652 lines
20 KiB
C
652 lines
20 KiB
C
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/*
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audio: audio output interface
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This wraps a bit of out123 now, with a layer of resampling using syn123.
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copyright ?-2020 by the mpg123 project - free software under the terms of the LGPL 2.1
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see COPYING and AUTHORS files in distribution or http://mpg123.org
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initially written by Michael Hipp
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Pitching can work in three ways:
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- fixed (native) decoder rate, varied hardware playback rate
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- fixed hardware playback rate, any decoder rate, resampler
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- pitched NtoM decoder rate, fixed hardware
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Just because ...
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If you force an output rate, either the NtoM resampler in libmpg123 or the syn123
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resampler wrapped here adapts the data.
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TODO: If the proper resampler is configured, it should fill in unsupported output
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rates automatically ... or not? Right now, I have the rule that you don't get
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the expensive resampler unless you forced an output rate. But you do get the
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libmpg123 NtoM resampling now if it finds only a working output rate that is
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reachable via that. Well, imposing a resampler that needs more CPU time than
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the decoder is just something that should not happen without being called for.
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So I rather provide a warning to the user that the NtoM resampler has been
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triggered. Yes, inform the user.
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*/
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#include <errno.h>
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#include "mpg123app.h"
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#include "audio.h"
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#include "out123.h"
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#include "syn123.h"
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#include "common.h"
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#include "metaprint.h"
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#include "sysutil.h"
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#ifdef HAVE_SYS_WAIT_H
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#include <sys/wait.h>
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#endif
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#include "common/debug.h"
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static syn123_handle *sh = NULL;
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static struct mpg123_fmt outfmt = { .encoding=0, .rate=0, .channels=0 };
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static int outch = 0; // currently used number of output channels
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// A convoluted way to say outch*4, for semantic clarity.
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#define RESAMPLE_FRAMESIZE(ch) ((ch)*MPG123_SAMPLESIZE(MPG123_ENC_FLOAT_32))
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#define OUTPUT_FRAMESIZE(ch) ((ch)*MPG123_SAMPLESIZE(outfmt.encoding))
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// Resampler buffers, first for resampling output, then for conversion.
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// Instead of sizing them for any possible input rate, I'll try to
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// loop in the output wrapper over a fixed buffer size. The syn123 resampler
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// can tell me how many samples to feed to avoid exceeding the output buffer.
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static char *resample_buffer = NULL;
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static char* resample_outbuf = NULL;
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static size_t resample_block = 0;
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// A good buffer size:
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// 1152*48/44.1*2*4 = 10032 ... let's go 16K.
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// This should work for final output data, too.
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// We'll loop over pieces if the buffer size is not enough for upsampling.
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static size_t resample_bytes = 1<<16;
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int do_resample = 0;
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int do_resample_now = 0; // really apply resampler for current stream.
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/* Quick-shot paired table setup with remembering search in it.
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this is for storing pairs of output sampling rate and decoding
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sampling rate. */
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struct ratepair { long a; long b; };
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static long *outrates = NULL;
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static struct ratepair *unpitch = NULL;
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static int audio_capabilities(out123_handle *ao, mpg123_handle *mh);
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#define CLEAN_POINTER(p, func) if(p) func(p); p = NULL;
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void audio_cleanup(void)
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{
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CLEAN_POINTER(outrates, free)
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CLEAN_POINTER(unpitch, free)
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CLEAN_POINTER(sh, syn123_del)
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CLEAN_POINTER(resample_outbuf, free)
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CLEAN_POINTER(resample_buffer, free)
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}
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int audio_setup(out123_handle *ao, mpg123_handle *mh)
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{
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do_resample = (param.force_rate > 0 && param.resample);
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resample_block = 0;
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// Settle formats.
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if(audio_capabilities(ao, mh))
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return -1;
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// Prepare resample.
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// Resampling only for forced rate for now. In future, pitching should
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// also be handled.
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if(do_resample)
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{
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int err;
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sh = syn123_new(outfmt.rate, 1, outfmt.encoding, 0, &err);
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if(!sh)
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{
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merror("Cannot initialize syn123: %s\n", syn123_strerror(err));
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return -1;
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}
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resample_buffer = malloc(resample_bytes*10);
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resample_outbuf = malloc(resample_bytes*10);
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if(!resample_buffer || !resample_outbuf)
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return -1;
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}
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return 0;
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}
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int audio_prepare( out123_handle *ao, mpg123_handle *mh
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, long rate, int channels, int encoding )
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{
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mdebug( "audio_prepare %ld Hz / %ld Hz, %i ch, enc %s"
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, rate, outfmt.rate, channels, out123_enc_name(encoding) );
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if(do_resample && param.pitch == 0. && rate == outfmt.rate)
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{
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do_resample_now = 0;
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if(param.verbose > 1)
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fprintf(stderr, "Note: resampler disabled for native rate\n");
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} else if(do_resample)
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{
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do_resample_now = 1;
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// Smooth option could be considered once pitching is implemented with the
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// resampler.The existing state might fit the coming data if this is two
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// seamless tracks. If not, it's jut the first few samples that differ
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// significantly depending on which data went through the resampler
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// previously.
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int err = syn123_setup_resample( sh, pitch_rate(rate), outfmt.rate, channels
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, (param.resample < 2), 0 );
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if(err)
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{
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merror("failed to set up resampler: %s", syn123_strerror(err));
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return -1;
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}
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outch = channels;
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// We can store a certain ammount of frames in the resampler buffer
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// and the final output buffer after conversion.
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size_t frames = resample_bytes / (
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RESAMPLE_FRAMESIZE(channels) > OUTPUT_FRAMESIZE(channels)
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? RESAMPLE_FRAMESIZE(channels)
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: OUTPUT_FRAMESIZE(channels) );
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// Minimum amount of input samples to fill the buffer.
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resample_block = syn123_resample_fillcount(pitch_rate(rate), outfmt.rate, frames);
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if(!resample_block)
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return -1; // WTF? No comment.
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if(param.verbose > 1)
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fprintf(stderr, "Note: resampler setup: %ld Hz -> %ld Hz\n", pitch_rate(rate), outfmt.rate);
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rate = outfmt.rate;
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encoding = outfmt.encoding;
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} else if(outfmt.rate)
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rate = outfmt.rate; // That's pitching with NtoM.
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else
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{
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struct mpg123_frameinfo fi;
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static int ntom_warn = 0;
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if( !ntom_warn && !param.quiet &&
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MPG123_OK == mpg123_info(mh, &fi) && fi.rate != rate )
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{
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fprintf(stderr,
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"\nWarning: You triggered the NtoM drop-sample resampler inside libmpg123.\n"
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"Warning: You could trade CPU for quality by forcing a supported output rate.\n" );
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ntom_warn = 1;
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}
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rate = pitch_rate(rate); // That's plain hardware pitching.
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}
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if(param.verbose > 1)
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{
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const char* encname = out123_enc_name(encoding);
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fprintf( stderr // No extra line break, as this is a follow-up note.
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, "Note: Hardware output format %li Hz, %i channels, encoding %s.\n"
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, rate, channels, encname ? encname : "???" );
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}
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return out123_start(ao, rate, channels, encoding);
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}
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// Loop over blocks with the resampler, think about intflag.
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size_t audio_play(out123_handle *ao, void *buffer, size_t bytes)
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{
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if(do_resample_now)
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{
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int fs = RESAMPLE_FRAMESIZE(outch);
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size_t pcmframes = bytes/fs;
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size_t done = 0;
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while(pcmframes && !intflag)
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{
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size_t block = resample_block > pcmframes
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? pcmframes
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: resample_block;
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size_t oblock = syn123_resample( sh, (float*)resample_buffer
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, (float*)((char*)buffer+done), block );
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if(!oblock)
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break;
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size_t obytes = 0;
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if(syn123_conv( resample_outbuf, outfmt.encoding, resample_bytes
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, resample_buffer, MPG123_ENC_FLOAT_32, oblock*fs
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, &obytes, NULL, sh ))
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break;
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size_t oplay = out123_play(ao, resample_outbuf, obytes);
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if(oplay < obytes)
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{
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// Need to translate that. How many input samples got played,
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// actually? A bit of roundoff error doesn't hurt, so let's just
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// wing it. Close is enough.
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size_t iframes = (size_t)((double)oplay/obytes*block);
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while(iframes >= block)
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--iframes;
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done += iframes*fs;
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break;
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}
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pcmframes -= block;
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done += block*fs;
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}
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return done;
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}
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else
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return out123_play(ao, buffer, bytes);
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}
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mpg123_string* audio_enclist(void)
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{
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int i;
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mpg123_string *list;
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size_t enc_count = 0;
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const int *enc_codes = NULL;
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/* Only the encodings supported by libmpg123 build
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Those returned by out123_enc_list() are a superset. */
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mpg123_encodings(&enc_codes, &enc_count);
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if((list = malloc(sizeof(*list))))
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mpg123_init_string(list);
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/* Further calls to mpg123 string lib are hardened against NULL. */
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for(i=0;i<enc_count;++i)
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{
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if(i>0)
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mpg123_add_string(list, " ");
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mpg123_add_string(list, out123_enc_name(enc_codes[i]));
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}
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return list;
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}
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static void capline(mpg123_handle *mh, long rate, struct mpg123_fmt *outfmt)
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{
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int enci;
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const int *encs;
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size_t num_encs;
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mpg123_encodings(&encs, &num_encs);
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fprintf(stderr," %5ld |", outfmt ? outfmt->rate : rate);
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for(enci=0; enci<num_encs; ++enci)
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{
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int fmt = outfmt
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? (encs[enci] == outfmt->encoding ? outfmt->channels : 0)
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: mpg123_format_support(mh, rate, encs[enci]);
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switch(fmt)
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{
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case MPG123_MONO: fprintf(stderr, " M "); break;
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case MPG123_STEREO: fprintf(stderr, " S "); break;
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case MPG123_MONO|MPG123_STEREO: fprintf(stderr, " M/S "); break;
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default: fprintf(stderr, " ");
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}
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}
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fprintf(stderr, "\n");
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}
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void print_capabilities(out123_handle *ao, mpg123_handle *mh)
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{
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int r,e;
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const long *rates;
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size_t num_rates;
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const int *encs;
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size_t num_encs;
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char *name;
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char *dev;
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out123_driver_info(ao, &name, &dev);
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mpg123_rates(&rates, &num_rates);
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mpg123_encodings(&encs, &num_encs);
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fprintf(stderr,"\nAudio driver: %s\nAudio device: ", name);
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print_outstr(stderr, dev, 0, stderr_is_term);
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fprintf(stderr, "\n");
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fprintf( stderr, "%s", "Audio capabilities:\n"
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"(matrix of [S]tereo or [M]ono support for sample format and rate in Hz)\n"
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"\n"
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" rate |" );
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for(e=0;e<num_encs;e++)
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{
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const char *encname = out123_enc_name(encs[e]);
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fprintf(stderr," %4s ", encname ? encname : "???");
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}
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fprintf(stderr,"\n -------");
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for(e=0;e<num_encs;e++) fprintf(stderr,"------");
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fprintf(stderr, "\n");
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for(r=0; r<num_rates; ++r) capline(mh, rates[r], NULL);
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if(param.force_rate)
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{
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fprintf(stderr," -------");
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for(e=0;e<num_encs;e++) fprintf(stderr,"------");
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fprintf(stderr, "\n");
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if(do_resample)
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capline(mh, 0, &outfmt);
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else
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capline(mh, bpitch_rate(param.force_rate), NULL);
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}
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fprintf(stderr,"\n");
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if(do_resample)
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{
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if(param.pitch != 0.)
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fprintf( stderr, "Resampler with pitch: %g\n"
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, param.pitch );
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else
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fprintf(stderr, "Resampler configured.\n");
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fprintf( stderr, "%s\n%s\n"
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, "Decoding to f32 as intermediate if needed."
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, "Resampler output format is in the last line." );
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}
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else if(param.force_rate)
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fprintf( stderr
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, "%s rate forced. Resulting format support shown in last line.\n"
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, param.pitch != 0. ? "Pitched decoder" : "Decoder" );
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else if(param.pitch != 0.)
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fprintf( stderr, "Actual output rates adjusted by pitch value %g.\n"
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, param.pitch );
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}
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long brate(struct ratepair *table, long arate, int count, int *last)
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{
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int i = 0;
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int j;
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for(j=0; j<2; ++j)
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{
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i = i ? 0 : *last;
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for(; i<count; ++i) if(table[i].a == arate)
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{
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*last = i;
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return table[i].b;
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}
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}
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return 0;
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}
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// Return one of the given list of encodings matching the mask,
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// in order. Zero if none.
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static int match_enc(int mask, const int *enc_list, size_t enc_count)
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{
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int enc = 0;
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for(size_t i=0; i<enc_count; ++i)
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{
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if((enc_list[i] & mask) == enc_list[i])
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{
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enc = enc_list[i];
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break;
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}
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}
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return enc;
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}
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/* This uses the currently opened audio device, queries its caps.
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In case of buffered playback, this works _once_ by querying the buffer for the caps before entering the main loop. */
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static int audio_capabilities(out123_handle *ao, mpg123_handle *mh)
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{
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int force_fmt = 0;
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size_t ri;
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/* Pitching introduces a difference between decoder rate and playback rate. */
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long decode_rate;
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const long *rates;
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struct mpg123_fmt *outfmts = NULL;
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int fmtcount;
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size_t num_rates;
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if(param.pitch < -0.99)
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param.pitch = -0.99;
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long ntom_rate = do_resample ? 0 : bpitch_rate(param.force_rate);
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outfmt.rate = param.force_rate;
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outfmt.channels = 0;
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outfmt.encoding = 0;
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debug("audio_capabilities");
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mpg123_rates(&rates, &num_rates);
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mpg123_format_none(mh); /* Start with nothing. */
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if(do_resample && param.verbose > 2)
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fprintf( stderr
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, "Note: decoder always forced to %s encoding for resampler\n"
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, out123_enc_name(MPG123_ENC_FLOAT_32) );
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if(param.force_encoding != NULL)
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{
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if(!param.quiet)
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fprintf(stderr, "Note: forcing output encoding %s\n", param.force_encoding);
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force_fmt = out123_enc_byname(param.force_encoding);
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if(!force_fmt)
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{
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char *fs = NULL;
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outstr(&fs, param.force_encoding, 0, stderr_is_term);
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error1("Failed to find an encoding to match requested \"%s\"!\n"
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, PSTR(fs));
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free(fs);
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|
return -1; /* No capabilities at all... */
|
||
|
}
|
||
|
else if(param.verbose > 2)
|
||
|
fprintf(stderr, "Note: forcing encoding code 0x%x (%s)\n"
|
||
|
, (unsigned)force_fmt, out123_enc_name(force_fmt));
|
||
|
}
|
||
|
|
||
|
// A possible optimization for resampling mode is to keep existing output
|
||
|
// format support configured and don't even interrupt the output device at
|
||
|
// all. If you change pitch, you just change a number for the resampler.
|
||
|
// But currently, the idea of re-opening the output device on format
|
||
|
// changes is rather ingrained in mpg123.
|
||
|
|
||
|
if(do_resample)
|
||
|
{
|
||
|
// If really doing the extra resampling, output will always run with
|
||
|
// this setup, regardless of decoder.
|
||
|
int enc1 = out123_encodings(ao, outfmt.rate, 1);
|
||
|
int enc2 = out123_encodings(ao, outfmt.rate, 2);
|
||
|
if(force_fmt)
|
||
|
{
|
||
|
enc1 &= force_fmt;
|
||
|
enc2 &= force_fmt;
|
||
|
} else
|
||
|
{
|
||
|
long propflags = 0;
|
||
|
out123_getparam_int(ao, OUT123_PROPFLAGS, &propflags);
|
||
|
if(!(propflags & OUT123_PROP_LIVE))
|
||
|
{
|
||
|
fmtcount = out123_formats(ao, NULL, 0, 1, 2, &outfmts);
|
||
|
if(fmtcount == 1 && outfmts[0].encoding > 0)
|
||
|
{
|
||
|
const char *encname = out123_enc_name(outfmts[0].encoding);
|
||
|
if(param.verbose > 1)
|
||
|
fprintf( stderr, "Note: honouring non-live default encoding of %s\n"
|
||
|
, encname ? encname : "???" );
|
||
|
enc1 &= outfmts[0].encoding;
|
||
|
enc2 &= outfmts[0].encoding;
|
||
|
}
|
||
|
free(outfmts);
|
||
|
outfmts = NULL;
|
||
|
} else if(param.verbose > 1)
|
||
|
fprintf(stderr, "Note: negotiating the best encoding with live sink\n");
|
||
|
}
|
||
|
mdebug("enc mono=0x%x stereo=0x%x", (unsigned)enc1, (unsigned)enc2);
|
||
|
if(!enc1 && !enc2)
|
||
|
{
|
||
|
error("Output device does not support forced rate and/or encoding.");
|
||
|
return -1;
|
||
|
} else if(enc1 && enc2)
|
||
|
{
|
||
|
// Should be the normal case: Mono and Stereo with the same formats.
|
||
|
// We'll store the common subset, which should normally be all.
|
||
|
outfmt.encoding = enc1 & enc2;
|
||
|
outfmt.channels = MPG123_MONO|MPG123_STEREO;
|
||
|
if(!outfmt.encoding)
|
||
|
{
|
||
|
error("No common decodings for mono and stereo output. Too weird.");
|
||
|
return -1;
|
||
|
}
|
||
|
} else if(enc1)
|
||
|
{ // Mono only.
|
||
|
outfmt.encoding = enc1;
|
||
|
outfmt.channels = 1;
|
||
|
} else
|
||
|
{ // Stereo only.
|
||
|
outfmt.encoding = enc2;
|
||
|
outfmt.channels = 2;
|
||
|
}
|
||
|
int pref_enc[] = { MPG123_ENC_FLOAT_32
|
||
|
, MPG123_ENC_SIGNED_32, MPG123_ENC_UNSIGNED_32
|
||
|
, MPG123_ENC_SIGNED_24, MPG123_ENC_UNSIGNED_24
|
||
|
, MPG123_ENC_SIGNED_16, MPG123_ENC_UNSIGNED_16 };
|
||
|
int nenc = match_enc(outfmt.encoding, pref_enc, sizeof(pref_enc)/sizeof(int));
|
||
|
if(!nenc)
|
||
|
{
|
||
|
const int *encs;
|
||
|
size_t num_encs;
|
||
|
mpg123_encodings(&encs, &num_encs);
|
||
|
nenc = match_enc(outfmt.encoding, encs, num_encs);
|
||
|
}
|
||
|
if(nenc)
|
||
|
outfmt.encoding = nenc;
|
||
|
else
|
||
|
{
|
||
|
merror( "Found no encoding to match mask 0x%08x."
|
||
|
, (unsigned int)outfmt.encoding );
|
||
|
if(force_fmt)
|
||
|
error("Perhaps your forced output encoding is not supported.");
|
||
|
return -1;
|
||
|
}
|
||
|
const char *encname = out123_enc_name(outfmt.encoding);
|
||
|
if(param.verbose > 1)
|
||
|
for(int ch=MPG123_MONO; ch<=MPG123_STEREO; ++ch)
|
||
|
if(outfmt.channels & ch)
|
||
|
fprintf(stderr, "Note: output format %li Hz, %s, %s\n"
|
||
|
, outfmt.rate, ch==MPG123_MONO ? "mono" : "stereo"
|
||
|
, encname ? encname : "???" );
|
||
|
}
|
||
|
|
||
|
// Either enable or disable rate forcing, whith ntom_rate non-zero or not.
|
||
|
if(mpg123_param(mh, MPG123_FORCE_RATE, ntom_rate, 0) != MPG123_OK)
|
||
|
{
|
||
|
merror("Cannot force NtoM rate: %s", mpg123_strerror(mh));
|
||
|
return -1;
|
||
|
}
|
||
|
|
||
|
if(ntom_rate)
|
||
|
{
|
||
|
// Only that one rate is enforced. Nothing else needs to be checked.
|
||
|
// For pitching, ntom_rate has been adjusted. The output uses outfmt.rate.
|
||
|
// Need to tell mpg123 about the forced rate to make it work.
|
||
|
for(int ch=1; ch<=2; ++ch)
|
||
|
{
|
||
|
int fmts = out123_encodings(ao, outfmt.rate, ch);
|
||
|
if(param.verbose > 2)
|
||
|
fprintf( stderr
|
||
|
, "Note: output support for %li Hz, %s: 0x%x\n"
|
||
|
, outfmt.rate, ch==MPG123_MONO ? "mono" : "stereo", fmts );
|
||
|
if(force_fmt)
|
||
|
fmts = ((fmts & force_fmt) == force_fmt) ? force_fmt : 0;
|
||
|
mpg123_format(mh, ntom_rate, ch, fmts);
|
||
|
}
|
||
|
} else if(do_resample)
|
||
|
{
|
||
|
// Support any decoding rate with float output for the resampler and also
|
||
|
// direct decoding to confiugred output format.
|
||
|
// One twist: Disable high rates with signal that the resampler will throw
|
||
|
// away anyway. This includes pitch. 22040 Hz output rate with pitch 0.5
|
||
|
// still wants the full 44100 Hz input data, as original signal up to
|
||
|
// 22040 Hz will be heard as up to 11020 Hz. So we want pitch_rate()
|
||
|
// to be above outfmt.rate. Final resampling ratio not above 2.
|
||
|
for(ri=0; ri<num_rates; ++ri)
|
||
|
{
|
||
|
if(rates[ri] > 12000 && pitch_rate(rates[ri]) > outfmt.rate*2)
|
||
|
break;
|
||
|
int fmt = (param.pitch == 0. && rates[ri] == outfmt.rate)
|
||
|
? outfmt.encoding
|
||
|
: MPG123_ENC_FLOAT_32;
|
||
|
mpg123_format(mh, rates[ri], outfmt.channels, fmt);
|
||
|
}
|
||
|
} else
|
||
|
{
|
||
|
// Finally, the old style, direct decoding to possibly pitched output.
|
||
|
if(!outrates)
|
||
|
outrates = malloc(sizeof(*rates)*num_rates);
|
||
|
if(!unpitch)
|
||
|
unpitch = malloc(sizeof(*unpitch)*num_rates);
|
||
|
if(!outrates || !unpitch)
|
||
|
{
|
||
|
CLEAN_POINTER(outrates, free)
|
||
|
CLEAN_POINTER(unpitch, free)
|
||
|
error("DOOM");
|
||
|
return -1;
|
||
|
}
|
||
|
for(ri = 0; ri<num_rates; ri++)
|
||
|
{
|
||
|
decode_rate = rates[ri];
|
||
|
outrates[ri] = pitch_rate(decode_rate);
|
||
|
unpitch[ri].a = outrates[ri];
|
||
|
unpitch[ri].b = decode_rate;
|
||
|
}
|
||
|
/* Actually query formats possible with given rates. */
|
||
|
fmtcount = out123_formats(ao, outrates, num_rates, 1, 2, &outfmts);
|
||
|
// Remember: First one is a default format, then come my rates.
|
||
|
if(fmtcount > 0)
|
||
|
{
|
||
|
int fi;
|
||
|
int unpitch_i = 0;
|
||
|
if(param.verbose > 1 && outfmts[0].encoding > 0)
|
||
|
{
|
||
|
const char *encname = out123_enc_name(outfmts[0].encoding);
|
||
|
fprintf(stderr, "Note: default format %li Hz, %i channels, %s\n"
|
||
|
, outfmts[0].rate, outfmts[0].channels
|
||
|
, encname ? encname : "???" );
|
||
|
}
|
||
|
for(fi=1; fi<fmtcount; ++fi)
|
||
|
{
|
||
|
int fmts = outfmts[fi].encoding;
|
||
|
if(param.verbose > 2)
|
||
|
fprintf( stderr
|
||
|
, "Note: output support for %li Hz, %i channels: 0x%x\n"
|
||
|
, outfmts[fi].rate, outfmts[fi].channels, outfmts[fi].encoding );
|
||
|
if(force_fmt)
|
||
|
fmts = ((fmts & force_fmt) == force_fmt) ? force_fmt : 0;
|
||
|
decode_rate = brate(unpitch, outfmts[fi].rate, num_rates, &unpitch_i);
|
||
|
mpg123_format(mh, decode_rate, outfmts[fi].channels, fmts);
|
||
|
}
|
||
|
}
|
||
|
free(outfmts);
|
||
|
}
|
||
|
|
||
|
if(param.verbose > 1) print_capabilities(ao, mh);
|
||
|
|
||
|
return 0;
|
||
|
}
|
||
|
|
||
|
int set_pitch(mpg123_handle *fr, out123_handle *ao, double new_pitch)
|
||
|
{
|
||
|
double old_pitch = param.pitch;
|
||
|
long rate;
|
||
|
int channels, format;
|
||
|
int smode = 0;
|
||
|
|
||
|
/* Be safe, check support. */
|
||
|
if(mpg123_getformat(fr, &rate, &channels, &format) != MPG123_OK)
|
||
|
{
|
||
|
/* We might just not have a track handy. */
|
||
|
error("There is no current audio format, cannot apply pitch. This might get fixed in future.");
|
||
|
return 0;
|
||
|
}
|
||
|
|
||
|
if(outfmt.rate && !do_resample)
|
||
|
{
|
||
|
error("Runtime pitching requires either proper resampler or flexible hardware rate.");
|
||
|
return 0;
|
||
|
}
|
||
|
|
||
|
param.pitch = new_pitch;
|
||
|
|
||
|
if(channels == 1) smode = MPG123_MONO;
|
||
|
if(channels == 2) smode = MPG123_STEREO;
|
||
|
|
||
|
out123_stop(ao);
|
||
|
/* Remember: This takes param.pitch into account. */
|
||
|
audio_capabilities(ao, fr);
|
||
|
if(!do_resample && !(mpg123_format_support(fr, rate, format) & smode))
|
||
|
{
|
||
|
|
||
|
/* Note: When using --pitch command line parameter, you can go higher
|
||
|
because a lower decoder sample rate is automagically chosen.
|
||
|
Here, we'd need to switch decoder rate during track... good? */
|
||
|
error("Reached a hardware limit there with pitch!");
|
||
|
param.pitch = old_pitch;
|
||
|
audio_capabilities(ao, fr);
|
||
|
}
|
||
|
else
|
||
|
mpg123_decoder(fr, NULL);
|
||
|
return audio_prepare(ao, fr, rate, channels, format);
|
||
|
}
|
||
|
|
||
|
int set_mute(out123_handle *ao, int mutestate)
|
||
|
{
|
||
|
return out123_param( ao
|
||
|
, mutestate ? OUT123_ADD_FLAGS : OUT123_REMOVE_FLAGS
|
||
|
, OUT123_MUTE, 0, NULL );
|
||
|
}
|