mirrors/pulseaudio
1
0
Fork 0
mirror of https://gitlab.freedesktop.org/pulseaudio/pulseaudio.git synced 2025-10-29 05:40:23 -04:00
Code Issues Projects Releases Packages Wiki Activity Actions

rtp: Add a GStreamer-based RTP implementation

Browse source 
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 RTCP, non-PCM audio, and potentially synchronised
playback.

Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
This commit is contained in:
Arun Raghavan 2016-05-12 19:26:55 +05:30
parent eb912d3605
commit 74f8456acb
12 changed files with 638 additions and 85 deletions
Download patch file Download diff file Expand all files Collapse all files

18
configure.ac
View file

@ -1310,6 +1310,22 @@ 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$HAVE_GSTREAMER" = "x1"], AC_DEFINE([HAVE_GSTREAMER], 1, [Have GStreamer?]))
#### Build and Install man pages ####
AC_ARG_ENABLE([manpages],
@ -1614,6 +1630,7 @@ AS_IF([test "x$HAVE_ADRIAN_EC" = "x1"], ENABLE_ADRIAN_EC=yes, ENABLE_ADRIAN_EC=n
AS_IF([test "x$HAVE_SPEEX" = "x1"], ENABLE_SPEEX=yes, ENABLE_SPEEX=no)
AS_IF([test "x$HAVE_SOXR" = "x1"], ENABLE_SOXR=yes, ENABLE_SOXR=no)
AS_IF([test "x$HAVE_WEBRTC" = "x1"], ENABLE_WEBRTC=yes, ENABLE_WEBRTC=no)
AS_IF([test "x$HAVE_GSTREAMER" = "x1"], ENABLE_GSTREAMER=yes, ENABLE_GSTREAMER=no)
AS_IF([test "x$HAVE_TDB" = "x1"], ENABLE_TDB=yes, ENABLE_TDB=no)
AS_IF([test "x$HAVE_GDBM" = "x1"], ENABLE_GDBM=yes, ENABLE_GDBM=no)
AS_IF([test "x$HAVE_SIMPLEDB" = "x1"], ENABLE_SIMPLEDB=yes, ENABLE_SIMPLEDB=no)
@ -1677,6 +1694,7 @@ echo "
Enable speex (resampler, AEC): ${ENABLE_SPEEX}
Enable soxr (resampler): ${ENABLE_SOXR}
Enable WebRTC echo canceller: ${ENABLE_WEBRTC}
Enable GStreamer-based RTP: ${ENABLE_GSTREAMER}
Enable gcov coverage: ${ENABLE_GCOV}
Enable unit tests: ${ENABLE_TESTS}
Database

10
meson.build
View file

@ -669,6 +669,15 @@ if webrtc_dep.found()
cdata.set('HAVE_WEBRTC', 1)
endif
gst_dep = dependency('gstreamer-1.0', required : get_option('gstreamer'))
gstapp_dep = dependency('gstreamer-app-1.0', required : get_option('gstreamer'))
gstrtp_dep = dependency('gstreamer-rtp-1.0', required : get_option('gstreamer'))
have_gstreamer = false
if gst_dep.found() and gstapp_dep.found() and gstrtp_dep.found()
have_gstreamer = true
endif
# These are required for the CMake file generation
cdata.set('PA_LIBDIR', libdir)
cdata.set('PA_INCDIR', includedir)
@ -815,6 +824,7 @@ summary = [
'Enable OpenSSL (for Airtunes): @0@'.format(openssl_dep.found()),
'Enable FFTW: @0@'.format(fftw_dep.found()),
'Enable ORC: @0@'.format(have_orcc),
'Enable GStreamer: @0@'.format(have_gstreamer),
'Enable Adrian echo canceller: @0@'.format(get_option('adrian-aec')),
'Enable Speex (resampler, AEC): @0@'.format(speex_dep.found()),
'Enable SoXR (resampler): @0@'.format(soxr_dep.found()),

3
meson_options.txt
View file

@ -93,6 +93,9 @@ option('glib',
option('gsettings',
type : 'feature', value : 'auto',
description : 'Optional GSettings support')
option('gstreamer',
type : 'feature', value : 'auto',
description : 'Optional GStreamer dependency for media-related functionality')
option('gtk',
type : 'feature', value : 'auto',
description : 'Optional Gtk+ 3 support')

4
po/POTFILES.in
View file

@ -66,7 +66,9 @@ src/modules/raop/raop-sink.c
src/modules/reserve-wrap.c
src/modules/rtp/module-rtp-recv.c
src/modules/rtp/module-rtp-send.c
src/modules/rtp/rtp.c
src/modules/rtp/rtp-common.c
src/modules/rtp/rtp-native.c
src/modules/rtp/rtp-gstreamer.c
src/modules/rtp/sap.c
src/modules/rtp/sdp.c
src/modules/x11/module-x11-bell.c

14
src/Makefile.am
View file

@ -1176,13 +1176,21 @@ libprotocol_esound_la_LIBADD = $(AM_LIBADD) libpulsecore-@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-@PA_MAJORMINOR@.la libpulsecommon-@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-util.c modules/raop/raop-util.h \
@ -2049,12 +2057,12 @@ 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) -DPA_MODULE_NAME=module_rtp_send
module_rtp_send_la_CFLAGS = $(AM_CFLAGS) $(GSTREAMER_CFLAGS) -DPA_MODULE_NAME=module_rtp_send
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) -DPA_MODULE_NAME=module_rtp_recv
module_rtp_recv_la_CFLAGS = $(AM_CFLAGS) $(GSTREAMER_CFLAGS) -DPA_MODULE_NAME=module_rtp_recv
# JACK

10
src/modules/rtp/meson.build
View file

@ -1,5 +1,5 @@
librtp_sources = [
'rtp.c',
'rtp-common.c',
'sdp.c',
'sap.c',
'rtsp_client.c',
@ -14,13 +14,19 @@ librtp_headers = [
'headerlist.h',
]
if have_gstreamer
librtp_sources += 'rtp-gstreamer.c'
else
librtp_sources += 'rtp-native.c'
endif
librtp = shared_library('rtp',
librtp_sources,
librtp_headers,
c_args : [pa_c_args, server_c_args],
link_args : [nodelete_link_args],
include_directories : [configinc, topinc],
dependencies : [libpulse_dep, libpulsecommon_dep, libpulsecore_dep, libatomic_ops_dep],
dependencies : [libpulse_dep, libpulsecommon_dep, libpulsecore_dep, libatomic_ops_dep, gst_dep, gstapp_dep, gstrtp_dep, gio_dep],
install : true,
install_rpath : privlibdir,
install_dir : modlibexecdir,

2
src/modules/rtp/module-rtp-recv.c
View file

@ -568,7 +568,7 @@ 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);

2
src/modules/rtp/module-rtp-send.c
View file

@ -488,7 +488,7 @@ 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);

97
src/modules/rtp/rtp-common.c Normal file
View file

@ -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;
}

480
src/modules/rtp/rtp-gstreamer.c Normal file
View file

@ -0,0 +1,480 @@
/***
This file is part of PulseAudio.
Copyright 2016 Arun Raghavan <mail@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)
struct pa_rtp_context {
pa_fdsem *fdsem;
pa_sample_spec ss;
GstElement *pipeline;
GstElement *appsrc;
GstElement *appsink;
uint32_t last_timestamp;
};
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->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_rtp_context_free(c);
return NULL;
}
/* Called from I/O thread context */
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);
}
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);
}
/* Called from I/O thread context */
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));
/* Actually cause the I/O thread to wake up and process the error */
pa_fdsem_post(c->fdsem);
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 receive 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;
}
/* Called from the GStreamer streaming thread */
static void appsink_eos(GstAppSink *appsink, gpointer userdata) {
pa_rtp_context *c = (pa_rtp_context *) userdata;
pa_fdsem_post(c->fdsem);
}
/* Called from the GStreamer streaming thread */
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_rtp_context_free(c);
return NULL;
}
/* Called from I/O thread context */
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_free(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);
}

79
src/modules/rtp/rtp.c → src/modules/rtp/rtp-native.c
View file

@ -58,7 +58,7 @@ typedef struct pa_rtp_context {
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* 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);
@ -70,7 +70,7 @@ pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, siz
c->timestamp = 0;
c->ssrc = (uint32_t) (rand()*rand());
c->payload = (uint8_t) (payload & 127U);
c->frame_size = frame_size;
c->frame_size = pa_frame_size(ss);
c->mtu = mtu;
c->recv_buf = NULL;
@ -169,14 +169,14 @@ int pa_rtp_send(pa_rtp_context *c, pa_memblockq *q) {
return 0;
}
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) {
pa_rtp_context *c;
c = pa_xnew0(pa_rtp_context, 1);
c->fd = fd;
c->payload = payload;
c->frame_size = frame_size;
c->frame_size = pa_frame_size(ss);
c->recv_buf_size = 2000;
c->recv_buf = pa_xmalloc(c->recv_buf_size);
@ -369,59 +369,6 @@ fail:
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_free(pa_rtp_context *c) {
pa_assert(c);
@ -434,24 +381,6 @@ void pa_rtp_context_free(pa_rtp_context *c) {
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;
}

4
src/modules/rtp/rtp.h
View file

@ -30,13 +30,13 @@
typedef struct pa_rtp_context pa_rtp_context;
int pa_rtp_context_init_send(pa_rtp_context *c, 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, 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_free(pa_rtp_context *c);