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b/src/alsadirect.cpp |
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/* Copyright (C) 2014 J.F.Dockes
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the
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* Free Software Foundation, Inc.,
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* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#include "config.h"
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#include <string.h>
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#include <sys/types.h>
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#include <iostream>
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#include <queue>
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#include <alsa/asoundlib.h>
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#include <samplerate.h>
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#include "log.h"
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#include "rcvqueue.h"
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using namespace std;
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#ifndef MIN
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#define MIN(A, B) ((A) < (B) ? (A) : (B))
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#endif
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// The queue for audio blocks ready for alsa
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static const unsigned int qs = 200;
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static const unsigned int qt = qs/2;
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// the 40 value should be computed from the alsa buffer size. It's
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// there becausee we have a jump on the first alsa write (alsa buffer
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// is empty).
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static const unsigned int qit = qs/2 + 40;
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static WorkQueue<AudioMessage*> alsaqueue("alsaqueue", qs);
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static snd_pcm_t *pcm;
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static void *alsawriter(void *p)
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{
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if (!alsaqueue.waitminsz(qit)) {
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LOGERR("alsawriter: waitminsz failed\n");
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return (void *)1;
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}
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while (true) {
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AudioMessage *tsk = 0;
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size_t qsz;
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if (!alsaqueue.take(&tsk, &qsz)) {
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// TBD: reset alsa?
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alsaqueue.workerExit();
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return (void*)1;
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}
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// Bufs
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snd_pcm_uframes_t frames =
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tsk->m_bytes / (tsk->m_chans * (tsk->m_bits/8));
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snd_pcm_sframes_t ret = snd_pcm_writei(pcm, tsk->m_buf, frames);
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if (ret != int(frames)) {
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LOGERR("snd-cm_writei(" << frames <<" frames) failed: ret: " <<
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ret << endl);
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if (ret < 0)
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snd_pcm_prepare(pcm);
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// return (void *)1;
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}
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}
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}
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static bool alsa_init(AudioMessage *tsk)
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{
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snd_pcm_hw_params_t *hw_params;
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int err;
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// static const string dev("plughw:CARD=PCH,DEV=0");
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static const string dev("hw:2,0");
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const char *cmd = "";
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unsigned int actual_rate = tsk->m_freq;
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int dir=0;
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if ((err = snd_pcm_open(&pcm, dev.c_str(),
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SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
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LOGERR("alsa_init: snd_pcm_open " << dev << " " <<
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snd_strerror(err) << endl);
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return false;;
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}
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if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
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LOGERR("alsa_init: snd_pcm_hw_params_malloc " <<
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snd_strerror(err) << endl);
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snd_pcm_close(pcm);
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return false;
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}
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cmd = "snd_pcm_hw_params_any";
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if ((err = snd_pcm_hw_params_any(pcm, hw_params)) < 0) {
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goto error;
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}
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cmd = "snd_pcm_hw_params_set_access";
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if ((err =
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snd_pcm_hw_params_set_access(pcm, hw_params,
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SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
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goto error;
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}
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cmd = "snd_pcm_hw_params_set_format";
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if ((err =
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snd_pcm_hw_params_set_format(pcm, hw_params,
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SND_PCM_FORMAT_S16_LE)) < 0) {
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goto error;
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}
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cmd = "snd_pcm_hw_params_set_rate_near";
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if ((err = snd_pcm_hw_params_set_rate_near(pcm, hw_params,
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&actual_rate, &dir)) < 0) {
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goto error;
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}
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cmd = "snd_pcm_hw_params_set_channels";
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if ((err = snd_pcm_hw_params_set_channels(pcm, hw_params,
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tsk->m_chans)) < 0) {
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goto error;
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}
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cmd = "snd_pcm_hw_params";
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if ((err = snd_pcm_hw_params(pcm, hw_params)) < 0) {
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goto error;
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}
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snd_pcm_hw_params_free(hw_params);
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return true;
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error:
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snd_pcm_hw_params_free(hw_params);
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LOGERR("alsa_init: " << cmd << " " << snd_strerror(err) << endl);
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return false;;
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}
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static void *audioEater(void *cls)
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{
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AudioEater::Context *ctxt = (AudioEater::Context*)cls;
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LOGDEB("alsaEater: queue " << ctxt->queue << endl);
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WorkQueue<AudioMessage*> *queue = ctxt->queue;
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delete ctxt;
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bool qinit = false;
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int src_error = 0;
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SRC_STATE *src_state = 0;
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SRC_DATA 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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float samplerate_ratio = 1.0;
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while (true) {
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AudioMessage *tsk = 0;
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size_t qsz;
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if (!queue->take(&tsk, &qsz)) {
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// TBD: reset alsa?
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queue->workerExit();
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return (void*)1;
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}
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if (src_state == 0) {
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if (!alsa_init(tsk)) {
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queue->workerExit();
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return (void *)1;
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}
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// BEST_QUALITY yields approx 25% cpu on a core i7
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// 4770T. Obviously too much, actually might not be
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// sustainable.
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// MEDIUM_QUALITY is around 10%
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// FASTEST is 4-5%. Given that this is process-wide, probably
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// a couple % in fact.
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// To be re-evaluated on the pi...
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src_state = src_new(SRC_SINC_FASTEST, tsk->m_chans, &src_error);
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}
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if (qinit) {
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float qs = alsaqueue.qsize();
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float t = ((qt - qs) / qt);
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float adj = t * t / 10;
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if (alsaqueue.qsize() < qt) {
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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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} else {
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samplerate_ratio -= adj;
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if (samplerate_ratio < 0.9)
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samplerate_ratio = 0.9;
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}
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}
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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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int bytes = tot_samples * sizeof(float);
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src_data.data_in = (float *)realloc(src_data.data_in, bytes);
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src_data.data_out = (float *)realloc(src_data.data_out, 2 * bytes);
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src_data.input_frames = tot_samples / tsk->m_chans;
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// Available space for output
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src_data.output_frames = 2 * src_data.input_frames;
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}
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src_data.src_ratio = samplerate_ratio;
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src_data.end_of_input = 0;
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switch (tsk->m_bits) {
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case 16: {
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const short *sp = (const short *)tsk->m_buf;
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for (unsigned int i = 0; i < tot_samples; i++) {
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src_data.data_in[i] = *sp++;
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}
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break;
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}
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case 24:
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case 32:
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default:
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abort();
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}
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int ret = src_process(src_state, &src_data);
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if (ret) {
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LOGERR("src_process: " << src_strerror(ret) << endl);
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continue;
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}
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{
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static int cnt;
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if (cnt++ == 100) {
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LOGDEB("samplerate: "
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" qsize " << alsaqueue.qsize() <<
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" ratio " << samplerate_ratio <<
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" in " << src_data.input_frames <<
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" consumed " << src_data.input_frames_used <<
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" out " << src_data.output_frames_gen << endl);
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cnt = 0;
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}
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}
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tot_samples = src_data.output_frames_gen * tsk->m_chans;
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if (src_data.output_frames_gen > src_data.input_frames) {
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tsk->m_bytes = tot_samples * (tsk->m_bits / 8);
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tsk->m_buf = (char *)realloc(tsk->m_buf, tsk->m_bytes);
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if (!tsk->m_buf)
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abort();
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}
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// Output is always 16 bits lsb first for now. We should
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// probably dither the lsb ?
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tsk->m_bits = 16;
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{
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#ifdef WORDS_BIGENDIAN
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unsigned char *ocp = (unsigned char *)tsk->m_buf;
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short val;
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unsigned char *icp = (unsigned char *)&val;
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for (unsigned int i = 0; i < tot_samples; i++) {
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val = src_data.data_out[i];;
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*ocp++ = icp[1];
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*ocp++ = icp[0];
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}
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tsk->m_bytes = ocp - tsk->m_buf;
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#else
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short *sp = (short *)tsk->m_buf;
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for (unsigned int i = 0; i < tot_samples; i++) {
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*sp++ = src_data.data_out[i];
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}
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tsk->m_bytes = (char *)sp - tsk->m_buf;
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#endif
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}
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if (!alsaqueue.put(tsk)) {
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LOGERR("alsaEater: queue put failed\n");
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return (void *)1;
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}
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if (alsaqueue.qsize() >= qit)
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qinit = true;
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}
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}
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AudioEater alsaAudioEater(AudioEater::BO_HOST, &audioEater);
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