From: Christophe Fergeau <cfergeau@xxxxxxxxxx>
Changes from v1:
- specifying that it is audio support in commit message
- rebased with last changes
---
To enable audio with GStreamer 1.0: --with-audio=gstreamer1
There is only a few changes between those versions, worth mentioning:
- audio capabilities "audio/x-raw,format=..." instead of
"audio/x-raw-int,..."
- appsink signal for new data changed from "new-buffer" to "new-sample"
---
configure.ac | 17 +++++++++++---
gtk/spice-audio.c | 4 ++--
gtk/spice-gstaudio.c | 64 +++++++++++++++++++++++++++++++++++++++++++++++++++-
3 files changed, 79 insertions(+), 6 deletions(-)
diff --git a/configure.ac b/configure.ac
index ce36d93..71830e2 100644
--- a/configure.ac
+++ b/configure.ac
@@ -288,7 +288,7 @@ AS_IF([test "x$have_phodav" = "xyes"],
AC_DEFINE([USE_PHODAV], [1], [Define if supporting phodav]))
AC_ARG_WITH([audio],
- AS_HELP_STRING([--with-audio=@<:@gstreamer/pulse/auto/no@:>@], [Select audio backend @<:@default=auto@:>@]),
+ AS_HELP_STRING([--with-audio=@<:@gstreamer/gstreamer1/pulse/auto/no@:>@], [Select audio backend @<:@default=auto@:>@]),
[],
[with_audio="auto"])
@@ -297,7 +297,7 @@ AS_IF([test "x$with_audio" = "xauto"], [
])
case "$with_audio" in
- gstreamer|pulse|no*)
+ gstreamer|gstreamer1|pulse|no*)
;;
*) AC_MSG_ERROR(Unsupported audio backend)
esac
@@ -326,7 +326,18 @@ AS_IF([test "x$have_gst" = "xyes"],
[AC_MSG_ERROR([GStreamer requested but not found])
])
])
-AM_CONDITIONAL([WITH_GSTAUDIO], [test "x$have_gst" = "xyes"])
+
+AS_IF([test "x$with_audio" = "xgstreamer1"],
+ [PKG_CHECK_MODULES(GST, gstreamer-1.0 gstreamer-base-1.0 gstreamer-app-1.0 gstreamer-audio-1.0, [have_gst1=yes], [have_gst1=no])],
+ [have_gst1=no])
+
+AS_IF([test "x$have_gst1" = "xyes"],
+ [AC_DEFINE([WITH_GST1AUDIO], 1, [Have GStreamer 1.0?])],
+ [AS_IF([test "x$with_audio" = "xgstreamer1"],
+ [AC_MSG_ERROR([GStreamer 1.0 requested but not found])
+ ])
+])
+AM_CONDITIONAL([WITH_GSTAUDIO], [test "x$have_gst" = "xyes" -o "x$have_gst1" = "xyes"])
AC_SUBST(GST_CFLAGS)
AC_SUBST(GST_LIBS)
diff --git a/gtk/spice-audio.c b/gtk/spice-audio.c
index e7af7d5..638f667 100644
--- a/gtk/spice-audio.c
+++ b/gtk/spice-audio.c
@@ -45,7 +45,7 @@
#ifdef WITH_PULSE
#include "spice-pulse.h"
#endif
-#ifdef WITH_GSTAUDIO
+#if defined(WITH_GSTAUDIO) || defined(WITH_GST1AUDIO)
#include "spice-gstaudio.h"
#endif
@@ -217,7 +217,7 @@ SpiceAudio *spice_audio_new(SpiceSession *session, GMainContext *context,
#ifdef WITH_PULSE
self = SPICE_AUDIO(spice_pulse_new(session, context, name));
#endif
-#ifdef WITH_GSTAUDIO
+#if defined(WITH_GSTAUDIO) || defined(WITH_GST1AUDIO)
self = SPICE_AUDIO(spice_gstaudio_new(session, context, name));
#endif
if (!self)
diff --git a/gtk/spice-gstaudio.c b/gtk/spice-gstaudio.c
index 69bfa78..e6cf6a9 100644
--- a/gtk/spice-gstaudio.c
+++ b/gtk/spice-gstaudio.c
@@ -19,9 +19,13 @@
#include <gst/gst.h>
#include <gst/app/gstappsrc.h>
-#include <gst/app/gstappbuffer.h>
#include <gst/app/gstappsink.h>
+#ifdef WITH_GST1AUDIO
+#include <gst/audio/streamvolume.h>
+#else
+#include <gst/app/gstappbuffer.h>
#include <gst/interfaces/streamvolume.h>
+#endif
#include "spice-gstaudio.h"
#include "spice-common.h"
@@ -131,7 +135,12 @@ static void record_new_buffer(GstAppSink *appsink, gpointer data)
g_return_if_fail(p != NULL);
+#ifdef WITH_GST1AUDIO
+ msg = gst_message_new_application(GST_OBJECT(p->record.pipe),
+ gst_structure_new_empty ("new-sample"));
+#else
msg = gst_message_new_application(GST_OBJECT(p->record.pipe), NULL);
+#endif
gst_element_post_message(p->record.pipe, msg);
}
@@ -153,6 +162,38 @@ static gboolean record_bus_cb(GstBus *bus, GstMessage *msg, gpointer data)
g_return_val_if_fail(p != NULL, FALSE);
switch (GST_MESSAGE_TYPE(msg)) {
+#ifdef WITH_GST1AUDIO
+ case GST_MESSAGE_APPLICATION: {
+ GstSample *s;
+ GstBuffer *buffer;
+ GstMapInfo mapping;
+
+ s = gst_app_sink_pull_sample(GST_APP_SINK(p->record.sink));
+ if (!s) {
+ if (!gst_app_sink_is_eos(GST_APP_SINK(p->record.sink)))
+ g_warning("eos not reached, but can't pull new sample");
+ return TRUE;
+ }
+
+ buffer = gst_sample_get_buffer(s);
+ if (!buffer) {
+ if (!gst_app_sink_is_eos(GST_APP_SINK(p->record.sink)))
+ g_warning("eos not reached, but can't pull new buffer");
+ return TRUE;
+ }
+ if (!gst_buffer_map(buffer, &mapping, GST_MAP_READ)) {
+ return TRUE;
+ }
+
+ spice_record_send_data(SPICE_RECORD_CHANNEL(p->rchannel),
+ /* FIXME: server side doesn't care about ts?
+ what is the unit? ms apparently */
+ mapping.data, mapping.size, 0);
+ gst_buffer_unmap(buffer, &mapping);
+ gst_sample_unref(s);
+ break;
+ }
+#else
case GST_MESSAGE_APPLICATION: {
GstBuffer *b;
@@ -169,6 +210,7 @@ static gboolean record_bus_cb(GstBus *bus, GstMessage *msg, gpointer data)
GST_BUFFER_DATA(b), GST_BUFFER_SIZE(b), 0);
break;
}
+#endif
default:
break;
}
@@ -196,9 +238,15 @@ static void record_start(SpiceRecordChannel *channel, gint format, gint channels
if (!p->record.pipe) {
GError *error = NULL;
GstBus *bus;
+#ifdef WITH_GST1AUDIO
+ gchar *audio_caps =
+ g_strdup_printf("audio/x-raw,format=\"S16LE\",channels=%d,rate=%d,"
+ "layout=interleaved", channels, frequency);
+#else
gchar *audio_caps =
g_strdup_printf("audio/x-raw-int,channels=%d,rate=%d,signed=(boolean)true,"
"width=16,depth=16,endianness=1234", channels, frequency);
+#endif
gchar *pipeline =
g_strdup_printf("autoaudiosrc name=audiosrc ! queue ! audioconvert ! audioresample ! "
"appsink caps=\"%s\" name=appsink", audio_caps);
@@ -219,8 +267,13 @@ static void record_start(SpiceRecordChannel *channel, gint format, gint channels
p->record.channels = channels;
gst_app_sink_set_emit_signals(GST_APP_SINK(p->record.sink), TRUE);
+#ifdef WITH_GST1AUDIO
+ spice_g_signal_connect_object(p->record.sink, "new-sample",
+ G_CALLBACK(record_new_buffer), gstaudio, 0);
+#else
spice_g_signal_connect_object(p->record.sink, "new-buffer",
G_CALLBACK(record_new_buffer), gstaudio, 0);
+#endif
lerr:
g_clear_error(&error);
@@ -311,8 +364,13 @@ static void playback_start(SpicePlaybackChannel *channel, gint format, gint chan
if (!p->playback.pipe) {
GError *error = NULL;
gchar *audio_caps =
+#ifdef WITH_GST1AUDIO
+ g_strdup_printf("audio/x-raw,format=\"S16LE\",channels=%d,rate=%d,"
+ "layout=interleaved", channels, frequency);
+#else
g_strdup_printf("audio/x-raw-int,channels=%d,rate=%d,signed=(boolean)true,"
"width=16,depth=16,endianness=1234", channels, frequency);
+#endif
gchar *pipeline = g_strdup (g_getenv("SPICE_GST_AUDIOSINK"));
if (pipeline == NULL)
pipeline = g_strdup_printf("appsrc is-live=1 do-timestamp=0 caps=\"%s\" name=\"appsrc\" ! queue ! "
@@ -354,7 +412,11 @@ static void playback_data(SpicePlaybackChannel *channel,
g_return_if_fail(p != NULL);
audio = g_memdup(audio, size); /* TODO: try to avoid memory copy */
+#ifdef WITH_GST1AUDIO
+ buf = gst_buffer_new_wrapped(audio, size);
+#else
buf = gst_app_buffer_new(audio, size, g_free, audio);
+#endif
gst_app_src_push_buffer(GST_APP_SRC(p->playback.src), buf);
}
--
2.1.0
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