pipeline_get_bus(GST_PIPELINE(c->pipeline));
+
+ while (ret && (message = gst_bus_pop(bus))) {
+ if (GST_MESSAGE_TYPE(message) == GST_MESSAGE_ERROR) {
+ GError *error = NULL;
+
+ ret = false;
+
+ gst_message_parse_error(message, &error, NULL);
+ pa_log("Got an error: %s", error->message);
+
+ g_error_free(error);
+
+ pa_fdsem_post(c->fdsem);
+ }
+
+ gst_message_unref(message);
+ }
+
+ gst_object_unref(bus);
+
+ return ret;
+}
+
+static void free_buffer(pa_memblock *memblock) {
+ pa_memblock_release(memblock);
+ pa_memblock_unref(memblock);
+}
+
+int pa_rtp_send(pa_rtp_context *c, pa_memblockq *q) {
+ pa_memchunk chunk = { 0, };
+ GstBuffer *buf;
+ void *data;
+ bool stop = false;
+ int ret = 0;
+
+ pa_assert(c);
+ pa_assert(q);
+
+ if (!process_bus_messages(c))
+ return -1;
+
+ while (!stop && pa_memblockq_peek(q, &chunk) == 0) {
+ pa_assert(chunk.memblock);
+
+ data = pa_memblock_acquire(chunk.memblock);
+
+ buf = gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_READONLY | GST_MEMORY_FLAG_PHYSICALLY_CONTIGUOUS,
+ data, chunk.length, chunk.index, chunk.length, chunk.memblock,
+ (GDestroyNotify) free_buffer);
+
+ if (gst_app_src_push_buffer(GST_APP_SRC(c->appsrc), buf) != GST_FLOW_OK) {
+ pa_log_error("Could not push buffer");
+ stop = true;
+ ret = -1;
+ }
+
+ pa_memblockq_drop(q, chunk.length);
+ }
+
+ return ret;
+}
+
+static GstCaps* rtp_caps_from_sample_spec(const pa_sample_spec *ss) {
+ if (ss->format != PA_SAMPLE_S16BE)
+ return NULL;
+
+ return gst_caps_new_simple("application/x-rtp",
+ "media", G_TYPE_STRING, "audio",
+ "encoding-name", G_TYPE_STRING, "L16",
+ "clock-rate", G_TYPE_INT, (int) ss->rate,
+ "payload", G_TYPE_INT, (int) pa_rtp_payload_from_sample_spec(ss),
+ "layout", G_TYPE_STRING, "interleaved",
+ NULL);
+}
+
+static void on_pad_added(GstElement *element, GstPad *pad, gpointer userdata) {
+ pa_rtp_context *c = (pa_rtp_context *) userdata;
+ GstElement *depay;
+ GstPad *sinkpad;
+ GstPadLinkReturn ret;
+
+ depay = gst_bin_get_by_name(GST_BIN(c->pipeline), "depay");
+ pa_assert(depay);
+
+ sinkpad = gst_element_get_static_pad(depay, "sink");
+
+ ret = gst_pad_link(pad, sinkpad);
+ if (ret != GST_PAD_LINK_OK) {
+ GstBus *bus;
+ GError *error;
+
+ bus = gst_pipeline_get_bus(GST_PIPELINE(c->pipeline));
+ error = g_error_new(GST_CORE_ERROR, GST_CORE_ERROR_PAD, "Could not link rtpbin to depayloader");
+ gst_bus_post(bus, gst_message_new_error(GST_OBJECT(c->pipeline), error, NULL));
+
+ g_error_free(error);
+ gst_object_unref(bus);
+ }
+
+ gst_object_unref(sinkpad);
+ gst_object_unref(depay);
+}
+
+static bool init_receive_pipeline(pa_rtp_context *c, int fd, const pa_sample_spec *ss) {
+ GstElement *udpsrc = NULL, *rtpbin = NULL, *depay = NULL, *appsink = NULL;
+ GstCaps *caps;
+ GSocket *socket;
+ GError *error = NULL;
+
+ MAKE_ELEMENT(udpsrc, "udpsrc");
+ MAKE_ELEMENT(rtpbin, "rtpbin");
+ MAKE_ELEMENT_NAMED(depay, "rtpL16depay", "depay");
+ MAKE_ELEMENT(appsink, "appsink");
+
+ c->pipeline = gst_pipeline_new(NULL);
+
+ gst_bin_add_many(GST_BIN(c->pipeline), udpsrc, rtpbin, depay, appsink, NULL);
+
+ socket = g_socket_new_from_fd(fd, &error);
+ if (error) {
+ pa_log("Could not create socket: %s", error->message);
+ g_error_free(error);
+ goto fail;
+ }
+
+ caps = rtp_caps_from_sample_spec(ss);
+ if (!caps) {
+ pa_log("Unsupported format to payload");
+ goto fail;
+ }
+
+ g_object_set(udpsrc, "socket", socket, "caps", caps, "auto-multicast" /* caller handles this */, FALSE, NULL);
+ g_object_set(rtpbin, "latency", 0, "buffer-mode", 0 /* none */, NULL);
+ g_object_set(appsink, "sync", FALSE, "enable-last-sample", FALSE, NULL);
+
+ gst_caps_unref(caps);
+ g_object_unref(socket);
+
+ if (!gst_element_link_pads(udpsrc, "src", rtpbin, "recv_rtp_sink_0") ||
+ !gst_element_link(depay, appsink)) {
+
+ pa_log("Could not set up send pipeline");
+ goto fail;
+ }
+
+ g_signal_connect(G_OBJECT(rtpbin), "pad-added", G_CALLBACK(on_pad_added), c);
+
+ if (gst_element_set_state(c->pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
+ pa_log("Could not start pipeline");
+ goto fail;
+ }
+
+ c->appsink = gst_object_ref(appsink);
+
+ return true;
+
+fail:
+ if (c->pipeline) {
+ gst_object_unref(c->pipeline);
+ } else {
+ /* These weren't yet added to pipeline, so we still have a ref */
+ if (udpsrc)
+ gst_object_unref(udpsrc);
+ if (depay)
+ gst_object_unref(depay);
+ if (rtpbin)
+ gst_object_unref(rtpbin);
+ if (appsink)
+ gst_object_unref(appsink);
+ }
+
+ return false;
+}
+
+static void appsink_eos(GstAppSink *appsink, gpointer userdata) {
+ pa_rtp_context *c = (pa_rtp_context *) userdata;
+
+ pa_fdsem_post(c->fdsem);
+}
+
+static GstFlowReturn appsink_new_sample(GstAppSink *appsink, gpointer userdata) {
+ pa_rtp_context *c = (pa_rtp_context *) userdata;
+
+ pa_fdsem_post(c->fdsem);
+
+ return GST_FLOW_OK;
+}
+
+pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample_spec *ss) {
+ pa_rtp_context *c = NULL;
+ GstAppSinkCallbacks callbacks = { 0, };
+ GError *error = NULL;
+
+ pa_assert(fd >= 0);
+
+ c = pa_xnew0(pa_rtp_context, 1);
+
+ c->fdsem = pa_fdsem_new();
+ c->ss = *ss;
+
+ if (!gst_init_check(NULL, NULL, &error)) {
+ pa_log_error("Could not initialise GStreamer: %s", error->message);
+ g_error_free(error);
+ goto fail;
+ }
+
+ if (!init_receive_pipeline(c, fd, ss))
+ goto fail;
+
+ callbacks.eos = appsink_eos;
+ callbacks.new_sample = appsink_new_sample;
+ gst_app_sink_set_callbacks(GST_APP_SINK(c->appsink), &callbacks, c, NULL);
+
+ return c;
+
+fail:
+ pa_xfree(c);
+ return NULL;
+}
+
+int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_t *rtp_tstamp, struct timeval *tstamp) {
+ GstSample *sample = NULL;
+ GstBuffer *buf;
+ GstMapInfo info;
+ void *data;
+
+ if (!process_bus_messages(c))
+ goto fail;
+
+ sample = gst_app_sink_pull_sample(GST_APP_SINK(c->appsink));
+ if (!sample) {
+ pa_log_warn("Could not get any more data");
+ goto fail;
+ }
+
+ buf = gst_sample_get_buffer(sample);
+
+ if (GST_BUFFER_IS_DISCONT(buf))
+ pa_log_info("Discontinuity detected, possibly lost some packets");
+
+ if (!gst_buffer_map(buf, &info, GST_MAP_READ))
+ goto fail;
+
+ pa_assert(pa_mempool_block_size_max(pool) >= info.size);
+
+ chunk->memblock = pa_memblock_new(pool, info.size);
+ chunk->index = 0;
+ chunk->length = info.size;
+
+ data = pa_memblock_acquire_chunk(chunk);
+ /* TODO: we could probably just provide an allocator and avoid a memcpy */
+ memcpy(data, info.data, info.size);
+ pa_memblock_release(chunk->memblock);
+
+ /* When buffer-mode = none, the buffer PTS is the RTP timestamp, converted
+ * to time units (instead of clock-rate units as is in the header) and
+ * wraparound-corrected, and the DTS is the pipeline clock timestamp from
+ * when the buffer was acquired at the source (this is actually the running
+ * time which is why we need to add base time). */
+ *rtp_tstamp = gst_util_uint64_scale_int(GST_BUFFER_PTS(buf), c->ss.rate, GST_SECOND) & 0xFFFFFFFFU;
+ pa_timeval_rtstore(tstamp, (GST_BUFFER_DTS(buf) + gst_element_get_base_time(c->pipeline)) / GST_USECOND, false);
+
+ gst_buffer_unmap(buf, &info);
+ gst_sample_unref(sample);
+
+ return 0;
+
+fail:
+ if (sample)
+ gst_sample_unref(sample);
+
+ if (chunk->memblock)
+ pa_memblock_unref(chunk->memblock);
+
+ return -1;
+}
+
+void pa_rtp_context_destroy(pa_rtp_context *c) {
+ pa_assert(c);
+
+ if (c->appsrc) {
+ gst_app_src_end_of_stream(GST_APP_SRC(c->appsrc));
+ gst_object_unref(c->appsrc);
+ }
+
+ if (c->appsink)
+ gst_object_unref(c->appsink);
+
+ if (c->pipeline) {
+ gst_element_set_state(c->pipeline, GST_STATE_NULL);
+ gst_object_unref(c->pipeline);
+ }
+
+ if (c->fdsem)
+ pa_fdsem_free(c->fdsem);
+
+ pa_xfree(c);
+}
+
+pa_rtpoll_item* pa_rtp_context_get_rtpoll_item(pa_rtp_context *c, pa_rtpoll *rtpoll) {
+ return pa_rtpoll_item_new_fdsem(rtpoll, PA_RTPOLL_LATE, c->fdsem);
+}
+
+size_t pa_rtp_context_get_frame_size(pa_rtp_context *c) {
+ return pa_frame_size(&c->ss);
+}
diff --git a/src/modules/rtp/rtp-native.c b/src/modules/rtp/rtp-native.c
new file mode 100644
index 0000000..5c1a8f2
--- /dev/null
+++ b/src/modules/rtp/rtp-native.c
@@ -0,0 +1,379 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2006 Lennart Poettering
+
+ PulseAudio 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.
+
+ PulseAudio 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
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+#include <unistd.h>
+#include <sys/ioctl.h>
+
+#ifdef HAVE_SYS_FILIO_H
+#include <sys/filio.h>
+#endif
+
+#ifdef HAVE_SYS_UIO_H
+#include <sys/uio.h>
+#endif
+
+#include <pulsecore/core-error.h>
+#include <pulsecore/log.h>
+#include <pulsecore/macro.h>
+#include <pulsecore/core-util.h>
+#include <pulsecore/arpa-inet.h>
+#include <pulsecore/poll.h>
+
+#include "rtp.h"
+
+typedef struct pa_rtp_context {
+ int fd;
+ uint16_t sequence;
+ uint32_t timestamp;
+ uint32_t ssrc;
+ uint8_t payload;
+ size_t frame_size;
+ size_t mtu;
+
+ pa_memchunk memchunk;
+} pa_rtp_context;
+
+pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss) {
+ pa_rtp_context *c;
+
+ pa_assert(fd >= 0);
+
+ c = pa_xnew0(pa_rtp_context, 1);
+
+ c->fd = fd;
+ c->sequence = (uint16_t) (rand()*rand());
+ c->timestamp = 0;
+ c->ssrc = (uint32_t) (rand()*rand());
+ c->payload = (uint8_t) (payload & 127U);
+ c->frame_size = pa_frame_size(ss);
+ c->mtu = mtu;
+
+ pa_memchunk_reset(&c->memchunk);
+
+ return c;
+}
+
+#define MAX_IOVECS 16
+
+int pa_rtp_send(pa_rtp_context *c, pa_memblockq *q) {
+ struct iovec iov[MAX_IOVECS];
+ pa_memblock* mb[MAX_IOVECS];
+ int iov_idx = 1;
+ size_t n = 0;
+
+ pa_assert(c);
+ pa_assert(q);
+
+ if (pa_memblockq_get_length(q) < c->mtu)
+ return 0;
+
+ for (;;) {
+ int r;
+ pa_memchunk chunk;
+
+ pa_memchunk_reset(&chunk);
+
+ if ((r = pa_memblockq_peek(q, &chunk)) >= 0) {
+
+ size_t k = n + chunk.length > c->mtu ? c->mtu - n : chunk.length;
+
+ pa_assert(chunk.memblock);
+
+ iov[iov_idx].iov_base = pa_memblock_acquire_chunk(&chunk);
+ iov[iov_idx].iov_len = k;
+ mb[iov_idx] = chunk.memblock;
+ iov_idx ++;
+
+ n += k;
+ pa_memblockq_drop(q, k);
+ }
+
+ pa_assert(n % c->frame_size == 0);
+
+ if (r < 0 || n >= c->mtu || iov_idx >= MAX_IOVECS) {
+ uint32_t header[3];
+ struct msghdr m;
+ ssize_t k;
+ int i;
+
+ if (n > 0) {
+ header[0] = htonl(((uint32_t) 2 << 30) | ((uint32_t) c->payload << 16) | ((uint32_t) c->sequence));
+ header[1] = htonl(c->timestamp);
+ header[2] = htonl(c->ssrc);
+
+ iov[0].iov_base = (void*)header;
+ iov[0].iov_len = sizeof(header);
+
+ m.msg_name = NULL;
+ m.msg_namelen = 0;
+ m.msg_iov = iov;
+ m.msg_iovlen = (size_t) iov_idx;
+ m.msg_control = NULL;
+ m.msg_controllen = 0;
+ m.msg_flags = 0;
+
+ k = sendmsg(c->fd, &m, MSG_DONTWAIT);
+
+ for (i = 1; i < iov_idx; i++) {
+ pa_memblock_release(mb[i]);
+ pa_memblock_unref(mb[i]);
+ }
+
+ c->sequence++;
+ } else
+ k = 0;
+
+ c->timestamp += (unsigned) (n/c->frame_size);
+
+ if (k < 0) {
+ if (errno != EAGAIN && errno != EINTR) /* If the queue is full, just ignore it */
+ pa_log("sendmsg() failed: %s", pa_cstrerror(errno));
+ return -1;
+ }
+
+ if (r < 0 || pa_memblockq_get_length(q) < c->mtu)
+ break;
+
+ n = 0;
+ iov_idx = 1;
+ }
+ }
+
+ return 0;
+}
+
+pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample_spec *ss) {
+ pa_rtp_context *c;
+
+ c = pa_xnew0(pa_rtp_context, 1);
+
+ c->fd = fd;
+ c->payload = payload;
+ c->frame_size = pa_frame_size(ss);
+
+ pa_memchunk_reset(&c->memchunk);
+
+ return c;
+}
+
+int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_t *rtp_tstamp, struct timeval *tstamp) {
+ int size;
+ struct msghdr m;
+ struct cmsghdr *cm;
+ struct iovec iov;
+ uint32_t header;
+ uint32_t ssrc;
+ uint8_t payload;
+ unsigned cc;
+ ssize_t r;
+ uint8_t aux[1024];
+ bool found_tstamp = false;
+
+ pa_assert(c);
+ pa_assert(chunk);
+
+ pa_memchunk_reset(chunk);
+
+ if (ioctl(c->fd, FIONREAD, &size) < 0) {
+ pa_log_warn("FIONREAD failed: %s", pa_cstrerror(errno));
+ goto fail;
+ }
+
+ if (size <= 0) {
+ /* size can be 0 due to any of the following reasons:
+ *
+ * 1. Somebody sent us a perfectly valid zero-length UDP packet.
+ * 2. Somebody sent us a UDP packet with a bad CRC.
+ *
+ * It is unknown whether size can actually be less than zero.
+ *
+ * In the first case, the packet has to be read out, otherwise the
+ * kernel will tell us again and again about it, thus preventing
+ * reception of any further packets. So let's just read it out
+ * now and discard it later, when comparing the number of bytes
+ * received (0) with the number of bytes wanted (1, see below).
+ *
+ * In the second case, recvmsg() will fail, thus allowing us to
+ * return the error.
+ *
+ * Just to avoid passing zero-sized memchunks and NULL pointers to
+ * recvmsg(), let's force allocation of at least one byte by setting
+ * size to 1.
+ */
+ size = 1;
+ }
+
+ if (c->memchunk.length < (unsigned) size) {
+ size_t l;
+
+ if (c->memchunk.memblock)
+ pa_memblock_unref(c->memchunk.memblock);
+
+ l = PA_MAX((size_t) size, pa_mempool_block_size_max(pool));
+
+ c->memchunk.memblock = pa_memblock_new(pool, l);
+ c->memchunk.index = 0;
+ c->memchunk.length = pa_memblock_get_length(c->memchunk.memblock);
+ }
+
+ pa_assert(c->memchunk.length >= (size_t) size);
+
+ chunk->memblock = pa_memblock_ref(c->memchunk.memblock);
+ chunk->index = c->memchunk.index;
+
+ iov.iov_base = pa_memblock_acquire_chunk(chunk);
+ iov.iov_len = (size_t) size;
+
+ m.msg_name = NULL;
+ m.msg_namelen = 0;
+ m.msg_iov = &iov;
+ m.msg_iovlen = 1;
+ m.msg_control = aux;
+ m.msg_controllen = sizeof(aux);
+ m.msg_flags = 0;
+
+ r = recvmsg(c->fd, &m, 0);
+
+ if (r != size) {
+ if (r < 0 && errno != EAGAIN && errno != EINTR)
+ pa_log_warn("recvmsg() failed: %s", r < 0 ? pa_cstrerror(errno) : "size mismatch");
+
+ goto fail;
+ }
+
+ if (size < 12) {
+ pa_log_warn("RTP packet too short.");
+ goto fail;
+ }
+
+ memcpy(&header, iov.iov_base, sizeof(uint32_t));
+ memcpy(rtp_tstamp, (uint8_t*) iov.iov_base + 4, sizeof(uint32_t));
+ memcpy(&ssrc, (uint8_t*) iov.iov_base + 8, sizeof(uint32_t));
+
+ pa_memblock_release(chunk->memblock);
+
+ header = ntohl(header);
+ *rtp_tstamp = ntohl(*rtp_tstamp);
+ ssrc = ntohl(c->ssrc);
+
+ if ((header >> 30) != 2) {
+ pa_log_warn("Unsupported RTP version.");
+ goto fail;
+ }
+
+ if ((header >> 29) & 1) {
+ pa_log_warn("RTP padding not supported.");
+ goto fail;
+ }
+
+ if ((header >> 28) & 1) {
+ pa_log_warn("RTP header extensions not supported.");
+ goto fail;
+ }
+
+ if (ssrc != c->ssrc) {
+ pa_log_debug("Got unexpected SSRC");
+ goto fail;
+ }
+
+ cc = (header >> 24) & 0xF;
+ payload = (uint8_t) ((header >> 16) & 127U);
+ c->sequence = (uint16_t) (header & 0xFFFFU);
+
+ if (payload != c->payload) {
+ pa_log_debug("Got unexpected payload: %u", payload);
+ goto fail;
+ }
+
+ if (12 + cc*4 > (unsigned) size) {
+ pa_log_warn("RTP packet too short. (CSRC)");
+ goto fail;
+ }
+
+ chunk->index += 12 + cc*4;
+ chunk->length = (size_t) size - 12 + cc*4;
+
+ if (chunk->length % c->frame_size != 0) {
+ pa_log_warn("Bad RTP packet size.");
+ goto fail;
+ }
+
+ c->memchunk.index = chunk->index + chunk->length;
+ c->memchunk.length = pa_memblock_get_length(c->memchunk.memblock) - c->memchunk.index;
+
+ if (c->memchunk.length <= 0) {
+ pa_memblock_unref(c->memchunk.memblock);
+ pa_memchunk_reset(&c->memchunk);
+ }
+
+ for (cm = CMSG_FIRSTHDR(&m); cm; cm = CMSG_NXTHDR(&m, cm))
+ if (cm->cmsg_level == SOL_SOCKET && cm->cmsg_type == SCM_TIMESTAMP) {
+ memcpy(tstamp, CMSG_DATA(cm), sizeof(struct timeval));
+ found_tstamp = true;
+ break;
+ }
+
+ if (!found_tstamp) {
+ pa_log_warn("Couldn't find SCM_TIMESTAMP data in auxiliary recvmsg() data!");
+ pa_zero(*tstamp);
+ }
+
+ return 0;
+
+fail:
+ if (chunk->memblock)
+ pa_memblock_unref(chunk->memblock);
+
+ return -1;
+}
+void pa_rtp_context_destroy(pa_rtp_context *c) {
+ pa_assert(c);
+
+ pa_assert_se(pa_close(c->fd) == 0);
+
+ if (c->memchunk.memblock)
+ pa_memblock_unref(c->memchunk.memblock);
+
+ pa_xfree(c);
+}
+
+size_t pa_rtp_context_get_frame_size(pa_rtp_context *c) {
+ return c->frame_size;
+}
+
+pa_rtpoll_item* pa_rtp_context_get_rtpoll_item(pa_rtp_context *c, pa_rtpoll *rtpoll) {
+ pa_rtpoll_item *item;
+ struct pollfd *p;
+
+ item = pa_rtpoll_item_new(rtpoll, PA_RTPOLL_LATE, 1);
+
+ p = pa_rtpoll_item_get_pollfd(item, NULL);
+ p->fd = c->fd;
+ p->events = POLLIN;
+ p->revents = 0;
+
+ return item;
+}
diff --git a/src/modules/rtp/rtp.c b/src/modules/rtp/rtp.c
deleted file mode 100644
index 5a24a03..0000000
--- a/src/modules/rtp/rtp.c
+++ /dev/null
@@ -1,451 +0,0 @@
-/***
- This file is part of PulseAudio.
-
- Copyright 2006 Lennart Poettering
-
- PulseAudio 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.
-
- PulseAudio 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
- General Public License for more details.
-
- You should have received a copy of the GNU Lesser General Public License
- along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
-***/
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <stdlib.h>
-#include <string.h>
-#include <errno.h>
-#include <unistd.h>
-#include <sys/ioctl.h>
-
-#ifdef HAVE_SYS_FILIO_H
-#include <sys/filio.h>
-#endif
-
-#ifdef HAVE_SYS_UIO_H
-#include <sys/uio.h>
-#endif
-
-#include <pulsecore/core-error.h>
-#include <pulsecore/log.h>
-#include <pulsecore/macro.h>
-#include <pulsecore/core-util.h>
-#include <pulsecore/arpa-inet.h>
-#include <pulsecore/poll.h>
-
-#include "rtp.h"
-
-typedef struct pa_rtp_context {
- int fd;
- uint16_t sequence;
- uint32_t timestamp;
- uint32_t ssrc;
- uint8_t payload;
- size_t frame_size;
- size_t mtu;
-
- pa_memchunk memchunk;
-} pa_rtp_context;
-
-pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, size_t frame_size) {
- pa_rtp_context *c;
-
- pa_assert(fd >= 0);
-
- c = pa_xnew0(pa_rtp_context, 1);
-
- c->fd = fd;
- c->sequence = (uint16_t) (rand()*rand());
- c->timestamp = 0;
- c->ssrc = (uint32_t) (rand()*rand());
- c->payload = (uint8_t) (payload & 127U);
- c->frame_size = frame_size;
- c->mtu = mtu;
-
- pa_memchunk_reset(&c->memchunk);
-
- return c;
-}
-
-#define MAX_IOVECS 16
-
-int pa_rtp_send(pa_rtp_context *c, pa_memblockq *q) {
- struct iovec iov[MAX_IOVECS];
- pa_memblock* mb[MAX_IOVECS];
- int iov_idx = 1;
- size_t n = 0;
-
- pa_assert(c);
- pa_assert(q);
-
- if (pa_memblockq_get_length(q) < c->mtu)
- return 0;
-
- for (;;) {
- int r;
- pa_memchunk chunk;
-
- pa_memchunk_reset(&chunk);
-
- if ((r = pa_memblockq_peek(q, &chunk)) >= 0) {
-
- size_t k = n + chunk.length > c->mtu ? c->mtu - n : chunk.length;
-
- pa_assert(chunk.memblock);
-
- iov[iov_idx].iov_base = pa_memblock_acquire_chunk(&chunk);
- iov[iov_idx].iov_len = k;
- mb[iov_idx] = chunk.memblock;
- iov_idx ++;
-
- n += k;
- pa_memblockq_drop(q, k);
- }
-
- pa_assert(n % c->frame_size == 0);
-
- if (r < 0 || n >= c->mtu || iov_idx >= MAX_IOVECS) {
- uint32_t header[3];
- struct msghdr m;
- ssize_t k;
- int i;
-
- if (n > 0) {
- header[0] = htonl(((uint32_t) 2 << 30) | ((uint32_t) c->payload << 16) | ((uint32_t) c->sequence));
- header[1] = htonl(c->timestamp);
- header[2] = htonl(c->ssrc);
-
- iov[0].iov_base = (void*)header;
- iov[0].iov_len = sizeof(header);
-
- m.msg_name = NULL;
- m.msg_namelen = 0;
- m.msg_iov = iov;
- m.msg_iovlen = (size_t) iov_idx;
- m.msg_control = NULL;
- m.msg_controllen = 0;
- m.msg_flags = 0;
-
- k = sendmsg(c->fd, &m, MSG_DONTWAIT);
-
- for (i = 1; i < iov_idx; i++) {
- pa_memblock_release(mb[i]);
- pa_memblock_unref(mb[i]);
- }
-
- c->sequence++;
- } else
- k = 0;
-
- c->timestamp += (unsigned) (n/c->frame_size);
-
- if (k < 0) {
- if (errno != EAGAIN && errno != EINTR) /* If the queue is full, just ignore it */
- pa_log("sendmsg() failed: %s", pa_cstrerror(errno));
- return -1;
- }
-
- if (r < 0 || pa_memblockq_get_length(q) < c->mtu)
- break;
-
- n = 0;
- iov_idx = 1;
- }
- }
-
- return 0;
-}
-
-pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, size_t frame_size) {
- pa_rtp_context *c;
-
- c = pa_xnew0(pa_rtp_context, 1);
-
- c->fd = fd;
- c->payload = payload;
- c->frame_size = frame_size;
-
- pa_memchunk_reset(&c->memchunk);
-
- return c;
-}
-
-int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_t *rtp_tstamp, struct timeval *tstamp) {
- int size;
- struct msghdr m;
- struct cmsghdr *cm;
- struct iovec iov;
- uint32_t header;
- uint32_t ssrc;
- uint8_t payload;
- unsigned cc;
- ssize_t r;
- uint8_t aux[1024];
- bool found_tstamp = false;
-
- pa_assert(c);
- pa_assert(chunk);
-
- pa_memchunk_reset(chunk);
-
- if (ioctl(c->fd, FIONREAD, &size) < 0) {
- pa_log_warn("FIONREAD failed: %s", pa_cstrerror(errno));
- goto fail;
- }
-
- if (size <= 0) {
- /* size can be 0 due to any of the following reasons:
- *
- * 1. Somebody sent us a perfectly valid zero-length UDP packet.
- * 2. Somebody sent us a UDP packet with a bad CRC.
- *
- * It is unknown whether size can actually be less than zero.
- *
- * In the first case, the packet has to be read out, otherwise the
- * kernel will tell us again and again about it, thus preventing
- * reception of any further packets. So let's just read it out
- * now and discard it later, when comparing the number of bytes
- * received (0) with the number of bytes wanted (1, see below).
- *
- * In the second case, recvmsg() will fail, thus allowing us to
- * return the error.
- *
- * Just to avoid passing zero-sized memchunks and NULL pointers to
- * recvmsg(), let's force allocation of at least one byte by setting
- * size to 1.
- */
- size = 1;
- }
-
- if (c->memchunk.length < (unsigned) size) {
- size_t l;
-
- if (c->memchunk.memblock)
- pa_memblock_unref(c->memchunk.memblock);
-
- l = PA_MAX((size_t) size, pa_mempool_block_size_max(pool));
-
- c->memchunk.memblock = pa_memblock_new(pool, l);
- c->memchunk.index = 0;
- c->memchunk.length = pa_memblock_get_length(c->memchunk.memblock);
- }
-
- pa_assert(c->memchunk.length >= (size_t) size);
-
- chunk->memblock = pa_memblock_ref(c->memchunk.memblock);
- chunk->index = c->memchunk.index;
-
- iov.iov_base = pa_memblock_acquire_chunk(chunk);
- iov.iov_len = (size_t) size;
-
- m.msg_name = NULL;
- m.msg_namelen = 0;
- m.msg_iov = &iov;
- m.msg_iovlen = 1;
- m.msg_control = aux;
- m.msg_controllen = sizeof(aux);
- m.msg_flags = 0;
-
- r = recvmsg(c->fd, &m, 0);
-
- if (r != size) {
- if (r < 0 && errno != EAGAIN && errno != EINTR)
- pa_log_warn("recvmsg() failed: %s", r < 0 ? pa_cstrerror(errno) : "size mismatch");
-
- goto fail;
- }
-
- if (size < 12) {
- pa_log_warn("RTP packet too short.");
- goto fail;
- }
-
- memcpy(&header, iov.iov_base, sizeof(uint32_t));
- memcpy(rtp_tstamp, (uint8_t*) iov.iov_base + 4, sizeof(uint32_t));
- memcpy(&ssrc, (uint8_t*) iov.iov_base + 8, sizeof(uint32_t));
-
- pa_memblock_release(chunk->memblock);
-
- header = ntohl(header);
- *rtp_tstamp = ntohl(*rtp_tstamp);
- ssrc = ntohl(c->ssrc);
-
- if ((header >> 30) != 2) {
- pa_log_warn("Unsupported RTP version.");
- goto fail;
- }
-
- if ((header >> 29) & 1) {
- pa_log_warn("RTP padding not supported.");
- goto fail;
- }
-
- if ((header >> 28) & 1) {
- pa_log_warn("RTP header extensions not supported.");
- goto fail;
- }
-
- if (ssrc != c->ssrc) {
- pa_log_debug("Got unexpected SSRC");
- goto fail;
- }
-
- cc = (header >> 24) & 0xF;
- payload = (uint8_t) ((header >> 16) & 127U);
- c->sequence = (uint16_t) (header & 0xFFFFU);
-
- if (payload != c->payload) {
- pa_log_debug("Got unexpected payload: %u", payload);
- goto fail;
- }
-
- if (12 + cc*4 > (unsigned) size) {
- pa_log_warn("RTP packet too short. (CSRC)");
- goto fail;
- }
-
- chunk->index += 12 + cc*4;
- chunk->length = (size_t) size - 12 + cc*4;
-
- if (chunk->length % c->frame_size != 0) {
- pa_log_warn("Bad RTP packet size.");
- goto fail;
- }
-
- c->memchunk.index = chunk->index + chunk->length;
- c->memchunk.length = pa_memblock_get_length(c->memchunk.memblock) - c->memchunk.index;
-
- if (c->memchunk.length <= 0) {
- pa_memblock_unref(c->memchunk.memblock);
- pa_memchunk_reset(&c->memchunk);
- }
-
- for (cm = CMSG_FIRSTHDR(&m); cm; cm = CMSG_NXTHDR(&m, cm))
- if (cm->cmsg_level == SOL_SOCKET && cm->cmsg_type == SCM_TIMESTAMP) {
- memcpy(tstamp, CMSG_DATA(cm), sizeof(struct timeval));
- found_tstamp = true;
- break;
- }
-
- if (!found_tstamp) {
- pa_log_warn("Couldn't find SCM_TIMESTAMP data in auxiliary recvmsg() data!");
- pa_zero(*tstamp);
- }
-
- return 0;
-
-fail:
- if (chunk->memblock)
- pa_memblock_unref(chunk->memblock);
-
- return -1;
-}
-
-uint8_t pa_rtp_payload_from_sample_spec(const pa_sample_spec *ss) {
- pa_assert(ss);
-
- if (ss->format == PA_SAMPLE_S16BE && ss->rate == 44100 && ss->channels == 2)
- return 10;
- if (ss->format == PA_SAMPLE_S16BE && ss->rate == 44100 && ss->channels == 1)
- return 11;
-
- return 127;
-}
-
-pa_sample_spec *pa_rtp_sample_spec_from_payload(uint8_t payload, pa_sample_spec *ss) {
- pa_assert(ss);
-
- switch (payload) {
- case 10:
- ss->channels = 2;
- ss->format = PA_SAMPLE_S16BE;
- ss->rate = 44100;
- break;
-
- case 11:
- ss->channels = 1;
- ss->format = PA_SAMPLE_S16BE;
- ss->rate = 44100;
- break;
-
- default:
- return NULL;
- }
-
- return ss;
-}
-
-pa_sample_spec *pa_rtp_sample_spec_fixup(pa_sample_spec * ss) {
- pa_assert(ss);
-
- if (!pa_rtp_sample_spec_valid(ss))
- ss->format = PA_SAMPLE_S16BE;
-
- pa_assert(pa_rtp_sample_spec_valid(ss));
- return ss;
-}
-
-int pa_rtp_sample_spec_valid(const pa_sample_spec *ss) {
- pa_assert(ss);
-
- if (!pa_sample_spec_valid(ss))
- return 0;
-
- return ss->format == PA_SAMPLE_S16BE;
-}
-
-void pa_rtp_context_destroy(pa_rtp_context *c) {
- pa_assert(c);
-
- pa_assert_se(pa_close(c->fd) == 0);
-
- if (c->memchunk.memblock)
- pa_memblock_unref(c->memchunk.memblock);
-
- pa_xfree(c);
-}
-
-const char* pa_rtp_format_to_string(pa_sample_format_t f) {
- switch (f) {
- case PA_SAMPLE_S16BE:
- return "L16";
- default:
- return NULL;
- }
-}
-
-pa_sample_format_t pa_rtp_string_to_format(const char *s) {
- pa_assert(s);
-
- if (pa_streq(s, "L16"))
- return PA_SAMPLE_S16BE;
- else
- return PA_SAMPLE_INVALID;
-}
-
-size_t pa_rtp_context_get_frame_size(pa_rtp_context *c) {
- return c->frame_size;
-}
-
-pa_rtpoll_item* pa_rtp_context_get_rtpoll_item(pa_rtp_context *c, pa_rtpoll *rtpoll) {
- pa_rtpoll_item *item;
- struct pollfd *p;
-
- item = pa_rtpoll_item_new(rtpoll, PA_RTPOLL_LATE, 1);
-
- p = pa_rtpoll_item_get_pollfd(item, NULL);
- p->fd = c->fd;
- p->events = POLLIN;
- p->revents = 0;
-
- return item;
-}
diff --git a/src/modules/rtp/rtp.h b/src/modules/rtp/rtp.h
index 1ddc794..c8ded8d 100644
--- a/src/modules/rtp/rtp.h
+++ b/src/modules/rtp/rtp.h
@@ -29,13 +29,13 @@
typedef struct pa_rtp_context pa_rtp_context;
-pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, size_t frame_size);
+pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss);
/* If the memblockq doesn't have a silence memchunk set, then the caller must
* guarantee that the current read index doesn't point to a hole. */
int pa_rtp_send(pa_rtp_context *c, pa_memblockq *q);
-pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, size_t frame_size);
+pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample_spec *ss);
int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_t *rtp_tstamp, struct timeval *tstamp);
void pa_rtp_context_destroy(pa_rtp_context *c);
--
2.5.0
