I'm new to libsox programming, and I want to reduce a chennel from a stereo audio which named 'a.wav' then generate a mono audio 'b.wav' with the following code:
sox_format_t * in, * out;
sox_effects_chain_t * chain;
sox_effect_t * e;
char * args[10];
sox_init();
in = sox_open_read("E:\\a.wav", NULL, NULL, NULL);
out = sox_open_write("E:\\b.wav", &in->signal, NULL, NULL, NULL, NULL);
out->signal.channels = 1;
chain = sox_create_effects_chain(&in->encoding, &out->encoding);
e = sox_create_effect(sox_find_effect("input"));
sox_add_effect(chain, e, &in->signal, &in->signal);
e = sox_create_effect(sox_find_effect("channels"));
sox_add_effect(chain, e, &in->signal, &out->signal);
e = sox_create_effect(sox_find_effect("output"));
sox_add_effect(chain, e, &in->signal, &out->signal);
sox_flow_effects(chain, NULL, NULL);
sox_delete_effects_chain(chain);
sox_close(out);
sox_close(in);
sox_format_quit();
After running the application, the mono audio 'b.wav' was generated, but the duration of the sound was half of the a.wav.
Is there something wrong with my code?
Any reply will be appreciated!
sox_add_effect() overwrites the input signal (third parameter) to describe the properties of the signal after this processing step, so that you can pass it to the next effect. However, in your case, the modified signal information is also used by the read handler, and the contents no longer match the file being read.
You’ll need to make a copy of the signal information as returned by sox_open_read(), which you can then pass as the third parameter to the sox_add_effect() calls:
sox_signalinfo_t interm_signal = in->signal;
...
sox_add_effect(chain, e, &interm_signal, &out->signal);
That’s why I warned you to look at the newest version of example3.c in the git repository, not in the released versions.
Related
Lets say that I am reading from data stream, and that stream is sending the content of an h264 video feed. Given I read from that stream and I have some amount of data consisting of an indeterminate number of frames (NAL?). Given that i know the framerate, and size of the originating video, how would I go about converting this snippet into a mp4 that i could view? The video does not contain audio.
I want to do this using nodejs? My attempts to do so have produced nothing resembling a valid h264 file to convert into mp4. My thoughts so far were to strip any data preceding the first found start code in the data and feed that into a file and use ffmpeg (currently just testing in the command line) to convert the file to mp4.
What's the correct way to go about doing this?
ie. something like this (it's in Typescript but same thing)
//We assume here that when this while loop exist at least one full frame of data will have been read and written to disk
let stream: WriteStream = fs.createWriteStream("./test.h264")
while(someDataStream.available()) { //just an example not real code
let data: Buffer = someDataStream.readSomeData() //just an example not a real method call
let file = null;
try {
file = fs.statSync("./test.h264");
} catch (error) {
console.error(error)
}
if(!stream.writable) {
console.error("stream not writable")
} else if(file == null || file.size <= 0) {
let index = data.indexOf(0x7C)
console.log("index: " + index)
if(index > 0) {
console.log("index2: " + data.slice(index).indexOf(0x7c))
stream.write(data.slice(index))
}
} else {
stream.write(data)
}
}
To handle a data stream, you'll need to emit fragmented MP4. Like all MP4, fMP4 streams begin with a preamble containing ftyp, moov, and styp boxes. Then each frame is encoded with a moof / mdat box pair.
In order to generate a useful preamble from your H.264 bitstream, you need to locate a SPS / PPS pair of NALUs in the H264 data, to set up the avc1 box within the moov box. Those two NALUs are often immediately followed by an I-frame (a key frame). The first frame in a stream must be an I-frame, and subsequent ones can be P- or B- frames. E
It's a fairly complex task involving lots of bit-banging and buffer-shuffling (those are technical terms ;-).
I've been working on a piece of js code to extract H.264 from webm and put it into fmp4. It's not yet complete. It's backed up by another piece of code to decode the parts of the H264 stream that are needed to pack it properly into fMP4.
I wish I could write, "here are the ten lines of code you need" but those formats (fMP4 and H264) aren't simple enough to make that possible.
Idk why none of those questions doesn't actually have an easy answer. Here you go, Node.js solution, i argument just in case you need to offset the search
const soi = Buffer.from([0x00, 0x00, 0x00, 0x01]);
function findStartFrame(buffer, i = -1) {
while ((i = buffer.indexOf(soi, i + 1)) !== -1) {
if ((buffer[i + 4] & 0x1F) === 7) return i
}
return -1
}
I am working on a testing tool for nvme-cli(written in c and can run on linux).
For SSD validation purpose, i was actually looking for a custom command(For e.g. I/O command, write and then read the same and finally compare if both the data are same)
For read the ioctl() function is used as shown in the below code.
struct nvme_user_io io = {
.opcode = opcode,
.flags = 0,
.control = control,
.nblocks = nblocks,
.rsvd = 0,
.metadata = (__u64)(uintptr_t) metadata,
.addr = (__u64)(uintptr_t) data,
.slba = slba,
.dsmgmt = dsmgmt,
.reftag = reftag,
.appmask = appmask,
.apptag = apptag,
};
err = ioctl(fd, NVME_IOCTL_SUBMIT_IO, &io);
Can I to where exactly the control of execution goes in order to understand the read.
Also I want to have another command that looks like
err = ioctl(fd,NVME_IOCTL_WRITE_AND_COMPARE_IO, &io);
so that I can internally do a write, then read the same location and finally compare the both data to ensure that the disk contains only the data that I wanted to write.
Since I am new to this nvme/ioctl(), if there is any mistakes please correct me.
nvme_io() is a main command handler that accepts as a parameter the NVMe opcode that you want to send to your device. According to the standard, you have separate commands (opcodes) for read, write and compare. You could either send those commands separately, or add a vendor specific command to calculate what you need.
I'm writing a MOD player, trying to playback a sample using Allegro5 raw stream capabilities, I can't figure out the exact init parameters for the stream to play the loaded sample data from the mod file.
This is what I have:
xf::ModLoader ml;
ml.loadFromFile("C:\\Users\\bubu\\Downloads\\agress.mod");
// getSampleLength() returns # of data words
int sample_length = ml.getSampleLength(1) * 2;
const int8_t* sample_data = ml.getSampleData(1);
ALLEGRO_MIXER* mixer = al_get_default_mixer();
ALLEGRO_AUDIO_STREAM* stream = al_create_audio_stream(1, sample_length, 8363, ALLEGRO_AUDIO_DEPTH_INT8, ALLEGRO_CHANNEL_CONF_1);
al_attach_audio_stream_to_mixer(stream, mixer);
al_set_audio_stream_gain(stream, 0.7f);
al_set_audio_stream_playmode(stream, ALLEGRO_PLAYMODE_ONCE);
al_set_audio_stream_playing(stream, true);
al_set_audio_stream_fragment(stream, (void*)sample_data);
al_drain_audio_stream(stream);
First of all, freq param is hardcoded for the test (8363Hz), but, playing back at the specified frequency I don't get what I expect, and al_drain_audio_stream() gets stuck forever playing garbage in a loop...
Any help would be appreciated.
At the very least, you need to be calling al_get_audio_stream_fragment before you call al_set_audio_stream_fragment. Typically you'd feed these streams in a while loop, while responding to the ALLEGRO_EVENT_AUDIO_STREAM_FRAGMENT event. See the ex_saw example in the Allegro's source for some sample code: https://github.com/liballeg/allegro5/blob/master/examples/ex_saw.c
When you create a WaveOut object, and initialize it with WaveOffsetStream, the PlaybackStopped event is not raised at the end of the playback. Code:
WaveOut myWaveOut = new WaveOut();
myWaveOut.PlaybackStopped += OnPlaybackStopped;
WaveOffsetStream OffsetStream = new WaveOffsetStream(MyOtherStream);
myWaveOut.Init(OffsetStream);
myWaveOut.Play();
WaveOutOffsetStream always returns the requested number of bytes from Read so it is a never-ending stream. You'd have to use something different, or detect when the Position went past the point you wanted to play to
I've written a simple Groovy script (below) to set the values of four of the ID3v1 and ID3v2 tag fields in mp3 files using the JAudioTagger library. The script successfully makes the changes but it also deletes the first 5 to 10 seconds of some of the files, other files are unaffected. It's not a big problem, but if anyone knows a simple fix, I would be grateful. All the files are from the same source, all have v1 and v2 tags, I can find no obvious difference in the source files to explain it.
import org.jaudiotagger.*
java.util.logging.Logger.getLogger("org.jaudiotagger").setLevel(java.util.logging.Level.OFF)
Integer trackNum = 0
Integer totalFiles = 0
Integer invalidFiles = 0
validMP3File = true
def dir = new File(/D:\Users\Jeremy\Music\Speech Radio\Unlistened\Z Temp Files to MP3 Tagged/)
dir.eachFile({curFile ->
totalFiles ++
try {
mp3File = org.jaudiotagger.audio.AudioFileIO.read(curFile)
} catch (org.jaudiotagger.audio.exceptions.CannotReadException e) {
validMP3File = false
invalidFiles ++
}
// Get the file name excluding the extension
baseFilename = org.jaudiotagger.audio.AudioFile.getBaseFilename(curFile)
// Check that it is an MP3 file
if (validMP3File) {
if (mp3File.getAudioHeader().getEncodingType() != 'mp3') {
validMP3File = false
invalidFiles ++
}
}
if (validMP3File) {
trackNum ++
if (mp3File.hasID3v1Tag()) {
curTagv1 = mp3File.getID3v1Tag()
} else {
curTagv1 = new org.jaudiotagger.tag.id3.ID3v1Tag()
}
if (mp3File.hasID3v2Tag()) {
curTagv2 = mp3File.getID3v2TagAsv24()
} else {
curTagv2 = new org.jaudiotagger.tag.id3.ID3v23Tag()
}
curTagv1.setField(org.jaudiotagger.tag.FieldKey.TITLE, baseFilename)
curTagv2.setField(org.jaudiotagger.tag.FieldKey.TITLE, baseFilename)
curTagv1.setField(org.jaudiotagger.tag.FieldKey.ARTIST, "BBC Radio")
curTagv2.setField(org.jaudiotagger.tag.FieldKey.ARTIST, "BBC Radio")
curTagv1.setField(org.jaudiotagger.tag.FieldKey.ALBUM, "BBC Radio - 20130205")
curTagv2.setField(org.jaudiotagger.tag.FieldKey.ALBUM, "BBC Radio - 20130205")
curTagv1.setField(org.jaudiotagger.tag.FieldKey.TRACK, trackNum.toString())
curTagv2.setField(org.jaudiotagger.tag.FieldKey.TRACK, trackNum.toString())
mp3File.setID3v1Tag(curTagv1)
mp3File.setID3v2Tag(curTagv2)
mp3File.save()
}
})
println """$trackNum tracks created from $totalFiles files with $invalidFiles invalid files"""
I'm still investigating and it appears that there is no problem with JAudioTagger. Before setting the tags, I use Total Recorder to reduce the quality of the download from 128kbps, 44,100Hz to 56kbps, 22,050Hz. This reduces the file size to less than half and the quality is fine for speech radio.
If I run my script on the original files, none of the audio track is deleted. The deletion of the first part of the audio track only occurs with the files that have been processed by Total Recorder.
Looking at the JAudioTagger logging for these files, there does appear to be a problem with the header:
Checking further because the ID3 Tag ends at 0x23f9 but the mp3 audio doesnt start until 0x7a77
Confirmed audio starts at 0x7a77 whether searching from start or from end of ID3 tag
This check is not performed for files that have not been processed by Total Recorder.
The log of the header read operation also shows (for a 27 minute track):
trackLength:06:52
It looks as though I shall have to find a new MP3 file editor!
Instead of
mp3File.save()
could you try:
mp3File.commit()
No idea if it will help, but that seems to be the documented method?