[libdmapsharing] Replace some assertions in dmap-gst-*-input-stream.c with error handling
- From: W. Michael Petullo <wmpetullo src gnome org>
- To: commits-list gnome org
- Cc:
- Subject: [libdmapsharing] Replace some assertions in dmap-gst-*-input-stream.c with error handling
- Date: Wed, 21 Aug 2013 02:12:49 +0000 (UTC)
commit 8c3fd3791a30cbe0d62f3cd9953a911fb6313e8b
Author: W. Michael Petullo <mike flyn org>
Date: Tue Aug 20 22:12:37 2013 -0400
Replace some assertions in dmap-gst-*-input-stream.c with error handling
Signed-off-by: W. Michael Petullo <mike flyn org>
libdmapsharing/daap-share.c | 2 +-
libdmapsharing/dmap-gst-mp3-input-stream.c | 149 ++++++++++++++--------
libdmapsharing/dmap-gst-qt-input-stream.c | 178 +++++++++++++++++---------
libdmapsharing/dmap-gst-wav-input-stream.c | 191 ++++++++++++++++++----------
4 files changed, 336 insertions(+), 184 deletions(-)
---
diff --git a/libdmapsharing/daap-share.c b/libdmapsharing/daap-share.c
index f251f47..6b719f5 100644
--- a/libdmapsharing/daap-share.c
+++ b/libdmapsharing/daap-share.c
@@ -543,7 +543,7 @@ _error:
}
if (NULL != stream) {
- g_input_stream_close (cd->stream, NULL, NULL);
+ g_input_stream_close (stream, NULL, NULL);
}
return;
diff --git a/libdmapsharing/dmap-gst-mp3-input-stream.c b/libdmapsharing/dmap-gst-mp3-input-stream.c
index 234fa74..b21b631 100644
--- a/libdmapsharing/dmap-gst-mp3-input-stream.c
+++ b/libdmapsharing/dmap-gst-mp3-input-stream.c
@@ -45,18 +45,18 @@ void dmap_gst_input_stream_new_buffer_cb (GstElement * element,
static void
pad_added_cb (GstElement * element,
GstPad * pad,
- DMAPGstMP3InputStream * stream)
+ GstElement *convert)
{
/* Link remaining pad after decodebin2 does its magic. */
GstPad *conv_pad;
- conv_pad = gst_element_get_static_pad (stream->priv->convert, "sink");
+ conv_pad = gst_element_get_static_pad (convert, "sink");
g_assert (conv_pad != NULL);
if (pads_compatible (pad, conv_pad)) {
g_assert (!GST_PAD_IS_LINKED
(gst_element_get_static_pad
- (stream->priv->convert, "sink")));
+ (convert, "sink")));
gst_pad_link (pad, conv_pad);
} else {
@@ -69,85 +69,132 @@ dmap_gst_mp3_input_stream_new (GInputStream * src_stream)
{
GstStateChangeReturn sret;
GstState state;
- DMAPGstMP3InputStream *stream;
+ DMAPGstMP3InputStream *stream = NULL;
- stream = DMAP_GST_MP3_INPUT_STREAM (g_object_new
- (DMAP_TYPE_GST_MP3_INPUT_STREAM,
- NULL));
+ GstElement *pipeline = NULL;
+ GstElement *src = NULL;
+ GstElement *decode = NULL;
+ GstElement *convert = NULL;
+ GstElement *audio_encode = NULL;
+ GstElement *sink = NULL;
- stream->priv->pipeline = gst_pipeline_new ("pipeline");
+ g_assert (G_IS_INPUT_STREAM (src_stream));
- stream->priv->src = gst_element_factory_make ("giostreamsrc", "src");
- g_assert (GST_IS_ELEMENT (stream->priv->src));
+ pipeline = gst_pipeline_new ("pipeline");
+ if (NULL == pipeline) {
+ g_warning ("Could not create GStreamer pipeline");
+ goto done;
+ }
- stream->priv->decode = gst_element_factory_make ("decodebin", "decode");
- g_assert (GST_IS_ELEMENT (stream->priv->decode));
+ src = gst_element_factory_make ("giostreamsrc", "src");
+ if (NULL == src) {
+ g_warning ("Could not create GStreamer giostreamsrc element");
+ goto done;
+ }
- stream->priv->convert = gst_element_factory_make ("audioconvert", "convert");
- g_assert (GST_IS_ELEMENT (stream->priv->convert));
+ decode = gst_element_factory_make ("decodebin", "decode");
+ if (NULL == decode) {
+ g_warning ("Could not create GStreamer decodebin element");
+ goto done;
+ }
- stream->priv->audio_encode = gst_element_factory_make ("lamemp3enc", "audioencode");
- g_assert (GST_IS_ELEMENT (stream->priv->audio_encode));
+ convert = gst_element_factory_make ("audioconvert", "convert");
+ if (NULL == convert) {
+ g_warning ("Could not create GStreamer audioconvert element");
+ goto done;
+ }
- stream->priv->sink = gst_element_factory_make ("appsink", "sink");
- g_assert (GST_IS_ELEMENT (stream->priv->sink));
+ audio_encode = gst_element_factory_make ("lamemp3enc", "audioencode");
+ if (NULL == audio_encode) {
+ g_warning ("Could not create GStreamer lamemp3enc element");
+ goto done;
+ }
- gst_bin_add_many (GST_BIN (stream->priv->pipeline),
- stream->priv->src,
- stream->priv->decode,
- stream->priv->convert,
- stream->priv->audio_encode,
- stream->priv->sink,
- NULL);
+ sink = gst_element_factory_make ("appsink", "sink");
+ if (NULL == sink) {
+ g_warning ("Could not create GStreamer appsink element");
+ goto done;
+ }
- if (gst_element_link (stream->priv->src,
- stream->priv->decode) == FALSE) {
+ gst_bin_add_many (GST_BIN (pipeline), src, decode, convert, audio_encode, sink, NULL);
+
+ if (FALSE == gst_element_link (src, decode)) {
g_warning ("Error linking source and decode elements");
+ goto done;
}
- if (gst_element_link_many (stream->priv->convert,
- stream->priv->audio_encode,
- stream->priv->sink,
- NULL) == FALSE) {
+ if (FALSE == gst_element_link_many (convert, audio_encode, sink, NULL)) {
g_warning ("Error linking convert through sink elements");
+ goto done;
}
- g_assert (G_IS_INPUT_STREAM (src_stream));
- g_object_set (G_OBJECT (stream->priv->src), "stream", src_stream,
- NULL);
+ g_object_set (G_OBJECT (src), "stream", src_stream, NULL);
/* quality=9 is important for fast, realtime transcoding: */
// FIXME: Causes crash; why?
- // g_object_set (G_OBJECT (stream->priv->audio_encode), "quality", 9, NULL);
- g_object_set (G_OBJECT (stream->priv->audio_encode), "bitrate", 128, NULL);
- g_object_set (G_OBJECT (stream->priv->audio_encode), "vbr", 0, NULL);
- g_signal_connect (stream->priv->decode, "pad-added",
- G_CALLBACK (pad_added_cb), stream);
+ // g_object_set (G_OBJECT (audio_encode), "quality", 9, NULL);
+ g_object_set (G_OBJECT (audio_encode), "bitrate", 128, NULL);
+ g_object_set (G_OBJECT (audio_encode), "vbr", 0, NULL);
- g_object_set (G_OBJECT (stream->priv->sink), "emit-signals", TRUE,
- "sync", FALSE, NULL);
- gst_app_sink_set_max_buffers (GST_APP_SINK (stream->priv->sink), GST_APP_MAX_BUFFERS);
- gst_app_sink_set_drop (GST_APP_SINK (stream->priv->sink), FALSE);
+ g_object_set (G_OBJECT (sink), "emit-signals", TRUE, "sync", FALSE, NULL);
+ gst_app_sink_set_max_buffers (GST_APP_SINK (sink), GST_APP_MAX_BUFFERS);
+ gst_app_sink_set_drop (GST_APP_SINK (sink), FALSE);
- g_signal_connect (stream->priv->sink, "new-sample",
- G_CALLBACK (dmap_gst_input_stream_new_buffer_cb),
- stream);
+ g_signal_connect (decode, "pad-added", G_CALLBACK (pad_added_cb), convert);
/* FIXME: this technique is shared with dmapd-daap-share.c */
- sret = gst_element_set_state (stream->priv->pipeline,
- GST_STATE_PLAYING);
+ sret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (GST_STATE_CHANGE_ASYNC == sret) {
if (GST_STATE_CHANGE_SUCCESS !=
- gst_element_get_state (GST_ELEMENT
- (stream->priv->pipeline), &state,
- NULL, 5 * GST_SECOND)) {
+ gst_element_get_state (GST_ELEMENT (pipeline), &state, NULL, 5 * GST_SECOND)) {
g_warning ("State change failed for stream.");
+ goto done;
}
} else if (sret != GST_STATE_CHANGE_SUCCESS) {
g_warning ("Could not read stream.");
+ goto done;
}
+ stream = DMAP_GST_MP3_INPUT_STREAM (g_object_new (DMAP_TYPE_GST_MP3_INPUT_STREAM, NULL));
+ if (NULL == stream) {
+ goto done;
+ }
g_assert (G_IS_SEEKABLE (stream));
+
+ g_signal_connect (sink, "new-sample", G_CALLBACK (dmap_gst_input_stream_new_buffer_cb), stream);
+
+ stream->priv->pipeline = gst_object_ref (pipeline);
+ stream->priv->src = gst_object_ref (src);
+ stream->priv->decode = gst_object_ref (decode);
+ stream->priv->convert = gst_object_ref (convert);
+ stream->priv->audio_encode = gst_object_ref (audio_encode);
+ stream->priv->sink = gst_object_ref (sink);
+
+done:
+ if (pipeline) {
+ gst_object_unref (pipeline);
+ }
+
+ if (src) {
+ gst_object_unref (src);
+ }
+
+ if (decode) {
+ gst_object_unref (decode);
+ }
+
+ if (convert) {
+ gst_object_unref (convert);
+ }
+
+ if (audio_encode) {
+ gst_object_unref (audio_encode);
+ }
+
+ if (sink) {
+ gst_object_unref (sink);
+ }
+
return G_INPUT_STREAM (stream);
}
diff --git a/libdmapsharing/dmap-gst-qt-input-stream.c b/libdmapsharing/dmap-gst-qt-input-stream.c
index eff4253..5be6215 100644
--- a/libdmapsharing/dmap-gst-qt-input-stream.c
+++ b/libdmapsharing/dmap-gst-qt-input-stream.c
@@ -46,18 +46,18 @@ void dmap_gst_input_stream_new_buffer_cb (GstElement * element,
static void
pad_added_cb (GstElement * element,
GstPad * pad,
- DMAPGstQtInputStream * stream)
+ GstElement *convert)
{
/* Link remaining pad after decodebin2 does its magic. */
GstPad *conv_pad;
- conv_pad = gst_element_get_static_pad (stream->priv->convert, "sink");
+ conv_pad = gst_element_get_static_pad (convert, "sink");
g_assert (conv_pad != NULL);
if (pads_compatible (pad, conv_pad)) {
g_assert (!GST_PAD_IS_LINKED
(gst_element_get_static_pad
- (stream->priv->convert, "sink")));
+ (convert, "sink")));
gst_pad_link (pad, conv_pad);
} else {
@@ -70,88 +70,138 @@ dmap_gst_qt_input_stream_new (GInputStream * src_stream)
{
GstStateChangeReturn sret;
GstState state;
- DMAPGstQtInputStream *stream;
+ DMAPGstQtInputStream *stream = NULL;
- stream = DMAP_GST_QT_INPUT_STREAM (g_object_new
- (DMAP_TYPE_GST_QT_INPUT_STREAM,
- NULL));
+ GstElement *pipeline = NULL;
+ GstElement *src = NULL;
+ GstElement *decode = NULL;
+ GstElement *convert = NULL;
+ GstElement *audio_encode = NULL;
+ GstElement *mux = NULL;
+ GstElement *sink = NULL;
- stream->priv->pipeline = gst_pipeline_new ("pipeline");
-
- stream->priv->src = gst_element_factory_make ("giostreamsrc", "src");
- g_assert (GST_IS_ELEMENT (stream->priv->src));
-
- stream->priv->decode =
- gst_element_factory_make ("decodebin", "decode");
- g_assert (GST_IS_ELEMENT (stream->priv->decode));
-
- stream->priv->convert =
- gst_element_factory_make ("audioconvert", "convert");
- g_assert (GST_IS_ELEMENT (stream->priv->convert));
-
- stream->priv->audio_encode = gst_element_factory_make ("avenc_aac", "audioencode");
- g_assert (GST_IS_ELEMENT (stream->priv->audio_encode));
-
- stream->priv->mux = gst_element_factory_make ("qtmux", "mux");
- g_assert (GST_IS_ELEMENT (stream->priv->mux));
-
- stream->priv->sink = gst_element_factory_make ("appsink", "sink");
- g_assert (GST_IS_ELEMENT (stream->priv->sink));
-
- gst_bin_add_many (GST_BIN (stream->priv->pipeline),
- stream->priv->src,
- stream->priv->decode,
- stream->priv->convert,
- stream->priv->audio_encode,
- stream->priv->mux,
- stream->priv->sink,
- NULL);
+ g_assert (G_IS_INPUT_STREAM (src_stream));
- if (gst_element_link (stream->priv->src,
- stream->priv->decode) == FALSE) {
+ pipeline = gst_pipeline_new ("pipeline");
+ if (NULL == pipeline) {
+ g_warning ("Could not create GStreamer pipeline");
+ goto done;
+ }
+
+ src = gst_element_factory_make ("giostreamsrc", "src");
+ if (NULL == src) {
+ g_warning ("Could not create GStreamer giostreamsrc element");
+ goto done;
+ }
+
+ decode = gst_element_factory_make ("decodebin", "decode");
+ if (NULL == decode) {
+ g_warning ("Could not create GStreamer decodebin element");
+ goto done;
+ }
+
+ convert = gst_element_factory_make ("audioconvert", "convert");
+ if (NULL == convert) {
+ g_warning ("Could not create GStreamer audioconvert element");
+ goto done;
+ }
+
+ audio_encode = gst_element_factory_make ("avenc_aac", "audioencode");
+ if (NULL == audio_encode) {
+ g_warning ("Could not create GStreamer avenc_aac element");
+ goto done;
+ }
+
+ mux = gst_element_factory_make ("qtmux", "mux");
+ if (NULL == audio_encode) {
+ g_warning ("Could not create GStreamer qtmux element");
+ goto done;
+ }
+
+ sink = gst_element_factory_make ("appsink", "sink");
+ if (NULL == sink) {
+ g_warning ("Could not create GStreamer appsink element");
+ goto done;
+ }
+
+ gst_bin_add_many (GST_BIN (pipeline), src, decode, convert, audio_encode, mux, sink, NULL);
+
+ if (FALSE == gst_element_link (stream->priv->src, stream->priv->decode)) {
g_warning ("Error linking source and decode elements");
+ goto done;
}
- if (gst_element_link_many (stream->priv->convert,
- stream->priv->audio_encode,
- stream->priv->mux,
- stream->priv->sink,
- NULL) == FALSE) {
+ if (FALSE == gst_element_link_many (convert, audio_encode, mux, sink, NULL)) {
g_warning ("Error linking convert through sink elements");
+ goto done;
}
- g_assert (G_IS_INPUT_STREAM (src_stream));
- g_object_set (G_OBJECT (stream->priv->src), "stream", src_stream,
- NULL);
-
- g_signal_connect (stream->priv->decode, "pad-added",
- G_CALLBACK (pad_added_cb), stream);
+ g_object_set (G_OBJECT (src), "stream", src_stream, NULL);
- g_object_set (G_OBJECT (stream->priv->sink), "emit-signals", TRUE,
- "sync", FALSE, NULL);
- gst_app_sink_set_max_buffers (GST_APP_SINK (stream->priv->sink),
- GST_APP_MAX_BUFFERS);
- gst_app_sink_set_drop (GST_APP_SINK (stream->priv->sink), FALSE);
+ g_object_set (G_OBJECT (sink), "emit-signals", TRUE, "sync", FALSE, NULL);
+ gst_app_sink_set_max_buffers (GST_APP_SINK (sink), GST_APP_MAX_BUFFERS);
+ gst_app_sink_set_drop (GST_APP_SINK (sink), FALSE);
- g_signal_connect (stream->priv->sink, "new-sample",
- G_CALLBACK (dmap_gst_input_stream_new_buffer_cb),
- stream);
+ g_signal_connect (decode, "pad-added", G_CALLBACK (pad_added_cb), convert);
/* FIXME: this technique is shared with dmapd-daap-share.c */
- sret = gst_element_set_state (stream->priv->pipeline,
- GST_STATE_PLAYING);
+ sret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (GST_STATE_CHANGE_ASYNC == sret) {
if (GST_STATE_CHANGE_SUCCESS !=
- gst_element_get_state (GST_ELEMENT
- (stream->priv->pipeline), &state,
- NULL, 5 * GST_SECOND)) {
+ gst_element_get_state (GST_ELEMENT (pipeline), &state, NULL, 5 * GST_SECOND)) {
g_warning ("State change failed for stream.");
+ goto done;
}
} else if (sret != GST_STATE_CHANGE_SUCCESS) {
g_warning ("Could not read stream.");
+ goto done;
}
- g_assert (G_IS_SEEKABLE (stream));
+ stream = DMAP_GST_QT_INPUT_STREAM (g_object_new (DMAP_TYPE_GST_QT_INPUT_STREAM, NULL));
+ if (NULL == stream) {
+ goto done;
+ }
+ g_assert (G_IS_SEEKABLE (stream));
+
+ g_signal_connect (sink, "new-sample", G_CALLBACK (dmap_gst_input_stream_new_buffer_cb), stream);
+
+ stream->priv->pipeline = gst_object_ref (pipeline);
+ stream->priv->src = gst_object_ref (src);
+ stream->priv->decode = gst_object_ref (decode);
+ stream->priv->convert = gst_object_ref (convert);
+ stream->priv->audio_encode = gst_object_ref (audio_encode);
+ stream->priv->mux = gst_object_ref (mux);
+ stream->priv->sink = gst_object_ref (sink);
+
+done:
+ if (pipeline) {
+ gst_object_unref (pipeline);
+ }
+
+ if (src) {
+ gst_object_unref (src);
+ }
+
+ if (decode) {
+ gst_object_unref (decode);
+ }
+
+ if (convert) {
+ gst_object_unref (convert);
+ }
+
+ if (audio_encode) {
+ gst_object_unref (audio_encode);
+ }
+
+ if (mux) {
+ gst_object_unref (mux);
+ }
+
+ if (sink) {
+ gst_object_unref (sink);
+ }
+
return G_INPUT_STREAM (stream);
}
diff --git a/libdmapsharing/dmap-gst-wav-input-stream.c b/libdmapsharing/dmap-gst-wav-input-stream.c
index c7817ba..0e98dce 100644
--- a/libdmapsharing/dmap-gst-wav-input-stream.c
+++ b/libdmapsharing/dmap-gst-wav-input-stream.c
@@ -46,18 +46,18 @@ void dmap_gst_input_stream_new_buffer_cb (GstElement * element,
static void
pad_added_cb (GstElement * element,
GstPad * pad,
- DMAPGstWAVInputStream * stream)
+ GstElement *convert)
{
/* Link remaining pad after decodebin2 does its magic. */
GstPad *conv_pad;
- conv_pad = gst_element_get_static_pad (stream->priv->convert, "sink");
+ conv_pad = gst_element_get_static_pad (convert, "sink");
g_assert (conv_pad != NULL);
if (pads_compatible (pad, conv_pad)) {
g_assert (!GST_PAD_IS_LINKED
(gst_element_get_static_pad
- (stream->priv->convert, "sink")));
+ (convert, "sink")));
gst_pad_link (pad, conv_pad);
} else {
@@ -70,92 +70,147 @@ dmap_gst_wav_input_stream_new (GInputStream * src_stream)
{
GstStateChangeReturn sret;
GstState state;
- DMAPGstWAVInputStream *stream;
-
- stream = DMAP_GST_WAV_INPUT_STREAM (g_object_new
- (DMAP_TYPE_GST_WAV_INPUT_STREAM,
- NULL));
-
- stream->priv->pipeline = gst_pipeline_new ("pipeline");
-
- stream->priv->src = gst_element_factory_make ("giostreamsrc", "src");
- g_assert (GST_IS_ELEMENT (stream->priv->src));
-
- stream->priv->decode =
- gst_element_factory_make ("decodebin", "decode");
- g_assert (GST_IS_ELEMENT (stream->priv->decode));
-
- stream->priv->convert =
- gst_element_factory_make ("audioconvert", "convert");
- g_assert (GST_IS_ELEMENT (stream->priv->convert));
-
+ DMAPGstWAVInputStream *stream = NULL;
+
+ GstElement *pipeline = NULL;
+ GstElement *src = NULL;
+ GstElement *decode = NULL;
+ GstElement *convert = NULL;
+ GstCaps *filter = NULL;
+ GstElement *audio_encode = NULL;
+ GstElement *sink = NULL;
+
+ g_assert (G_IS_INPUT_STREAM (src_stream));
+
+ pipeline = gst_pipeline_new ("pipeline");
+ if (NULL == pipeline) {
+ g_warning ("Could not create GStreamer pipeline");
+ goto done;
+ }
+
+ src = gst_element_factory_make ("giostreamsrc", "src");
+ if (NULL == src) {
+ g_warning ("Could not create GStreamer giostreamsrc element");
+ goto done;
+ }
+
+ decode = gst_element_factory_make ("decodebin", "decode");
+ if (NULL == decode) {
+ g_warning ("Could not create GStreamer decodebin element");
+ goto done;
+ }
+
+ convert = gst_element_factory_make ("audioconvert", "convert");
+ if (NULL == convert) {
+ g_warning ("Could not create GStreamer audioconvert element");
+ goto done;
+ }
+
+ /* FIXME: This needs to be retested with Roku hardware after GStreamer 1.0 upgrade. */
/* Roku clients support a subset of the WAV format. */
- stream->priv->filter = gst_caps_new_simple ("audio/x-raw-int",
- "channels", G_TYPE_INT, 2,
- "width", G_TYPE_INT, 16,
- "depth", G_TYPE_INT, 16,
- NULL);
- stream->priv->audio_encode = gst_element_factory_make ("wavenc", "audioencode");
- g_assert (GST_IS_ELEMENT (stream->priv->audio_encode));
-
- stream->priv->sink = gst_element_factory_make ("appsink", "sink");
- g_assert (GST_IS_ELEMENT (stream->priv->sink));
-
- gst_bin_add_many (GST_BIN (stream->priv->pipeline),
- stream->priv->src,
- stream->priv->decode,
- stream->priv->convert,
- stream->priv->audio_encode,
- stream->priv->sink,
- NULL);
-
- if (gst_element_link (stream->priv->src,
- stream->priv->decode) == FALSE) {
- g_warning ("Error linking source through decode elements");
+ filter = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, "S16LE",
+ "channels", G_TYPE_INT, 2,
+ /* Pre-GStreamer 1.0 "width", G_TYPE_INT, 16,
+ * "depth", G_TYPE_INT, 16,
+ */
+ NULL);
+
+ audio_encode = gst_element_factory_make ("wavenc", "audioencode");
+ if (NULL == audio_encode) {
+ g_warning ("Could not create GStreamer wavenc element");
+ goto done;
+ }
+
+ sink = gst_element_factory_make ("appsink", "sink");
+ if (NULL == sink) {
+ g_warning ("Could not create GStreamer appsink element");
+ goto done;
+ }
+
+ gst_bin_add_many (GST_BIN (pipeline), src, decode, convert, audio_encode, sink, NULL);
+
+ if (FALSE == gst_element_link (src, decode)) {
+ g_warning ("Error linking source and decode elements");
+ goto done;
}
- if (gst_element_link_filtered (stream->priv->convert,
- stream->priv->audio_encode,
- stream->priv->filter) == FALSE) {
+ if (FALSE == gst_element_link_filtered (convert, audio_encode, filter)) {
g_warning ("Error linking convert and audioencode elements");
+ goto done;
}
- if (gst_element_link (stream->priv->audio_encode,
- stream->priv->sink) == FALSE) {
+ if (FALSE == gst_element_link (audio_encode, sink)) {
g_warning ("Error linking audioencode and sink elements");
+ goto done;
}
- g_object_set (G_OBJECT (stream->priv->src), "stream", src_stream,
- NULL);
+ g_object_set (G_OBJECT (src), "stream", src_stream, NULL);
- g_signal_connect (stream->priv->decode, "pad-added",
- G_CALLBACK (pad_added_cb), stream);
+ g_object_set (G_OBJECT (sink), "emit-signals", TRUE, "sync", FALSE, NULL);
+ gst_app_sink_set_max_buffers (GST_APP_SINK (sink), GST_APP_MAX_BUFFERS);
+ gst_app_sink_set_drop (GST_APP_SINK (sink), FALSE);
- g_object_set (G_OBJECT (stream->priv->sink), "emit-signals", TRUE,
- "sync", FALSE, NULL);
- gst_app_sink_set_max_buffers (GST_APP_SINK (stream->priv->sink),
- GST_APP_MAX_BUFFERS);
- gst_app_sink_set_drop (GST_APP_SINK (stream->priv->sink), FALSE);
-
- g_signal_connect (stream->priv->sink, "new-buffer",
- G_CALLBACK (dmap_gst_input_stream_new_buffer_cb),
- stream);
+ g_signal_connect (decode, "pad-added", G_CALLBACK (pad_added_cb), convert);
/* FIXME: this technique is shared with dmapd-daap-share.c */
- sret = gst_element_set_state (stream->priv->pipeline,
- GST_STATE_PLAYING);
+ sret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (GST_STATE_CHANGE_ASYNC == sret) {
if (GST_STATE_CHANGE_SUCCESS !=
- gst_element_get_state (GST_ELEMENT
- (stream->priv->pipeline), &state,
- NULL, 5 * GST_SECOND)) {
+ gst_element_get_state (GST_ELEMENT (pipeline), &state, NULL, 5 * GST_SECOND)) {
g_warning ("State change failed for stream.");
+ goto done;
}
} else if (sret != GST_STATE_CHANGE_SUCCESS) {
g_warning ("Could not read stream.");
+ goto done;
+ }
+
+ stream = DMAP_GST_WAV_INPUT_STREAM (g_object_new (DMAP_TYPE_GST_WAV_INPUT_STREAM, NULL));
+ if (NULL == stream) {
+ goto done;
+ }
+ g_assert (G_IS_SEEKABLE (stream));
+
+ g_signal_connect (sink, "new-sample", G_CALLBACK (dmap_gst_input_stream_new_buffer_cb), stream);
+
+ stream->priv->pipeline = gst_object_ref (pipeline);
+ stream->priv->src = gst_object_ref (src);
+ stream->priv->decode = gst_object_ref (decode);
+ stream->priv->convert = gst_object_ref (convert);
+ stream->priv->filter = gst_caps_ref (filter);
+ stream->priv->audio_encode = gst_object_ref (audio_encode);
+ stream->priv->sink = gst_object_ref (sink);
+
+done:
+ if (pipeline) {
+ gst_object_unref (pipeline);
+ }
+
+ if (src) {
+ gst_object_unref (src);
+ }
+
+ if (decode) {
+ gst_object_unref (decode);
+ }
+
+ if (convert) {
+ gst_object_unref (convert);
+ }
+
+ if (filter) {
+ gst_caps_unref (filter);
}
- g_assert (G_IS_SEEKABLE (stream));
+ if (audio_encode) {
+ gst_object_unref (audio_encode);
+ }
+
+ if (sink) {
+ gst_object_unref (sink);
+ }
+
return G_INPUT_STREAM (stream);
}
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