Re: [LAD] need help mixing on the fly
On Sun, 19 Feb 2017, David Griffith wrote: I have created a test game to be loaded by Frotz for testing audio development. Oops. I forgot to add an URL to the test game: http://661.org/soundtest2.blb -- David Griffith d...@661.org A: Because it fouls the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail? ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev
Re: [LAD] need help mixing on the fly
On Sat, 18 Feb 2017, Fons Adriaensen wrote: On Sat, Feb 18, 2017 at 01:44:32AM +, David Griffith wrote: For a couple years I've been trying, without success, to reliably mix two audio streams on the fly in C. I'm using libao, libmodplug, libsamplerate, libsndfile, and libvorbis. The program is Frotz (https://github.com/DavidGriffith/frotz), a Z-machine emulator for playing old Infocom games as well as newer ones. Could I please get some help with this? Mixing audio from two audio streams just requires adding the streams sample by sample. I'm pretty sure that the real problem here is not mixing but something else. But with the limited info you provide we can only guess. Where are the streams coming from ? What are the formats ? Where is the result supposed to go ? etc. etc. etc. The audio comes from AIFF, OGGV, and MOD files which are embedded in an IFF container file which also contains game data. Using libmodplug, libsamplerate, libsndfile, and libvorbis; these audio chunks are turned into stereo streams of floats. There are only two streams. If two streams are active, mixing is done. The resulting single stream of floats is converted to pcm16 and fed into libao for output. The two streams are either "music" or "bleep". The music stream can be fed with audio data from an OGGV chunk or MOD chunk, but not both. The bleep stream is fed from an AIFF chunk. If a new stream of the same type is started, the new one immediately takes over the old one. This works. If both types are played at once, they're supposed to be mixed. That process doesn't go right and I don't know where or why, though I suspect trouble with threads, mutexes, and/or semaphores. The problem manifests in distorted sound and usually a segfault. More detail... A mixer thread is spawned when Frotz starts up. It waits for one or both float buffers to fill whereupon it mixes their contents, converts the result to pcm16 and calls libao to play it. playaiff() or playmusic() are spawned by the main thread as separate threads to read audio data from the container IFF and fill up the float buffers for the mixer to read. The defective code is in https://github.com/DavidGriffith/frotz/blob/ao-curses/src/curses/ux_audio.c I have created a test game to be loaded by Frotz for testing audio development. It's at http://661.org/soundtest2.blb. The game is a text adventure that simulates a single room which contains a Commodore 64, a Commodore Amiga, and a small box with buttons on it. Turning on the C64 causes an OGGV file to play. The Amiga plays a MOD file. The buttons on the small box cause AIFFs to play. To tickle the bug, turn on one of the computers and then press a button on the box. I'm not interested in using SDL or Pulseaudio for this project. -- David Griffith d...@661.org A: Because it fouls the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail? ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev
[LAD] need help mixing on the fly
For a couple years I've been trying, without success, to reliably mix two audio streams on the fly in C. I'm using libao, libmodplug, libsamplerate, libsndfile, and libvorbis. The program is Frotz (https://github.com/DavidGriffith/frotz), a Z-machine emulator for playing old Infocom games as well as newer ones. Could I please get some help with this? -- David Griffith d...@661.org ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev
Re: [LAD] mixing while using libao and libsndfile
On Tue, 17 May 2016, Andrea Del Signore wrote: On Tue, May 17, 2016 at 12:25 AM, David Griffith <d...@661.org> wrote: On Mon, 16 May 2016, Andrea Del Signore wrote: > I'm not simply trying to mix two files. My main project is a > game engine in which two sounds are allowed at any one time. > For instance, there can be constant background music punctuated > by sound effects. I can't get these to mix correctly. Hi, in that case you can just skip the right number of frames before starting playing sounds. I modified my code to take the start time for each file and schedule the play time with frame accuracy. http://pastebin.com/0PMyfPvK If you want your timing to be sample accurate the algorithm is a bit more complex. That won't work. Your code schedules things ahead of time before anything else happens. I need to be able to fire off sound effects the instant the player does something to cause them. I can't know in advance. -- David Griffith d...@661.org___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev
Re: [LAD] mixing while using libao and libsndfile
On May 16, 2016 3:25:48 PM PDT, David Griffith <d...@661.org> wrote: > Earlier you set up filebuffer like this: > >buflen = BUFFSIZE * sf_info[0].channels; >filebuffer = malloc(buflen * sizeof(float)); > >The size of filebuffer is BUFFSIZE float-sized frames. Therefore when >you >specify BUFFSIZE as the number of floats to read, they all fit in >filebuffer. Sorry. I meant BUFFSIZE as the number of /frames/. -- David Griffith d...@661.org ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev
Re: [LAD] mixing while using libao and libsndfile
On Mon, 16 May 2016, Andrea Del Signore wrote: > I've been knocking my head against a wall for more than a year trying to > figure out how to correctly mix two streams of audio while using > libsndfile for input and libao for output. My main requirement is that > I cannot assume anything about the output drivers -- that is, I cannot > depend on the output driver (ALSA, OSS, Sun, etc) being able to do the > mixing for me. Many of my target platforms lack any sort of mixing > services. I need to do this myself. I tried starting a mixer/player > thread that would work in a producer/consumer relationship with one or > two audio file decoder threads. I can play one sound at a time just > fine. When I try to do both, I get distortion followed by a segfault. Hi, not sure if I understood correctly: do you just want to mix N files? Like you I'm learning libsndfile and libao so this is my attempt to mix some audio files: http://pastebin.com/dm7z8b3Z HTH, Andrea P.S. Can someone explain line 88 (I already read the sndfile FAQ)? I'm not simply trying to mix two files. My main project is a game engine in which two sounds are allowed at any one time. For instance, there can be constant background music punctuated by sound effects. I can't get these to mix correctly. Regarding your line 88, I had trouble with this too: sf_count_t item_read = sf_read_float (sndfile[i], filebuffer, BUFFSIZE); // WHY BUFFSIZE? Shouldn't be BUFFSIZE * channels? The third parameter is for the number of items or frames. A frame is made up of one sample per channel. Earlier you set up filebuffer like this: buflen = BUFFSIZE * sf_info[0].channels; filebuffer = malloc(buflen * sizeof(float)); The size of filebuffer is BUFFSIZE float-sized frames. Therefore when you specify BUFFSIZE as the number of floats to read, they all fit in filebuffer. -- David Griffith d...@661.org___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev
[LAD] mixing while using libao and libsndfile
I've been knocking my head against a wall for more than a year trying to
figure out how to correctly mix two streams of audio while using
libsndfile for input and libao for output. My main requirement is that I
cannot assume anything about the output drivers -- that is, I cannot
depend on the output driver (ALSA, OSS, Sun, etc) being able to do the
mixing for me. Many of my target platforms lack any sort of mixing
services. I need to do this myself. I tried starting a mixer/player
thread that would work in a producer/consumer relationship with one or two
audio file decoder threads. I can play one sound at a time just fine.
When I try to do both, I get distortion followed by a segfault.
So, I'm back to a demo program. What must I do to this program to cause
it to start playing one audio file, then play another N seconds later?
David Griffith
d...@661.org
===begin code===
/*
* gcc -o mixer mixer.c -lao -lsndfile
*
*/
#include
#include
#include
#include
#include
#include
#define BUFFSIZE 512
int playfile(FILE *);
int main(int argc, char *argv[])
{
FILE *fp1, *fp2;
if (argc < 2) {
printf("usage: %s .ogg .aiff\n", argv[0]);
exit(1);
}
fp1 = fopen(argv[1], "rb");
if (fp1 == NULL) {
printf("Cannot open %s.\n", argv[1]);
exit(2);
}
fp2 = fopen(argv[1], "rb");
if (fp2 == NULL) {
printf("Cannot open %s.\n", argv[1]);
exit(3);
}
ao_initialize();
playfile(fp1);
playfile(fp2);
ao_shutdown();
return 0;
}
int playfile(FILE *fp)
{
int default_driver;
ao_device *device;
ao_sample_format format;
SNDFILE *sndfile;
SF_INFO sf_info;
short *shortbuffer;
int64_t toread;
int64_t frames_read;
int64_t count;
sndfile = sf_open_fd(fileno(fp), SFM_READ, _info, 1);
memset(, 0, sizeof(ao_sample_format));
shortbuffer = malloc(BUFFSIZE * sf_info.channels * sizeof(short));
frames_read = 0;
toread = sf_info.frames * sf_info.channels;
count = 0;
default_driver = ao_default_driver_id();
memset(, 0, sizeof(ao_sample_format));
format.byte_format = AO_FMT_NATIVE;
format.bits = 16;
format.channels = sf_info.channels;
format.rate = sf_info.samplerate;
device = ao_open_live(default_driver, , NULL);
if (device == NULL) {
printf("Error opening sound device.\n");
exit(4);
}
while (count < toread) {
frames_read = sf_read_short(sndfile, shortbuffer, BUFFSIZE);
count += frames_read;
ao_play(device, (char *)shortbuffer, frames_read *
sizeof(short));
}
ao_close(device);
sf_close(sndfile);
}
===end code===
--
David Griffith
d...@661.org
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