813 lines
26 KiB
C
813 lines
26 KiB
C
/*
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* Copyright (C) 2010 DSD Author
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* GPG Key ID: 0x3F1D7FD0 (74EF 430D F7F2 0A48 FCE6 F630 FAA2 635D 3F1D 7FD0)
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*
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* Permission to use, copy, modify, and/or distribute this software for any
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* purpose with or without fee is hereby granted, provided that the above
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* copyright notice and this permission notice appear in all copies.
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*
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* THE SOFTWARE IS PROVIDED "AS IS" AND ISC DISCLAIMS ALL WARRANTIES WITH
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* REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY
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* AND FITNESS. IN NO EVENT SHALL ISC BE LIABLE FOR ANY SPECIAL, DIRECT,
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* INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES WHATSOEVER RESULTING FROM
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* LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE
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* OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR
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* PERFORMANCE OF THIS SOFTWARE.
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*/
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#include "dsd.h"
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pa_sample_spec ss;
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pa_sample_spec tt;
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pa_sample_spec zz;
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pa_sample_spec cc;
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//pa_channel_map channel_map;
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void openPulseOutput(dsd_opts * opts)
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{
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ss.format = PA_SAMPLE_S16NE;
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ss.channels = opts->pulse_raw_out_channels; //doing tests with 2 channels at 22050 for 44100 audio default in pulse
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ss.rate = opts->pulse_raw_rate_out; //48000
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tt.format = PA_SAMPLE_S16NE;
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tt.channels = opts->pulse_digi_out_channels; //doing tests with 2 channels at 22050 for 44100 audio default in pulse
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tt.rate = opts->pulse_digi_rate_out; //48000, switches to 8000 when using RTL dongle
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//fprintf (stderr,"digi rate out = %d\n", opts->pulse_digi_rate_out);
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//pa_channel_map_init_stereo(&channel_map);
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//ss
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if (opts->monitor_input_audio == 1)
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{
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opts->pulse_raw_dev_out = pa_simple_new(NULL, "DSD FME", PA_STREAM_PLAYBACK, NULL, "Raw Audio Out", &ss, NULL, NULL, NULL);
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}
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//tt
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pa_channel_map* left = 0; //NULL and 0 are same in this context
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pa_channel_map* right = 0; //NULL and 0 are same in this context
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//pa_channel_map* right;
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//pa_channel_map_init(right);
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opts->pulse_digi_dev_out = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, opts->output_name, &tt, left, NULL, NULL);
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//opts->pulse_digi_dev_out = pa_simple_new(NULL, "DSD FME", PA_STREAM_PLAYBACK, NULL, "Digi Audio Out LEFT", &tt, left, NULL, NULL);
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//opts->pulse_raw_dev_out = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, "Digi Audio Out RIGHT", &tt, right, NULL, NULL);
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}
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void openPulseInput(dsd_opts * opts)
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{
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//zz.format = PA_SAMPLE_S16NE;
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//zz.channels = opts->pulse_raw_in_channels;
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//zz.rate = opts->pulse_raw_rate_in; //48000
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cc.format = PA_SAMPLE_S16NE;
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cc.channels = opts->pulse_digi_in_channels;
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cc.rate = opts->pulse_digi_rate_in; //48000
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//zz
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if (opts->monitor_input_audio == 1)
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{
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//opts->pulse_raw_dev_in = pa_simple_new(NULL, "DSD FME", PA_STREAM_RECORD, NULL, "Raw Audio In", &zz, NULL, NULL, NULL);
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}
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//cc
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opts->pulse_digi_dev_in = pa_simple_new(NULL, "DSD-FME", PA_STREAM_RECORD, NULL, opts->output_name, &cc, NULL, NULL, NULL);
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}
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void playRawAudio(dsd_opts * opts, dsd_state * state) {
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short obuf, outl, i, sample2, something;
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something = state->samplesPerSymbol / 1; //1 for buffer steal loops, 5 for rtl loop
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//something = 1;
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if (opts->audio_in_type == 0 && opts->audio_out_type == 0){ //hack, but might as well for this particular type since its nearly perfect
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for (i=0; i < something; i++){
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//read (opts->audio_in_fd, &sample2, 2); //reading here seems to get same speed as gfsk modulation
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//pa_simple_read(opts->pulse_raw_dev_in, &state->pulse_raw_out_buffer, 2, NULL );
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//obuf = state->pulse_raw_out_buffer / 4; //dividing here 'quietens' it down a little bit, helps with clicking and clipping
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//obuf = state->input_sample_buffer;
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obuf = state->pulse_raw_out_buffer;
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//write (opts->audio_out_fd, (void*)&obuf, 2);
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pa_simple_write(opts->pulse_raw_dev_out, (void*)&obuf, 2, NULL);
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}
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}
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#ifdef USE_PORTAUDIO
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if (opts->audio_in_type == 2 && opts->audio_out_type == 0){ //hack, but might as well for this particular type since its nearly perfect
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for (i=0; i < something * 5; i++){ //not sure if this should be 'something' or something * 5 like is was with OSS
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//Pa_ReadStream( opts->audio_in_pa_stream, &sample2, 1 ); //reading here seems to get same speed as gfsk modulation
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obuf = state->pulse_raw_out_buffer;
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//obuf = sample2 / 4; //dividing here 'quietens' it down a little bit, helps with clicking and clipping
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//write (opts->audio_out_fd, (void*)&obuf, 2);
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pa_simple_write(opts->pulse_raw_dev_out, (void*)&obuf, 2, NULL);
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}
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}
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if (opts->audio_in_type == 0 && opts->audio_out_type == 2){ //hack, but might as well for this particular type since its nearly perfect
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for (i=0; i < something; i++){
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//read (opts->audio_in_fd, &sample2, 2); //reading here seems to get same speed as gfsk modulation
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//pa_simple_read(opts->pulse_raw_dev_in, &sample2, 2, NULL );
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//obuf = sample2 / 4; //dividing here 'quietens' it down a little bit, helps with clicking and clipping
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obuf = state->pulse_raw_out_buffer;
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Pa_StartStream( opts->audio_out_pa_stream ); //no promises this work
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Pa_WriteStream( opts->audio_out_pa_stream, (void*)&obuf, 2); //no promises this work
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}
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}
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if (opts->audio_in_type == 2 && opts->audio_out_type == 2){ //hack, but might as well for this particular type since its nearly perfect
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for (i=0; i < something * 5; i++){ //not sure if this should be 'something' or something * 5 like is was with OSS
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//Pa_ReadStream( opts->audio_in_pa_stream, &sample2, 1 ); //reading here seems to get same speed as gfsk modulation
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//obuf = sample2 / 4; //dividing here 'quietens' it down a little bit, helps with clicking and clipping
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obuf = state->pulse_raw_out_buffer;
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Pa_StartStream( opts->audio_out_pa_stream ); //no promises this work
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Pa_WriteStream( opts->audio_out_pa_stream, (void*)&obuf, 2); //no promises this work
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}
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}
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#endif
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something = state->samplesPerSymbol / 5; //change to 5 for rtl samples, flips back to 1 up top next loop
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//something = 1;
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if (opts->audio_in_type == 1 || opts->audio_in_type == 3){ //only plays at 83% and sounds like s sadly, but can't get samples directly from sdr when its already being polled for samples
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//also, can't get samples from stdin more than once either it seems, unless I borked it when I tried it earlier, so lumping it in here as well
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for (i=0; i < something; i++)
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{
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obuf = state->input_sample_buffer; //dividing here 'quietens' it down a little bit, helps with clicking and clipping
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outl = sizeof(obuf) ; //obuf length, I think its always equal to 2
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if (opts->audio_out_type == 0){
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//write (opts->audio_out_fd, (void*)&obuf, outl);
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pa_simple_write(opts->pulse_raw_dev_out, (void*)&obuf, outl, NULL);
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//fprintf (stderr,"writing samples");
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}
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#ifdef USE_PORTAUDIO //no idea if this method will work, since I can't test it
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if (opts->audio_out_type == 2){
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short err;
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err = Pa_IsStreamActive( opts->audio_out_pa_stream );
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if(err == 0)
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{
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err = Pa_StartStream( opts->audio_out_pa_stream );
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if( err != paNoError ){
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fprintf (stderr,"Can't Start PA Stream");
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}
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err = Pa_WriteStream( opts->audio_out_pa_stream, (void*)&obuf, outl );
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if( err != paNoError ){
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fprintf (stderr,"Can't Write PA Stream");
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}
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}
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}
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#endif
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} // end for loop
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} //end if statement
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} //end playRawAudio
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void
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processAudio (dsd_opts * opts, dsd_state * state)
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{
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int i, n;
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float aout_abs, max, gainfactor, gaindelta, maxbuf;
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if (opts->audio_gain == (float) 0)
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{
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// detect max level
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max = 0;
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state->audio_out_temp_buf_p = state->audio_out_temp_buf;
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//state->audio_out_temp_buf_pR = state->audio_out_temp_bufR;
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for (n = 0; n < 160; n++)
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{
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aout_abs = fabsf (*state->audio_out_temp_buf_p);
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if (aout_abs > max)
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{
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max = aout_abs;
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}
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state->audio_out_temp_buf_p++;
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//state->audio_out_temp_buf_pR++;
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}
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*state->aout_max_buf_p = max;
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//*state->aout_max_buf_pR = max;
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state->aout_max_buf_p++;
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//state->aout_max_buf_pR++;
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state->aout_max_buf_idx++;
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//state->aout_max_buf_idxR++;
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if (state->aout_max_buf_idx > 24)
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{
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state->aout_max_buf_idx = 0;
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state->aout_max_buf_p = state->aout_max_buf;
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}
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//if (state->aout_max_buf_idxR > 24)
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// {
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// state->aout_max_buf_idxR = 0;
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// state->aout_max_buf_pR = state->aout_max_bufR;
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// }
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// lookup max history
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for (i = 0; i < 25; i++)
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{
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maxbuf = state->aout_max_buf[i];
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if (maxbuf > max)
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{
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max = maxbuf;
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}
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}
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// determine optimal gain level
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if (max > (float) 0)
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{
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gainfactor = ((float) 30000 / max);
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}
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else
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{
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gainfactor = (float) 50;
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}
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if (gainfactor < state->aout_gain)
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{
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state->aout_gain = gainfactor;
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gaindelta = (float) 0;
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}
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else
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{
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if (gainfactor > (float) 50)
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{
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gainfactor = (float) 50;
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}
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gaindelta = gainfactor - state->aout_gain;
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if (gaindelta > ((float) 0.05 * state->aout_gain))
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{
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gaindelta = ((float) 0.05 * state->aout_gain);
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}
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}
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gaindelta /= (float) 160;
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}
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else
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{
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gaindelta = (float) 0;
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}
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if(opts->audio_gain >= 0)
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{
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// adjust output gain
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state->audio_out_temp_buf_p = state->audio_out_temp_buf;
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for (n = 0; n < 160; n++)
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{
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*state->audio_out_temp_buf_p = (state->aout_gain + ((float) n * gaindelta)) * (*state->audio_out_temp_buf_p);
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state->audio_out_temp_buf_p++;
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}
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state->aout_gain += ((float) 160 * gaindelta);
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}
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// copy audio data to output buffer and upsample if necessary
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state->audio_out_temp_buf_p = state->audio_out_temp_buf;
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if (opts->split == 0)
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{
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for (n = 0; n < 160; n++)
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{
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upsample (state, *state->audio_out_temp_buf_p);
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state->audio_out_temp_buf_p++;
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state->audio_out_float_buf_p += 6;
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state->audio_out_idx += 6;
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state->audio_out_idx2 += 6;
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}
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state->audio_out_float_buf_p -= (960 + opts->playoffset);
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// copy to output (short) buffer
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for (n = 0; n < 960; n++)
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{
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if (*state->audio_out_float_buf_p > 32767.0F)
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{
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*state->audio_out_float_buf_p = 32767.0F;
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}
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else if (*state->audio_out_float_buf_p < -32768.0F)
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{
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*state->audio_out_float_buf_p = -32768.0F;
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}
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*state->audio_out_buf_p = (short) *state->audio_out_float_buf_p;
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state->audio_out_buf_p++;
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state->audio_out_float_buf_p++;
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}
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state->audio_out_float_buf_p += opts->playoffset;
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}
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else
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{
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for (n = 0; n < 160; n++)
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{
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if (*state->audio_out_temp_buf_p > 32767.0F)
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{
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*state->audio_out_temp_buf_p = 32767.0F;
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}
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else if (*state->audio_out_temp_buf_p < -32768.0F)
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{
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*state->audio_out_temp_buf_p = -32768.0F;
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}
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*state->audio_out_buf_p = (short) *state->audio_out_temp_buf_p;
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state->audio_out_buf_p++;
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state->audio_out_temp_buf_p++;
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state->audio_out_idx++;
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state->audio_out_idx2++;
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}
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}
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}
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void
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writeSynthesizedVoice (dsd_opts * opts, dsd_state * state)
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{
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int n;
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short aout_buf[160];
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short *aout_buf_p;
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// for(n=0; n<160; n++)
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// fprintf (stderr,"%d ", ((short*)(state->audio_out_temp_buf))[n]);
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// fprintf (stderr,"\n");
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aout_buf_p = aout_buf;
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state->audio_out_temp_buf_p = state->audio_out_temp_buf;
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for (n = 0; n < 160; n++)
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{
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if (*state->audio_out_temp_buf_p > (float) 32767)
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{
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*state->audio_out_temp_buf_p = (float) 32767;
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}
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else if (*state->audio_out_temp_buf_p < (float) -32768)
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{
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*state->audio_out_temp_buf_p = (float) -32768;
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}
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*aout_buf_p = (short) *state->audio_out_temp_buf_p;
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aout_buf_p++;
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state->audio_out_temp_buf_p++;
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}
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sf_write_short(opts->wav_out_f, aout_buf, 160);
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/*
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int n;
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short aout_buf[160];
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short *aout_buf_p;
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ssize_t result;
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aout_buf_p = aout_buf;
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state->audio_out_temp_buf_p = state->audio_out_temp_buf;
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for (n = 0; n < 160; n++)
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{
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if (*state->audio_out_temp_buf_p > (float) 32760)
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{
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*state->audio_out_temp_buf_p = (float) 32760;
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}
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else if (*state->audio_out_temp_buf_p < (float) -32760)
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{
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*state->audio_out_temp_buf_p = (float) -32760;
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}
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*aout_buf_p = (short) *state->audio_out_temp_buf_p;
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aout_buf_p++;
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state->audio_out_temp_buf_p++;
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}
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result = write (opts->wav_out_fd, aout_buf, 320);
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fflush (opts->wav_out_f);
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state->wav_out_bytes += 320;
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*/
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}
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void
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playSynthesizedVoice (dsd_opts * opts, dsd_state * state)
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{
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ssize_t result;
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if (state->audio_out_idx > opts->delay)
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{
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// output synthesized speech to sound card
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if(opts->audio_out_type == 2)
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{
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#ifdef USE_PORTAUDIO
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PaError err = paNoError;
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do
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{
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long available = Pa_GetStreamWriteAvailable( opts->audio_out_pa_stream );
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if(available < 0)
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err = available;
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//fprintf (stderr,"Frames available: %d\n", available);
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if( err != paNoError )
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break;
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if(available > SAMPLE_RATE_OUT * PA_LATENCY_OUT)
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{
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//It looks like this might not be needed for very small latencies. However, it's definitely needed for a bit larger ones.
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//When PA_LATENCY_OUT == 0.500 I get output buffer underruns if I don't use this. With PA_LATENCY_OUT <= 0.100 I don't see those happen.
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//But with PA_LATENCY_OUT < 0.100 I run the risk of choppy audio and stream errors.
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fprintf (stderr,"\nSyncing voice output stream\n");
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err = Pa_StopStream( opts->audio_out_pa_stream );
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if( err != paNoError )
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break;
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}
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err = Pa_IsStreamActive( opts->audio_out_pa_stream );
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if(err == 0)
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{
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fprintf (stderr,"Start voice output stream\n");
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err = Pa_StartStream( opts->audio_out_pa_stream );
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}
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else if(err == 1)
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{
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err = paNoError;
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}
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if( err != paNoError )
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break;
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err = Pa_WriteStream( opts->audio_out_pa_stream, (state->audio_out_buf_p - state->audio_out_idx), state->audio_out_idx );
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if( err != paNoError )
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break;
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} while(0);
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if( err != paNoError )
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{
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fprintf (stderr, "An error occured while using the portaudio output stream\n" );
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fprintf (stderr, "Error number: %d\n", err );
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fprintf (stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
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}
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#endif
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}
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else
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if (opts->monitor_input_audio == 1){
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//pa_simple_flush(opts->pulse_raw_dev_in, NULL);
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//pa_simple_flush(opts->pulse_raw_dev_out, NULL);
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//state->pulse_raw_out_buffer = 0;
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//pa_simple_write(opts->pulse_raw_dev_out, (void*)&state->pulse_raw_out_buffer, 2, NULL);
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}
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//two slot audio testing, still need to seperate channels first internally, but this will play them out of different streams
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/*
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if(state->currentslot == 0)
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{
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pa_simple_write(opts->pulse_digi_dev_out, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2), NULL); //Yay! It works.
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}
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if(state->currentslot == 1)
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{
|
|
pa_simple_write(opts->pulse_raw_dev_out, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2), NULL); //Yay! It works.
|
|
//pa_simple_write(opts->pulse_raw_dev_out, (state->audio_out_buf_pR - state->audio_out_idxR), (state->audio_out_idxR * 2), NULL); //Yay! It works.
|
|
}
|
|
*/
|
|
|
|
pa_simple_write(opts->pulse_digi_dev_out, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2), NULL); //Yay! It works.
|
|
state->audio_out_idx = 0;
|
|
}
|
|
|
|
if (state->audio_out_idx2 >= 800000)
|
|
{
|
|
state->audio_out_float_buf_p = state->audio_out_float_buf + 100;
|
|
state->audio_out_buf_p = state->audio_out_buf + 100;
|
|
memset (state->audio_out_float_buf, 0, 100 * sizeof (float));
|
|
memset (state->audio_out_buf, 0, 100 * sizeof (short));
|
|
state->audio_out_idx2 = 0;
|
|
}
|
|
}
|
|
|
|
#ifdef USE_PORTAUDIO
|
|
int getPADevice(char* dev, int input, PaStream** stream)
|
|
{
|
|
int devnum = atoi(dev + 3);
|
|
fprintf (stderr,"Using portaudio device %d.\n", devnum);
|
|
|
|
PaError err;
|
|
|
|
int numDevices = Pa_GetDeviceCount();
|
|
if( numDevices < 0 )
|
|
{
|
|
fprintf (stderr, "ERROR: Pa_GetDeviceCount returned 0x%x\n", numDevices );
|
|
err = numDevices;
|
|
goto error;
|
|
}
|
|
if( devnum >= numDevices)
|
|
{
|
|
fprintf (stderr, "ERROR: Requested device %d is larger than number of devices.\n", devnum );
|
|
return(1);
|
|
}
|
|
|
|
const PaDeviceInfo *deviceInfo = Pa_GetDeviceInfo( devnum );
|
|
|
|
/* print device name */
|
|
#ifdef WIN32
|
|
{ /* Use wide char on windows, so we can show UTF-8 encoded device names */
|
|
wchar_t wideName[MAX_PATH];
|
|
MultiByteToWideChar(CP_UTF8, 0, deviceInfo->name, -1, wideName, MAX_PATH-1);
|
|
wprintf ( L"Name = %s\n", wideName );
|
|
}
|
|
#else
|
|
fprintf (stderr, "Name = %s\n", deviceInfo->name );
|
|
#endif
|
|
if((input == 1) && (deviceInfo->maxInputChannels == 0))
|
|
{
|
|
fprintf (stderr, "ERROR: Requested device %d is not an input device.\n", devnum );
|
|
return(1);
|
|
}
|
|
if((input == 0) && (deviceInfo->maxOutputChannels == 0))
|
|
{
|
|
fprintf (stderr, "ERROR: Requested device %d is not an output device.\n", devnum );
|
|
return(1);
|
|
}
|
|
|
|
//Create stream parameters
|
|
PaStreamParameters parameters;
|
|
parameters.device = devnum;
|
|
parameters.channelCount = 1; /* mono */
|
|
parameters.sampleFormat = paInt16; //Shorts
|
|
parameters.suggestedLatency = (input == 1) ? PA_LATENCY_IN : PA_LATENCY_OUT;
|
|
parameters.hostApiSpecificStreamInfo = NULL;
|
|
|
|
//Open stream
|
|
err = Pa_OpenStream(
|
|
stream,
|
|
(input == 1) ? ¶meters : NULL,
|
|
(input == 0) ? ¶meters : NULL,
|
|
(input == 1) ? SAMPLE_RATE_IN : SAMPLE_RATE_OUT,
|
|
PA_FRAMES_PER_BUFFER,
|
|
paClipOff,
|
|
NULL /*callback*/,
|
|
NULL );
|
|
if( err != paNoError ) goto error;
|
|
|
|
return 0;
|
|
|
|
error:
|
|
fprintf (stderr, "An error occured while initializing a portaudio stream\n" );
|
|
fprintf (stderr, "Error number: %d\n", err );
|
|
fprintf (stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
|
|
return err;
|
|
}
|
|
#endif
|
|
|
|
void
|
|
openAudioOutDevice (dsd_opts * opts, int speed)
|
|
{
|
|
// get info of device/file
|
|
/*
|
|
if(strncmp(opts->audio_out_dev, "pulse", 5) == 0)
|
|
{
|
|
opts->audio_out_type == 0;
|
|
}
|
|
*/
|
|
if(strncmp(opts->audio_out_dev, "pa:", 3) == 0)
|
|
{
|
|
opts->audio_out_type = 2;
|
|
#ifdef USE_PORTAUDIO
|
|
int err = getPADevice(opts->audio_out_dev, 0, &opts->audio_out_pa_stream);
|
|
if(err != 0)
|
|
exit(err);
|
|
#else
|
|
fprintf (stderr,"Error, Portaudio support not compiled.\n");
|
|
exit(1);
|
|
#endif
|
|
}
|
|
if(strncmp(opts->audio_in_dev, "pulse", 5) == 0)
|
|
{
|
|
opts->audio_in_type = 0;
|
|
}
|
|
else
|
|
{
|
|
struct stat stat_buf;
|
|
if(stat(opts->audio_out_dev, &stat_buf) != 0 && strncmp(opts->audio_out_dev, "pulse", 5 != 0)) //HERE
|
|
{
|
|
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_out_dev);
|
|
exit(1);
|
|
}
|
|
|
|
if( (!(S_ISCHR(stat_buf.st_mode) || S_ISBLK(stat_buf.st_mode))) && strncmp(opts->audio_out_dev, "pulse", 5 != 0))
|
|
{
|
|
// this is not a device
|
|
fprintf (stderr,"Error, %s is not a device. use -w filename for wav output.\n", opts->audio_out_dev);
|
|
exit(1);
|
|
}
|
|
/*
|
|
#ifdef SOLARIS
|
|
sample_info_t aset, aget;
|
|
|
|
opts->audio_out_fd = open (opts->audio_out_dev, O_WRONLY);
|
|
if (opts->audio_out_fd == -1)
|
|
{
|
|
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_out_dev);
|
|
//exit (1);
|
|
}
|
|
|
|
// get current
|
|
ioctl (opts->audio_out_fd, AUDIO_GETINFO, &aset);
|
|
aset.record.sample_rate = speed;
|
|
aset.play.sample_rate = speed;
|
|
aset.record.channels = 1;
|
|
aset.play.channels = 1;
|
|
aset.record.precision = 16;
|
|
aset.play.precision = 16;
|
|
aset.record.encoding = AUDIO_ENCODING_LINEAR;
|
|
aset.play.encoding = AUDIO_ENCODING_LINEAR;
|
|
|
|
if (ioctl (opts->audio_out_fd, AUDIO_SETINFO, &aset) == -1)
|
|
{
|
|
fprintf (stderr,"Error setting sample device parameters\n");
|
|
exit (1);
|
|
}
|
|
#endif
|
|
|
|
#if defined(BSD) && !defined(__APPLE__)
|
|
|
|
int fmt;
|
|
|
|
opts->audio_out_fd = open (opts->audio_out_dev, O_WRONLY);
|
|
if (opts->audio_out_fd == -1)
|
|
{
|
|
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_out_dev);
|
|
opts->audio_out = 0;
|
|
//exit(1);
|
|
}
|
|
|
|
fmt = 0;
|
|
if (ioctl (opts->audio_out_fd, SNDCTL_DSP_RESET) < 0)
|
|
{
|
|
fprintf (stderr,"ioctl reset error \n");
|
|
}
|
|
fmt = speed;
|
|
if (ioctl (opts->audio_out_fd, SNDCTL_DSP_SPEED, &fmt) < 0)
|
|
{
|
|
fprintf (stderr,"ioctl speed error \n");
|
|
}
|
|
fmt = 0;
|
|
if (ioctl (opts->audio_out_fd, SNDCTL_DSP_STEREO, &fmt) < 0)
|
|
{
|
|
fprintf (stderr,"ioctl stereo error \n");
|
|
}
|
|
fmt = AFMT_S16_LE;
|
|
if (ioctl (opts->audio_out_fd, SNDCTL_DSP_SETFMT, &fmt) < 0)
|
|
{
|
|
fprintf (stderr,"ioctl setfmt error \n");
|
|
}
|
|
|
|
#endif
|
|
*/
|
|
}
|
|
fprintf (stderr,"Audio Out Device: %s\n", opts->audio_out_dev);
|
|
}
|
|
|
|
void
|
|
openAudioInDevice (dsd_opts * opts)
|
|
{
|
|
// get info of device/file
|
|
if(strncmp(opts->audio_in_dev, "-", 1) == 0)
|
|
{
|
|
opts->audio_in_type = 1;
|
|
opts->audio_in_file_info = calloc(1, sizeof(SF_INFO));
|
|
opts->audio_in_file_info->samplerate=48000;
|
|
opts->pulse_digi_rate_out = 8000; //set out rate to 8000 for stdin input
|
|
opts->audio_in_file_info->channels=1;
|
|
opts->audio_in_file_info->seekable=0;
|
|
opts->audio_in_file_info->format=SF_FORMAT_RAW|SF_FORMAT_PCM_16|SF_ENDIAN_LITTLE;
|
|
opts->audio_in_file = sf_open_fd(fileno(stdin), SFM_READ, opts->audio_in_file_info, 0);
|
|
|
|
if(opts->audio_in_file == NULL) {
|
|
fprintf (stderr,"Error, couldn't open stdin with libsndfile: %s\n", sf_strerror(NULL));
|
|
exit(1); //had this one disabled, re-enabling it now
|
|
}
|
|
}
|
|
else if(strncmp(opts->audio_in_dev, "pa:", 2) == 0)
|
|
{
|
|
opts->audio_in_type = 2;
|
|
#ifdef USE_PORTAUDIO
|
|
int err = getPADevice(opts->audio_in_dev, 1, &opts->audio_in_pa_stream);
|
|
if(err != 0)
|
|
exit(err);
|
|
|
|
if (opts->split == 0)
|
|
{
|
|
int err = getPADevice(opts->audio_in_dev, 0, &opts->audio_out_pa_stream);
|
|
if(err != 0)
|
|
exit(err);
|
|
}
|
|
|
|
#else
|
|
fprintf (stderr,"Error, Portaudio support not compiled.\n");
|
|
exit(1);
|
|
#endif
|
|
}
|
|
else if(strncmp(opts->audio_in_dev, "rtl", 3) == 0)
|
|
{
|
|
opts->audio_in_type = 3;
|
|
}
|
|
else if(strncmp(opts->audio_in_dev, "pulse", 5) == 0)
|
|
{
|
|
opts->audio_in_type = 0;
|
|
}
|
|
else
|
|
{
|
|
struct stat stat_buf;
|
|
if (stat(opts->audio_in_dev, &stat_buf) != 0)
|
|
{
|
|
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_in_dev);
|
|
exit(1);
|
|
}
|
|
if (S_ISREG(stat_buf.st_mode))
|
|
{
|
|
// is this a regular file? then process with libsndfile.
|
|
//opts->pulse_digi_rate_out = 8000; //this for wav files input?
|
|
opts->audio_in_type = 1;
|
|
opts->audio_in_file_info = calloc(1, sizeof(SF_INFO));
|
|
opts->audio_in_file_info->channels = 1;
|
|
opts->audio_in_file = sf_open(opts->audio_in_dev, SFM_READ, opts->audio_in_file_info);
|
|
|
|
if(opts->audio_in_file == NULL)
|
|
{
|
|
fprintf (stderr,"Error, couldn't open file %s\n", opts->audio_in_dev);
|
|
exit(1);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// this is a device, use old handling, pulse audio now
|
|
opts->audio_in_type = 0;
|
|
/*
|
|
#ifdef SOLARIS
|
|
sample_info_t aset, aget;
|
|
int rgain;
|
|
|
|
rgain = 64;
|
|
|
|
if (opts->split == 1)
|
|
{
|
|
opts->audio_in_fd = open (opts->audio_in_dev, O_RDONLY);
|
|
}
|
|
else
|
|
{
|
|
opts->audio_in_fd = open (opts->audio_in_dev, O_RDWR);
|
|
}
|
|
if (opts->audio_in_fd == -1)
|
|
{
|
|
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_in_dev);
|
|
//exit(1);
|
|
}
|
|
|
|
// get current
|
|
ioctl (opts->audio_in_fd, AUDIO_GETINFO, &aset);
|
|
|
|
aset.record.sample_rate = SAMPLE_RATE_IN;
|
|
aset.play.sample_rate = SAMPLE_RATE_IN;
|
|
aset.record.channels = 1;
|
|
aset.play.channels = 1;
|
|
aset.record.precision = 16;
|
|
aset.play.precision = 16;
|
|
aset.record.encoding = AUDIO_ENCODING_LINEAR;
|
|
aset.play.encoding = AUDIO_ENCODING_LINEAR;
|
|
aset.record.port = AUDIO_LINE_IN;
|
|
aset.record.gain = rgain;
|
|
|
|
if (ioctl (opts->audio_in_fd, AUDIO_SETINFO, &aset) == -1)
|
|
{
|
|
fprintf (stderr,"Error setting sample device parameters\n");
|
|
exit (1);
|
|
}
|
|
#endif
|
|
*/
|
|
/*
|
|
#if defined(BSD) && !defined(__APPLE__)
|
|
int fmt;
|
|
|
|
if (opts->split == 1)
|
|
{
|
|
opts->audio_in_fd = open (opts->audio_in_dev, O_RDONLY);
|
|
}
|
|
else
|
|
{
|
|
opts->audio_in_fd = open (opts->audio_in_dev, O_RDWR);
|
|
}
|
|
|
|
if (opts->audio_in_fd == -1)
|
|
{
|
|
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_in_dev);
|
|
opts->audio_out = 0;
|
|
}
|
|
|
|
fmt = 0;
|
|
if (ioctl (opts->audio_in_fd, SNDCTL_DSP_RESET) < 0)
|
|
{
|
|
fprintf (stderr,"ioctl reset error \n");
|
|
}
|
|
fmt = SAMPLE_RATE_IN;
|
|
if (ioctl (opts->audio_in_fd, SNDCTL_DSP_SPEED, &fmt) < 0)
|
|
{
|
|
fprintf (stderr,"ioctl speed error \n");
|
|
}
|
|
fmt = 0;
|
|
if (ioctl (opts->audio_in_fd, SNDCTL_DSP_STEREO, &fmt) < 0)
|
|
{
|
|
fprintf (stderr,"ioctl stereo error \n");
|
|
}
|
|
fmt = AFMT_S16_LE;
|
|
if (ioctl (opts->audio_in_fd, SNDCTL_DSP_SETFMT, &fmt) < 0)
|
|
{
|
|
fprintf (stderr,"ioctl setfmt error \n");
|
|
}
|
|
#endif
|
|
*/
|
|
}
|
|
}
|
|
|
|
if (opts->split == 1)
|
|
{
|
|
fprintf (stderr,"Audio In Device: %s\n", opts->audio_in_dev);
|
|
}
|
|
else
|
|
{
|
|
fprintf (stderr,"Audio In/Out Device: %s\n", opts->audio_in_dev);
|
|
}
|
|
}
|