mirror of
https://code.videolan.org/videolan/vlc
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19ea8feb6d
waveout, directx, qnx, beos, win32, macosx, and the AltiVec modules.
* ALL: removed mention of AC3 in favour of A52.
* ./configure.in, ./Makefile*: modules can now be built deeper than 1
directory. As a consequence, the build is even slower (but I'm fixing
this) and make clean doesn't work anymore.
125 lines
5.2 KiB
Plaintext
125 lines
5.2 KiB
Plaintext
From - Mon Nov 5 09:19:09 2001
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for <mad-dev@lists.mars.org>; Mon, 22 Oct 2001 01:31:30 -0700
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From: Rob Leslie <rob@mars.org>
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To: mad-dev@lists.mars.org
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Subject: Re: [mad-dev] Some information about programming with MAD (in synchronous mode)
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In-reply-to: Your message of "21 Oct 2001 16:13:19 EDT."
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Date: Mon, 22 Oct 2001 01:31:30 -0700
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X-UIDL: 1879-1001307689
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Joe Drew wrote some good info on the MAD high-level API that I hope will be
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helpful to others.
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By way of clarification, MAD also has a low-level API which does not use
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callbacks. You can control the entire decoding process yourself more or less
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as follows:
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/* load buffer with your MPEG audio data */
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mad_stream_buffer(&stream, buffer, buflen);
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while (1) {
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mad_frame_decode(&frame, &stream);
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mad_synth_frame(&synth, &frame);
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/* output PCM samples in synth.pcm */
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}
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This is vastly simplified, but it shows the general idea. mad_frame_decode()
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decodes the next frame's header and subband samples. mad_synth_frame() takes
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those subband samples and synthesizes PCM samples.
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It is also possible to call mad_header_decode() before mad_frame_decode().
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This just gives you the frame's header info, in case that's all you want, or
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perhaps to help you decide whether you want to decode the rest of the frame.
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As Joe mentions, each of the stream, frame, and synth structs needs to be
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initialized and "finished" before and after use:
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struct mad_stream stream;
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struct mad_frame frame;
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struct mad_synth synth;
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mad_stream_init(&stream);
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mad_frame_init(&frame);
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mad_synth_init(&synth);
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/* ... */
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mad_synth_finish(&synth);
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mad_frame_finish(&frame);
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mad_stream_finish(&stream);
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You can work with just a struct mad_header instead of a struct mad_frame if
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you only want to decode frame headers.
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Joe writes:
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> MAD always outputs 32-bit (well, mad_fixed_t) little-endian data. Take
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> this into account when outputting samples to the sound card.
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This isn't quite right: the mad_fixed_t type is not necessarily little-endian.
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It's the same endianness as the native integer types. Also, it's only
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guaranteed to be *at least* 32 bits wide.
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The fixed-point sample format is important to understand, and I recommend
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reading the comments in libmad/fixed.h. The thing to remember when converting
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MAD's fixed-point integer samples to 16-bit PCM (or whatever) is that MAD
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encodes samples as numbers in the full-scale range [-1.0, +1.0) where the
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binary point is placed 28 (MAD_F_FRACBITS) bits to the left of the integer.
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However, you need to be prepared to handle clipping as some numbers may be
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less than -1.0 (-MAD_F_ONE) or greater than or equal to +1.0 (MAD_F_ONE, aka
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1 << MAD_F_FRACBITS).
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> Information on the other (error, filter, message) functions would be
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> appreciated, though I think in knowing this information anyone should be
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> able to puzzle it out.
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In the high-level API, the error callback function is called whenever a
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decoding error occurs. The error number is in stream->error.
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The filter callback function is called after decoding a frame, but before
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synthesis. Here it is possible to modify the frame's subband samples, for
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example to perform a uniform attenuation/amplification, or to do other special
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processing in the frequency domain.
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The message callback function is only used with MAD_DECODER_MODE_ASYNC, and is
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called whenever the parent process sends a message via mad_decoder_message().
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This callback can generate a reply by overwriting the message buffer that is
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passed to it. (The size of the reply must be the same or smaller than the
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message.)
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Cheers,
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-rob
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