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a/src/alsadirect.cpp |
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b/src/alsadirect.cpp |
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*/
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*/
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#include "config.h"
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#include "config.h"
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#include <string.h>
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#include <string.h>
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#include <sys/types.h>
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#include <sys/types.h>
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#include <math.h>
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#include <iostream>
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#include <iostream>
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#include <queue>
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#include <queue>
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#include <alsa/asoundlib.h>
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#include <alsa/asoundlib.h>
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snd_pcm_prepare(pcm);
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snd_pcm_prepare(pcm);
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}
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}
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} else {
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} else {
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qinit = true;
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qinit = true;
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}
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}
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delete tsk;
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}
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}
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}
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}
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static bool alsa_init(const string& dev, AudioMessage *tsk)
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static bool alsa_init(const string& dev, AudioMessage *tsk)
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{
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{
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buf[i] = 1.0;
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buf[i] = 1.0;
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sum += buf[i];
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sum += buf[i];
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}
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}
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}
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}
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float operator()(float ns) {
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float operator()(float ns) {
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#if 0
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if (old == 0.0) {
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old = ns;
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} else {
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old = (old + ns) / 2;
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}
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return old;
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#endif
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#if 1
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buf[idx++] = ns;
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buf[idx++] = ns;
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sum += ns;
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sum += ns;
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if (idx == FNS)
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if (idx == FNS)
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idx = 0;
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idx = 0;
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sum -= buf[idx];
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sum -= buf[idx];
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return sum/FNS;
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return sum/FNS;
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#endif
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}
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}
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float old;
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float old;
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float buf[FNS];
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float buf[FNS];
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float sum;
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float sum;
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int idx;
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int idx;
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SRC_STATE *src_state = 0;
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SRC_STATE *src_state = 0;
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SRC_DATA src_data;
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SRC_DATA src_data;
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memset(&src_data, 0, sizeof(src_data));
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memset(&src_data, 0, sizeof(src_data));
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alsaqueue.start(1, alsawriter, 0);
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alsaqueue.start(1, alsawriter, 0);
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// Integral term. We do not use it at the moment
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// double it = 0;
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while (true) {
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while (true) {
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AudioMessage *tsk = 0;
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AudioMessage *tsk = 0;
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size_t qsz;
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size_t qsz;
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if (!queue->take(&tsk, &qsz)) {
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if (!queue->take(&tsk, &qsz)) {
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bufframes = tsk->m_bytes / (tsk->m_chans * (tsk->m_bits/8));
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bufframes = tsk->m_bytes / (tsk->m_chans * (tsk->m_bits/8));
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}
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}
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// Computing the samplerate conversion factor. We want to keep
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// Computing the samplerate conversion factor. We want to keep
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// the queue at its target size to control the delay. The
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// the queue at its target size to control the delay. The
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// present hack sort of works but has a tendancy to keep
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// present hack sort of works but could probably benefit from
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// oscillating (with a very small amplitude). It should be
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// a more scientific approach
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// replaced by a proper filter
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// Qsize in songcast buffers. This is the variable to control
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float qs;
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double qs;
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if (qinit) {
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if (qinit) {
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qs = alsaqueue.qsize() + alsadelay() / bufframes;
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qs = alsaqueue.qsize() + alsadelay() / bufframes;
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// Error term
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float t = ((qstarg - qs) / qstarg);
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double et = ((qstarg - qs) / qstarg);
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float adj = t * t;
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if (qs < qstarg) {
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// Integral. Not used, made it worse each time I tried
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// it += et;
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// Error correction coef
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double ce = 0.1;
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// Integral coef
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//double ci = 0.0001;
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// Compute command
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double adj = ce * et /* + ci * it*/;
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// Also tried a quadratic correction, worse.
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// double adj = et * ((et < 0) ? -et : et);
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// Computed ratio
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samplerate_ratio = 1.0 + adj;
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samplerate_ratio = 1.0 + adj;
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if (samplerate_ratio > 1.1)
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samplerate_ratio = 1.1;
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// Limit extension
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} else {
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samplerate_ratio = 1.0 - adj;
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if (samplerate_ratio < 0.9)
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if (samplerate_ratio < 0.9)
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samplerate_ratio = 0.9;
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samplerate_ratio = 0.9;
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}
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if (samplerate_ratio > 1.1)
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samplerate_ratio = 1.1;
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} else {
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} else {
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// Starting up, wait for more info
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qs = alsaqueue.qsize();
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qs = alsaqueue.qsize();
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samplerate_ratio = 1.0;
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samplerate_ratio = 1.0;
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// it = 0;
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}
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}
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// Average the rate value to eliminate fast oscillations
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samplerate_ratio = filter(samplerate_ratio);
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samplerate_ratio = filter(samplerate_ratio);
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unsigned int tot_samples = tsk->m_bytes / (tsk->m_bits/8);
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unsigned int tot_samples = tsk->m_bytes / (tsk->m_bits/8);
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if ((unsigned int)src_data.input_frames < tot_samples / tsk->m_chans) {
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if ((unsigned int)src_data.input_frames < tot_samples / tsk->m_chans) {
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int bytes = tot_samples * sizeof(float);
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int bytes = tot_samples * sizeof(float);
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