[pulseaudio-commits] [Git][pulseaudio/pulseaudio][master] rtp: Enable support for OPUS
PulseAudio Marge Bot (@pulseaudio-merge-bot)
gitlab at gitlab.freedesktop.org
Fri Jul 30 13:13:12 UTC 2021
PulseAudio Marge Bot pushed to branch master at PulseAudio / pulseaudio
Commits:
86d1dd0d by Sanchayan Maity at 2021-07-30T13:10:08+00:00
rtp: Enable support for OPUS
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/510>
- - - - -
8 changed files:
- src/modules/rtp/module-rtp-recv.c
- src/modules/rtp/module-rtp-send.c
- src/modules/rtp/rtp-common.c
- src/modules/rtp/rtp-gstreamer.c
- src/modules/rtp/rtp-native.c
- src/modules/rtp/rtp.h
- src/modules/rtp/sdp.c
- src/modules/rtp/sdp.h
Changes:
=====================================
src/modules/rtp/module-rtp-recv.c
=====================================
@@ -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, &s->sdp_info.sample_spec)))
+ if (!(s->rtp_context = pa_rtp_context_new_recv(fd, sdp_info->payload, &s->sdp_info.sample_spec, sdp_info->enable_opus)))
goto fail;
pa_hashmap_put(s->userdata->by_origin, s->sdp_info.origin, s);
=====================================
src/modules/rtp/module-rtp-send.c
=====================================
@@ -67,6 +67,7 @@ PA_MODULE_USAGE(
"ttl=<ttl value> "
"inhibit_auto_suspend=<always|never|only_with_non_monitor_sources>"
"stream_name=<name of the stream>"
+ "enable_opus=<enable OPUS codec>"
);
#define DEFAULT_PORT 46000
@@ -92,6 +93,7 @@ static const char* const valid_modargs[] = {
"ttl",
"inhibit_auto_suspend",
"stream_name",
+ "enable_opus",
NULL
};
@@ -228,6 +230,7 @@ int pa__init(pa_module*m) {
socklen_t k;
char hn[128], *n;
bool loop = false;
+ bool enable_opus = false;
enum inhibit_auto_suspend inhibit_auto_suspend = INHIBIT_AUTO_SUSPEND_ONLY_WITH_NON_MONITOR_SOURCES;
const char *inhibit_auto_suspend_str;
pa_source_output_new_data data;
@@ -249,6 +252,11 @@ int pa__init(pa_module*m) {
goto fail;
}
+ if (pa_modargs_get_value_boolean(ma, "enable_opus", &enable_opus) < 0) {
+ pa_log("Failed to parse \"use_opus\" parameter.");
+ goto fail;
+ }
+
if ((inhibit_auto_suspend_str = pa_modargs_get_value(ma, "inhibit_auto_suspend", NULL))) {
if (pa_streq(inhibit_auto_suspend_str, "always"))
inhibit_auto_suspend = INHIBIT_AUTO_SUSPEND_ALWAYS;
@@ -263,7 +271,7 @@ int pa__init(pa_module*m) {
}
ss = s->sample_spec;
- pa_rtp_sample_spec_fixup(&ss);
+ pa_rtp_sample_spec_fixup(&ss, enable_opus);
cm = s->channel_map;
if (pa_modargs_get_sample_spec(ma, &ss) < 0) {
pa_log("Failed to parse sample specification");
@@ -275,6 +283,11 @@ int pa__init(pa_module*m) {
goto fail;
}
+ if (enable_opus && ss.rate != 48000) {
+ pa_log_warn("OPUS requires sample rate as 48 KHz. Setting rate=48000.");
+ ss.rate = 48000;
+ }
+
if (ss.channels != cm.channels)
pa_channel_map_init_auto(&cm, ss.channels, PA_CHANNEL_MAP_AIFF);
@@ -476,19 +489,19 @@ int pa__init(pa_module*m) {
p = pa_sdp_build(af,
(void*) &((struct sockaddr_in*) &sa_dst)->sin_addr,
(void*) &dst_sa4.sin_addr,
- n, (uint16_t) port, payload, &ss);
+ n, (uint16_t) port, payload, &ss, enable_opus);
#ifdef HAVE_IPV6
} else {
p = pa_sdp_build(af,
(void*) &((struct sockaddr_in6*) &sa_dst)->sin6_addr,
(void*) &dst_sa6.sin6_addr,
- n, (uint16_t) port, payload, &ss);
+ n, (uint16_t) port, payload, &ss, enable_opus);
#endif
}
pa_xfree(n);
- if (!(u->rtp_context = pa_rtp_context_new_send(fd, payload, mtu, &ss)))
+ if (!(u->rtp_context = pa_rtp_context_new_send(fd, payload, mtu, &ss, enable_opus)))
goto fail;
pa_sap_context_init_send(&u->sap_context, sap_fd, p);
=====================================
src/modules/rtp/rtp-common.c
=====================================
@@ -52,6 +52,12 @@ pa_sample_spec *pa_rtp_sample_spec_from_payload(uint8_t payload, pa_sample_spec
ss->rate = 44100;
break;
+ case 127:
+ ss->channels = 2;
+ ss->format = PA_SAMPLE_S16LE;
+ ss->rate = 48000;
+ break;
+
default:
return NULL;
}
@@ -59,10 +65,12 @@ pa_sample_spec *pa_rtp_sample_spec_from_payload(uint8_t payload, pa_sample_spec
return ss;
}
-pa_sample_spec *pa_rtp_sample_spec_fixup(pa_sample_spec * ss) {
+pa_sample_spec *pa_rtp_sample_spec_fixup(pa_sample_spec * ss, bool enable_opus) {
pa_assert(ss);
- if (!pa_rtp_sample_spec_valid(ss))
+ if (!pa_rtp_sample_spec_valid(ss) && enable_opus)
+ ss->format = PA_SAMPLE_S16LE;
+ else if (!pa_rtp_sample_spec_valid(ss) || !enable_opus)
ss->format = PA_SAMPLE_S16BE;
pa_assert(pa_rtp_sample_spec_valid(ss));
@@ -75,22 +83,25 @@ int pa_rtp_sample_spec_valid(const pa_sample_spec *ss) {
if (!pa_sample_spec_valid(ss))
return 0;
- return ss->format == PA_SAMPLE_S16BE;
+ return ss->format == PA_SAMPLE_S16BE || ss->format == PA_SAMPLE_S16LE;
}
const char* pa_rtp_format_to_string(pa_sample_format_t f) {
switch (f) {
case PA_SAMPLE_S16BE:
+ case PA_SAMPLE_S16LE:
return "L16";
default:
return NULL;
}
}
-pa_sample_format_t pa_rtp_string_to_format(const char *s) {
+pa_sample_format_t pa_rtp_string_to_format(const char *s, bool enable_opus) {
pa_assert(s);
- if (pa_streq(s, "L16"))
+ if (pa_streq(s, "L16") && enable_opus)
+ return PA_SAMPLE_S16LE;
+ else if (pa_streq(s, "L16"))
return PA_SAMPLE_S16BE;
else
return PA_SAMPLE_INVALID;
=====================================
src/modules/rtp/rtp-gstreamer.c
=====================================
@@ -45,6 +45,14 @@
#define MAKE_ELEMENT(v, e) MAKE_ELEMENT_NAMED((v), (e), NULL)
#define RTP_HEADER_SIZE 12
+/*
+ * As per RFC 7587, the RTP payload type for OPUS is to be assigned
+ * dynamically. Considering that pa_rtp_payload_from_sample_spec uses
+ * 127 for anything other than format == S16BE and rate == 44.1 KHz,
+ * we use 127 for OPUS here as rate == 48 KHz for OPUS.
+ */
+#define RTP_OPUS_PAYLOAD_TYPE 127
+
struct pa_rtp_context {
pa_fdsem *fdsem;
pa_sample_spec ss;
@@ -61,20 +69,21 @@ struct pa_rtp_context {
size_t mtu;
};
-static GstCaps* caps_from_sample_spec(const pa_sample_spec *ss) {
- if (ss->format != PA_SAMPLE_S16BE)
+static GstCaps* caps_from_sample_spec(const pa_sample_spec *ss, bool enable_opus) {
+ if (ss->format != PA_SAMPLE_S16BE && ss->format != PA_SAMPLE_S16LE)
return NULL;
return gst_caps_new_simple("audio/x-raw",
- "format", G_TYPE_STRING, "S16BE",
+ "format", G_TYPE_STRING, enable_opus ? "S16LE" : "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) {
+static bool init_send_pipeline(pa_rtp_context *c, int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss, bool enable_opus) {
GstElement *appsrc = NULL, *pay = NULL, *capsf = NULL, *rtpbin = NULL, *sink = NULL;
+ GstElement *opusenc = NULL;
GstCaps *caps;
GSocket *socket;
GInetSocketAddress *addr;
@@ -83,7 +92,12 @@ static bool init_send_pipeline(pa_rtp_context *c, int fd, uint8_t payload, size_
gchar *addr_str;
MAKE_ELEMENT(appsrc, "appsrc");
- MAKE_ELEMENT(pay, "rtpL16pay");
+ if (enable_opus) {
+ MAKE_ELEMENT(opusenc, "opusenc");
+ MAKE_ELEMENT(pay, "rtpopuspay");
+ } else {
+ MAKE_ELEMENT(pay, "rtpL16pay");
+ }
MAKE_ELEMENT(capsf, "capsfilter");
MAKE_ELEMENT(rtpbin, "rtpbin");
MAKE_ELEMENT(sink, "udpsink");
@@ -92,7 +106,10 @@ static bool init_send_pipeline(pa_rtp_context *c, int fd, uint8_t payload, size_
gst_bin_add_many(GST_BIN(c->pipeline), appsrc, pay, capsf, rtpbin, sink, NULL);
- caps = caps_from_sample_spec(ss);
+ if (enable_opus)
+ gst_bin_add_many(GST_BIN(c->pipeline), opusenc, NULL);
+
+ caps = caps_from_sample_spec(ss, enable_opus);
if (!caps) {
pa_log("Unsupported format to payload");
goto fail;
@@ -125,17 +142,33 @@ static bool init_send_pipeline(pa_rtp_context *c, int fd, uint8_t payload, size_
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);
+ if (enable_opus)
+ caps = gst_caps_new_simple("application/x-rtp", "payload", G_TYPE_INT, (int) RTP_OPUS_PAYLOAD_TYPE, "encoding-name", G_TYPE_STRING, "OPUS", NULL);
+ else
+ caps = gst_caps_new_simple("application/x-rtp", "payload", G_TYPE_INT, (int) payload, "encoding-name", G_TYPE_STRING, "L16", 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")) {
+ if (enable_opus) {
+ if (!gst_element_link(appsrc, opusenc) ||
+ !gst_element_link(opusenc, 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;
+ pa_log("Could not set up send pipeline");
+ goto fail;
+ }
+ } else {
+ 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) {
@@ -154,6 +187,8 @@ fail:
/* These weren't yet added to pipeline, so we still have a ref */
if (appsrc)
gst_object_unref(appsrc);
+ if (opusenc)
+ gst_object_unref(opusenc);
if (pay)
gst_object_unref(pay);
if (capsf)
@@ -167,7 +202,7 @@ fail:
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* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss, bool enable_opus) {
pa_rtp_context *c = NULL;
GError *error = NULL;
@@ -175,6 +210,9 @@ pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, con
pa_log_info("Initialising GStreamer RTP backend for send");
+ if (enable_opus)
+ pa_log_info("Using OPUS encoding for RTP send");
+
c = pa_xnew0(pa_rtp_context, 1);
c->ss = *ss;
@@ -187,7 +225,7 @@ pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, con
goto fail;
}
- if (!init_send_pipeline(c, fd, payload, mtu, ss))
+ if (!init_send_pipeline(c, fd, payload, mtu, ss, enable_opus))
goto fail;
return c;
@@ -313,10 +351,18 @@ int pa_rtp_send(pa_rtp_context *c, pa_memblockq *q) {
return 0;
}
-static GstCaps* rtp_caps_from_sample_spec(const pa_sample_spec *ss) {
- if (ss->format != PA_SAMPLE_S16BE)
+static GstCaps* rtp_caps_from_sample_spec(const pa_sample_spec *ss, bool enable_opus) {
+ if (ss->format != PA_SAMPLE_S16BE && ss->format != PA_SAMPLE_S16LE)
return NULL;
+ if (enable_opus)
+ return gst_caps_new_simple("application/x-rtp",
+ "media", G_TYPE_STRING, "audio",
+ "encoding-name", G_TYPE_STRING, "OPUS",
+ "clock-rate", G_TYPE_INT, (int) 48000,
+ "payload", G_TYPE_INT, (int) RTP_OPUS_PAYLOAD_TYPE,
+ NULL);
+
return gst_caps_new_simple("application/x-rtp",
"media", G_TYPE_STRING, "audio",
"encoding-name", G_TYPE_STRING, "L16",
@@ -373,22 +419,32 @@ static GstPadProbeReturn udpsrc_buffer_probe(GstPad *pad, GstPadProbeInfo *info,
return GST_PAD_PROBE_OK;
}
-static bool init_receive_pipeline(pa_rtp_context *c, int fd, const pa_sample_spec *ss) {
+static bool init_receive_pipeline(pa_rtp_context *c, int fd, const pa_sample_spec *ss, bool enable_opus) {
GstElement *udpsrc = NULL, *rtpbin = NULL, *depay = NULL, *appsink = NULL;
- GstCaps *caps;
+ GstElement *resample = NULL, *opusdec = NULL;
+ GstCaps *caps, *sink_caps;
GstPad *pad;
GSocket *socket;
GError *error = NULL;
MAKE_ELEMENT(udpsrc, "udpsrc");
MAKE_ELEMENT(rtpbin, "rtpbin");
- MAKE_ELEMENT_NAMED(depay, "rtpL16depay", "depay");
+ if (enable_opus) {
+ MAKE_ELEMENT_NAMED(depay, "rtpopusdepay", "depay");
+ MAKE_ELEMENT(opusdec, "opusdec");
+ MAKE_ELEMENT(resample, "audioresample");
+ } else {
+ 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);
+ if (enable_opus)
+ gst_bin_add_many(GST_BIN(c->pipeline), opusdec, resample, NULL);
+
socket = g_socket_new_from_fd(fd, &error);
if (error) {
pa_log("Could not create socket: %s", error->message);
@@ -396,7 +452,7 @@ static bool init_receive_pipeline(pa_rtp_context *c, int fd, const pa_sample_spe
goto fail;
}
- caps = rtp_caps_from_sample_spec(ss);
+ caps = rtp_caps_from_sample_spec(ss, enable_opus);
if (!caps) {
pa_log("Unsupported format to payload");
goto fail;
@@ -406,14 +462,37 @@ static bool init_receive_pipeline(pa_rtp_context *c, int fd, const pa_sample_spe
g_object_set(rtpbin, "latency", 0, "buffer-mode", 0 /* none */, NULL);
g_object_set(appsink, "sync", FALSE, "enable-last-sample", FALSE, NULL);
+ if (enable_opus) {
+ sink_caps = gst_caps_new_simple("audio/x-raw",
+ "format", G_TYPE_STRING, "S16LE",
+ "layout", G_TYPE_STRING, "interleaved",
+ "clock-rate", G_TYPE_INT, (int) ss->rate,
+ "channels", G_TYPE_INT, (int) ss->channels,
+ NULL);
+ g_object_set(appsink, "caps", sink_caps, NULL);
+ g_object_set(opusdec, "plc", TRUE, NULL);
+ gst_caps_unref(sink_caps);
+ }
+
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)) {
+ if (enable_opus) {
+ if (!gst_element_link_pads(udpsrc, "src", rtpbin, "recv_rtp_sink_0") ||
+ !gst_element_link(depay, opusdec) ||
+ !gst_element_link(opusdec, resample) ||
+ !gst_element_link(resample, appsink)) {
- pa_log("Could not set up receive pipeline");
- goto fail;
+ pa_log("Could not set up receive pipeline");
+ goto fail;
+ }
+ } else {
+ 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);
@@ -446,6 +525,10 @@ fail:
gst_object_unref(depay);
if (rtpbin)
gst_object_unref(rtpbin);
+ if (opusdec)
+ gst_object_unref(opusdec);
+ if (resample)
+ gst_object_unref(resample);
if (appsink)
gst_object_unref(appsink);
}
@@ -469,7 +552,7 @@ static GstFlowReturn appsink_new_sample(GstAppSink *appsink, gpointer userdata)
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* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample_spec *ss, bool enable_opus) {
pa_rtp_context *c = NULL;
GstAppSinkCallbacks callbacks = { 0, };
GError *error = NULL;
@@ -478,6 +561,9 @@ pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample
pa_log_info("Initialising GStreamer RTP backend for receive");
+ if (enable_opus)
+ pa_log_info("Using OPUS encoding for RTP recv");
+
c = pa_xnew0(pa_rtp_context, 1);
c->fdsem = pa_fdsem_new();
@@ -491,7 +577,7 @@ pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample
goto fail;
}
- if (!init_receive_pipeline(c, fd, ss))
+ if (!init_receive_pipeline(c, fd, ss, enable_opus))
goto fail;
callbacks.eos = appsink_eos;
=====================================
src/modules/rtp/rtp-native.c
=====================================
@@ -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, const pa_sample_spec *ss) {
+pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss, bool enable_opus) {
pa_rtp_context *c;
pa_assert(fd >= 0);
@@ -171,7 +171,7 @@ 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, const pa_sample_spec *ss) {
+pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample_spec *ss, bool enable_opus) {
pa_rtp_context *c;
pa_log_info("Initialising native RTP backend for receive");
=====================================
src/modules/rtp/rtp.h
=====================================
@@ -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, const pa_sample_spec *ss);
+pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss, bool enable_opus);
/* 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, const pa_sample_spec *ss);
+pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample_spec *ss, bool enable_opus);
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);
@@ -44,13 +44,13 @@ void pa_rtp_context_free(pa_rtp_context *c);
size_t pa_rtp_context_get_frame_size(pa_rtp_context *c);
pa_rtpoll_item* pa_rtp_context_get_rtpoll_item(pa_rtp_context *c, pa_rtpoll *rtpoll);
-pa_sample_spec* pa_rtp_sample_spec_fixup(pa_sample_spec *ss);
+pa_sample_spec* pa_rtp_sample_spec_fixup(pa_sample_spec *ss, bool enable_opus);
int pa_rtp_sample_spec_valid(const pa_sample_spec *ss);
uint8_t pa_rtp_payload_from_sample_spec(const pa_sample_spec *ss);
pa_sample_spec *pa_rtp_sample_spec_from_payload(uint8_t payload, pa_sample_spec *ss);
const char* pa_rtp_format_to_string(pa_sample_format_t f);
-pa_sample_format_t pa_rtp_string_to_format(const char *s);
+pa_sample_format_t pa_rtp_string_to_format(const char *s, bool enable_opus);
#endif
=====================================
src/modules/rtp/sdp.c
=====================================
@@ -39,8 +39,9 @@
#include "sdp.h"
#include "rtp.h"
-char *pa_sdp_build(int af, const void *src, const void *dst, const char *name, uint16_t port, uint8_t payload, const pa_sample_spec *ss) {
+char *pa_sdp_build(int af, const void *src, const void *dst, const char *name, uint16_t port, uint8_t payload, const pa_sample_spec *ss, bool enable_opus) {
uint32_t ntp;
+ uint32_t rate, channels;
char buf_src[64], buf_dst[64], un[64];
const char *u, *f;
@@ -53,7 +54,15 @@ char *pa_sdp_build(int af, const void *src, const void *dst, const char *name, u
pa_assert(af == AF_INET);
#endif
- pa_assert_se(f = pa_rtp_format_to_string(ss->format));
+ if (enable_opus) {
+ f = "OPUS";
+ rate = 48000;
+ channels = 2;
+ } else {
+ pa_assert_se(f = pa_rtp_format_to_string(ss->format));
+ rate = ss->rate;
+ channels = ss->channels;
+ }
if (!(u = pa_get_user_name(un, sizeof(un))))
u = "-";
@@ -78,7 +87,7 @@ char *pa_sdp_build(int af, const void *src, const void *dst, const char *name, u
af == AF_INET ? "IP4" : "IP6", buf_dst,
(unsigned long) ntp,
port, payload,
- payload, f, ss->rate, ss->channels);
+ payload, f, rate, channels);
}
static pa_sample_spec *parse_sdp_sample_spec(pa_sample_spec *ss, char *c) {
@@ -89,6 +98,9 @@ static pa_sample_spec *parse_sdp_sample_spec(pa_sample_spec *ss, char *c) {
if (pa_startswith(c, "L16/")) {
ss->format = PA_SAMPLE_S16BE;
c += 4;
+ } else if (pa_startswith(c, "OPUS/")) {
+ ss->format = PA_SAMPLE_S16LE;
+ c += 5;
} else
return NULL;
@@ -218,6 +230,9 @@ pa_sdp_info *pa_sdp_parse(const char *t, pa_sdp_info *i, int is_goodbye) {
if (parse_sdp_sample_spec(&i->sample_spec, c))
ss_valid = true;
+
+ if (pa_startswith(c, "OPUS/"))
+ i->enable_opus = true;
}
}
}
=====================================
src/modules/rtp/sdp.h
=====================================
@@ -37,9 +37,11 @@ typedef struct pa_sdp_info {
pa_sample_spec sample_spec;
uint8_t payload;
+
+ bool enable_opus;
} pa_sdp_info;
-char *pa_sdp_build(int af, const void *src, const void *dst, const char *name, uint16_t port, uint8_t payload, const pa_sample_spec *ss);
+char *pa_sdp_build(int af, const void *src, const void *dst, const char *name, uint16_t port, uint8_t payload, const pa_sample_spec *ss, bool enable_opus);
pa_sdp_info *pa_sdp_parse(const char *t, pa_sdp_info *info, int is_goodbye);
View it on GitLab: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/86d1dd0d70d6943cb67346c6187171444f764774
--
View it on GitLab: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/86d1dd0d70d6943cb67346c6187171444f764774
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