[gst-devel] audioresample
Jesu Anuroop Suresh
jesuas at gmail.com
Mon Jan 17 07:17:32 CET 2011
Hi Cai,
Thanks for your response.
Below is the code which works and converts any input stream into 22KHz
S16LE.
static gboolean bus_call(GstBus *bus,GstMessage *msg,gpointer data)
{
GMainLoop *loop = (GMainLoop*)data;
switch(GST_MESSAGE_TYPE(msg))
{
case GST_MESSAGE_EOS:
g_print("End of stream\n");
g_main_loop_quit(loop);
break;
case GST_MESSAGE_ERROR:
{
gchar *debug;
GError *error;
gst_message_parse_error(msg,&error,&debug);
g_free(debug);
g_print("Error: %s\n",error->message);
g_error_free(error);
g_main_loop_quit(loop);
break;
}
case GST_STATE_CHANGE_READY_TO_NULL:
default:
//g_print("Unkown message 0x%x\n",GST_MESSAGE_TYPE(msg));
break;
}
return TRUE;
}
static void
on_pad_added (GstElement *element,
GstPad *pad,
gpointer data)
{
GstPad *sinkpad;
GstElement *decoder = (GstElement *) data;
/* We can now link this pad with the vorbis-decoder sink pad */
g_print ("Dynamic pad created, linking demuxer/decoder\n");
sinkpad = gst_element_get_static_pad (decoder, "sink");
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
}
int main(int argc,char *argv[])
{
GMainLoop *playerloop;
GstBus *playerbus;
GstElement *pipeline,*source, *demuxer, *decoder, *conv, *sink,
*resample,*resmux;
GstCaps *caps;
playerloop = g_main_loop_new(NULL,FALSE);
gst_init(&argc,&argv);
/* Create gstreamer elements */
pipeline = gst_pipeline_new ("audio-player");
source = gst_element_factory_make ("filesrc", "file-source");
sink = gst_element_factory_make ("alsasink", "audio-output");
resample = gst_element_factory_make ("audioresample", "audio-resample");
conv = gst_element_factory_make ("audioconvert", "converter1");
resmux = gst_element_factory_make ("capsfilter", "filter");
caps = gst_caps_new_simple ("audio/x-raw-int",
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, 22050,
"channels",G_TYPE_INT, 2, NULL
);
if (!pipeline || !source || !sink ||
!resample || !caps || !conv || !resmux )
{
g_print ("NO MEM Exiting.\n");
return 1;
}
/* we set the input filename to the source element */
g_object_set (G_OBJECT (source), "location", argv[1], NULL);
demuxer = gst_element_factory_make ("id3demux", "id3-demuxer");
decoder = gst_element_factory_make ("mad", "mp3-decoder");
if (!demuxer || !decoder || !conv)
{
g_print ("NO MEM Exiting.\n");
return 1;
}
g_object_set (G_OBJECT (resmux), "caps", caps, NULL);
gst_caps_unref (caps);
/* file-source -> demuxer -> decoder -> alsa-output */
gst_bin_add_many (GST_BIN (pipeline),
source, demuxer, decoder, conv, resample, resmux, sink,
NULL);
gst_element_link (source, demuxer);
gst_element_link_many (decoder, conv, resample,resmux,NULL);
if ( !gst_element_link_filtered(resmux,sink,caps) ){
g_printerr("Failed to link elements resample and alsa-sink");
}
g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added),
decoder);
playerbus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
gst_bus_add_watch(playerbus,bus_call,playerloop);
gst_object_unref(playerbus);
gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING);
g_main_loop_run(playerloop);
gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_NULL);
gst_object_unref(GST_OBJECT(pipeline));
g_print("Exit\n");
return 0;
}
With Warm Regards
Jesu Anuroop Suresh
"Any intelligent fool can make things bigger, more complex, and more
violent. It takes a touch of genius -- and a lot of courage -- to move in
the opposite direction."
"Anyone who has never made a mistake has never tried anything new."
On Mon, Jan 17, 2011 at 7:54 AM, Cai Yuanqing [via GStreamer-devel] <
ml-node+3220635-1402660758-203210 at n4.nabble.com<ml-node%2B3220635-1402660758-203210 at n4.nabble.com>
> wrote:
> Hi,
>
> On 01/13/2011 01:39 PM, Anuroop Jesu wrote:
>
> > Hi ,
> >
> > I Tried the suggestion provided for the by Cai. Thanks for the code Cai.
> >
> > It works something like this It only allows to playback the 22KHz
> > S16LE audio.
> >
> > What I was trying is to convert the any input format into a 22KHz
> > S16LE so I can mux it with other stream of the same property and mux
> > multiple streams using alsasink plug:dmix.
> I see what you mean :-)
> I tried pipeline like this:
> gst-launch-0.10 filesrc location=yellow.mp3 ! id3demux ! mad !
> audioconvert ! audioresample !
> 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true'
>
> ! alsasink
>
> It works well to first decode any type of mp3 files into PCM,and then
> re-sample them into
> 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true.
>
> So you can playback this stream ,or you can replace 'alsasink' to other
> elements:
>
> $ file yellow.mp3
> yellow.mp3: Audio file with ID3 version 2.3.0, contains: MPEG ADTS,
> layer III, v1, 128 kbps, 44.1 kHz, JntStereo
>
> $ gst-launch-0.10 filesrc location=yellow.mp3 ! id3demux ! mad !
> audioconvert ! audioresample !
> 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true'
>
> ! lame ! filesink location=haha.mp3
>
> $ file haha.mp3
> haha.mp3: MPEG ADTS, layer III, v2, 128 kbps, 22.05 kHz, JntStereo
>
>
> The pipeline above turn your stream to encode int a mp3 file with
> property as
> 'audio/x-raw-int,width=16,depth=16,rate=22050,channels=2,endianness=1234,signed=true'.
>
>
> Just add elements behind audioresample and caps in the C code.
>
> Hope it helps :-)
>
> Thanks.
>
>
>
>
> >
> > With Warm Regards
> > Jesu Anuroop Suresh
> >
> > "Any intelligent fool can make things bigger, more complex, and more
> > violent. It takes a touch of genius -- and a lot of courage -- to move
> > in the opposite direction."
> > "Anyone who has never made a mistake has never tried anything new."
> >
> >
> >
> >
> >
> >
> > On Thu, Jan 13, 2011 at 9:51 AM, Jesu Anuroop Suresh <[hidden email]<http://user/SendEmail.jtp?type=node&node=3220635&i=0>
> > <mailto:[hidden email]<http://user/SendEmail.jtp?type=node&node=3220635&i=1>>>
> wrote:
> >
> > Hi Cai,
> >
> >
> > Thanks for you response, I Will try out your suggestion of using
> > the filtered link.
> >
> > Sorry there was some typoerror in my code what I shared.
> >
> > I did initialized the 'resmux' as capasity filter and used the
> > conv not conv1.
> >
> > The cocde works for me for mp3 playback in its original settings.
> >
> >
> > resample = gst_element_factory_make ("audioresample",
> > "audio-resample");
> > conv = gst_element_factory_make ("audioconvert",
> > "converter1");
> > resmux = gst_element_factory_make ("capsfilter", "filter");
> >
> > caps = gst_caps_new_simple ("audio/x-raw-int",
> > "width", G_TYPE_INT, 16,
> > "depth", G_TYPE_INT, 16,
> > "rate", G_TYPE_INT, 22050,
> > "channels",G_TYPE_INT, 2, NULL
> > );
> >
> > if (!musicPlayer.playPipeline || !source || !sink ||
> > !resample || !resmux || !caps || !conv)
> > {
> > g_print ("NO MEM Exiting.\n");
> > return 1;
> > }
> >
> > /* we set the input filename to the source element */
> > g_object_set (G_OBJECT (source), "location", filePath, NULL);
>
> >
> > demuxer = gst_element_factory_make ("id3demux",
> > "id3-demuxer");
> > decoder = gst_element_factory_make ("mad", "mp3-decoder");
> >
> > if (!demuxer || !decoder || !conv)
> > {
> > g_print ("NO MEM Exiting.\n");
> > return 1;
> > }
> >
> > With Warm Regards
> > Jesu Anuroop Suresh
> >
> > "Any intelligent fool can make things bigger, more complex, and
> > more violent. It takes a touch of genius -- and a lot of courage
> > -- to move in the opposite direction."
> > "Anyone who has never made a mistake has never tried anything new."
> >
> > On Thu, Jan 13, 2011 at 7:16 AM, Cai Yuanqing [via
> > GStreamer-devel] <[hidden email]
> > <http://user/SendEmail.jtp?type=node&node=3215225&i=0>> wrote:
> >
> > Hi Suresh:
> > Your application have a little problem. :-)
> >
> >
> > On 01/12/2011 08:41 PM, Jesu Anuroop Suresh wrote:
> >
> > > Hi Sean,
> > >
> > > Yes, what I was trying is to resample the decoded mp3 data
> > to the
> > > fixed (22KHZ S16LE) formate,
> > >
> > > no matter what is the input rate using a C application.
> > >
> > > Thanks for your response.
> > >
> > > Here is the piece of the code for the same but it does not
> > work with
> > > audioresample with the caps filter 'resmux'. This code does
> > work
> > > without the caps filter 'resmux'.
> > >
> > > GstElement *source, *demuxer, *decoder, *conv, *sink,
> > > *resample, *resmux;
> > > GstCaps *caps;
> > >
> > > gst_init(NULL, NULL);
> > >
> > > /* Create gstreamer elements */
> > > musicPlayer.playPipeline = gst_pipeline_new
> > ("audio-player");
> > > source = gst_element_factory_make ("filesrc",
> > "file-source");
> > > sink = gst_element_factory_make ("alsasink",
> > "audio-output");
> > > resample = gst_element_factory_make ("audioresample",
> > > "audio-resample");
> > > conv = gst_element_factory_make ("audioconvert",
> > > "converter1");
> > >
> > > caps = gst_caps_new_simple ("audio/x-raw-int",
> > > "width", G_TYPE_INT, 16,
> > > "depth", G_TYPE_INT, 16,
> > > "rate", G_TYPE_INT,
> > 22050,
> > > "channels",G_TYPE_INT,
> > 2, NULL
> > > );
> > >
> > > if (!musicPlayer.playPipeline || !source || !sink ||
> > > !resample || !resmux || !caps || !conv)
> > > {
> > > g_print ("NO MEM Exiting.\n");
> > > return 1;
> > > }
> > resmux is not initialized yet,here maybe some random
> > value,you'd better
> > remove it from check list.
> >
> > >
> > > /* we set the input filename to the source element */
> > > g_object_set (G_OBJECT (source), "location",
> > filePath, NULL);
> > >
> > > demuxer = gst_element_factory_make ("id3demux",
> > "id3-demuxer");
> > > decoder = gst_element_factory_make ("mad",
> > "mp3-decoder");
> > >
> > > if (!demuxer || !decoder || !conv1)
> > conv1 ? dose it should be conv?
> >
> > > {
> > > g_print ("NO MEM Exiting.\n");
> > > return 1;
> > > }
> > >
> > > g_object_set (G_OBJECT (resmux), "caps", caps, NULL);
> > > gst_caps_unref (caps);
> > >
> > as I said before,resmux haven't initialized ,that's not quite
> > right.
> > and I suggest you to remove these two lines.
> >
> > > /* file-source -> demuxer -> decoder ->
> > alsa-output */
> > > gst_bin_add_many (GST_BIN (musicPlayer.playPipeline),
> > > source, demuxer, decoder, conv,
> > resample,
> > > resmux,sink, NULL);
> > >
> > > gst_element_link (source, demuxer);
> > > gst_element_link_many (decoder, conv,
> > resample,resmux,sink, NULL);
> > You can use gst_element_link_filtered to link resample and
> > sink with
> > caps instead of this way.
> > something like:
> > gst_element_link (source, demuxer);
> > gst_element_link_many (decoder, conv, resample, NULL);
> > if ( !gst_element_link_filtered(resample,sink,caps) ){
> > g_printerr("Failed to link elements resample and
> > alsa-sink");
> > }
> >
> >
> > > g_signal_connect (demuxer, "pad-added", G_CALLBACK
> > > (on_pad_added), decoder);
> > >
> > > GstBus *bus =
> > > gst_pipeline_get_bus(GST_PIPELINE(musicPlayer.playPipeline));
> > > gst_bus_add_watch(bus, bus_call, NULL);
> > > gst_object_unref(bus);
> > >
> > >
> > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline),
> > > GST_STATE_PLAYING);
> > >
> > > musicPlayer.playLoop = g_main_loop_new(NULL, FALSE);
> > >
> > > g_main_loop_run(musicPlayer.playLoop);
> > >
> > >
> > gst_element_set_state(GST_ELEMENT(musicPlayer.playPipeline),
> > > GST_STATE_NULL);
> > > gst_object_unref(GST_OBJECT(musicPlayer.playPipeline));
>
> > >
> > >
> > >
> > >
> > > With Warm Regards
> > > Jesu Anuroop Suresh
> > >
> > > "Any intelligent fool can make things bigger, more complex,
> > and more
> > > violent. It takes a touch of genius -- and a lot of courage
> > -- to move
> > > in the opposite direction."
> > > "Anyone who has never made a mistake has never tried
> > anything new."
> > >
> > >
> > I attached my modified source code ,you can try it.
> > Hope it helps.
> >
> > Thanks.
> >
> >
> > --
> > B.R
> >
> > Cai Yuanqing
> >
> >
> >
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> > Learn about various malware tactics and how to avoid them. Understand
>
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> --
> B.R
>
> Cai Yuanqing
>
>
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