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DSD demod plugin: DSD proper integration interim state #1

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This commit is contained in:
f4exb 2016-04-09 13:12:48 +02:00
parent 9ef7497c40
commit 5872ee0ca7
10 changed files with 526 additions and 403 deletions
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53
dsd/dsd_audio.c
View File

@ -17,8 +17,7 @@
#include "dsd.h"
void
processAudio (dsd_opts * opts, dsd_state * state)
void processAudio(dsd_opts * opts, dsd_state * state)
{
int i, n;
@ -90,12 +89,14 @@ processAudio (dsd_opts * opts, dsd_state * state)
gaindelta = (float) 0;
}
if(opts->audio_gain >= 0){
if (opts->audio_gain >= 0)
{
// adjust output gain
state->audio_out_temp_buf_p = state->audio_out_temp_buf;
for (n = 0; n < 160; n++)
{
*state->audio_out_temp_buf_p = (state->aout_gain + ((float) n * gaindelta)) * (*state->audio_out_temp_buf_p);
*state->audio_out_temp_buf_p = (state->aout_gain
+ ((float) n * gaindelta)) * (*state->audio_out_temp_buf_p);
state->audio_out_temp_buf_p++;
}
state->aout_gain += ((float) 160 * gaindelta);
@ -103,7 +104,7 @@ processAudio (dsd_opts * opts, dsd_state * state)
// copy audio datat to output buffer and upsample if necessary
state->audio_out_temp_buf_p = state->audio_out_temp_buf;
if (opts->split == 0)
if ((opts->split == 0) || (opts->upsample))
{
for (n = 0; n < 160; n++)
{
@ -152,8 +153,7 @@ processAudio (dsd_opts * opts, dsd_state * state)
}
}
void
writeSynthesizedVoice (dsd_opts * opts, dsd_state * state)
void writeSynthesizedVoice(dsd_opts * opts, dsd_state * state)
{
#ifdef USE_LIBSNDFILE
int n;
@ -214,8 +214,7 @@ writeSynthesizedVoice (dsd_opts * opts, dsd_state * state)
#endif
}
void
playSynthesizedVoice (dsd_opts * opts, dsd_state * state)
void playSynthesizedVoice(dsd_opts * opts, dsd_state * state)
{
ssize_t result;
@ -225,12 +224,16 @@ playSynthesizedVoice (dsd_opts * opts, dsd_state * state)
// output synthesized speech to sound card
if (opts->audio_out_fd == -1)
{
memcpy(state->output_buffer + state->output_offset, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2));
memcpy(state->output_buffer + state->output_offset,
(state->audio_out_buf_p - state->audio_out_idx),
(state->audio_out_idx * 2));
state->output_offset += state->audio_out_idx;
}
else
{
result = write (opts->audio_out_fd, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2));
result = write(opts->audio_out_fd,
(state->audio_out_buf_p - state->audio_out_idx),
(state->audio_out_idx * 2));
}
state->audio_out_idx = 0;
}
@ -245,18 +248,21 @@ playSynthesizedVoice (dsd_opts * opts, dsd_state * state)
}
}
void
openAudioOutDevice (dsd_opts * opts, int speed)
void openAudioOutDevice(dsd_opts * opts, int speed)
{
// get info of device/file
struct stat stat_buf;
if(stat(opts->audio_out_dev, &stat_buf) != 0) {
if (stat(opts->audio_out_dev, &stat_buf) != 0)
{
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))) { // this is not a device
fprintf(stderr, "Error, %s is not a device. use -w filename for wav output.\n", opts->audio_out_dev);
if (!(S_ISCHR(stat_buf.st_mode) || S_ISBLK(stat_buf.st_mode)))
{ // 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
@ -325,27 +331,30 @@ openAudioOutDevice (dsd_opts * opts, int speed)
fprintf(stderr, "Audio Out Device: %s\n", opts->audio_out_dev);
}
void
openAudioInDevice (dsd_opts * opts)
void openAudioInDevice(dsd_opts * opts)
{
#ifdef USE_LIBSNDFILE
// get info of device/file
struct stat stat_buf;
if (stat(opts->audio_in_dev, &stat_buf) != 0) {
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.
if(S_ISREG(stat_buf.st_mode))
{ // is this a regular file? then process with libsndfile.
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) {
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
else
{ // this is a device, use old handling
opts->audio_in_type = 0;
#ifdef SOLARIS
sample_info_t aset, aget;

1
dsd/dsd_opts.c
View File

@ -78,5 +78,6 @@ void initOpts(dsd_opts * opts)
opts->delay = 0;
opts->use_cosine_filter = 1;
opts->unmute_encrypted_p25 = 0;
opts->upsample = 0; // by default stay with 8k audio
}

1
dsd/dsd_state.c
View File

@ -132,5 +132,6 @@ void initState(dsd_state * state)
}
state->dsd_running = 0;
state->output_phasor = 0;
}

4
dsd/dsd_state.h
View File

@ -111,9 +111,11 @@ typedef struct
short *output_buffer; //!< Output of decoder single S16LE
int output_offset; //!< producer pointer
short *output_samples; //!< L+R channels S16LE ready for writing to audio FIFO
int output_buffers_size; //!< Size of buffers in number of samples
int output_num_samples; //!< Number of L+R samples available in the above buffer
int output_length; //!< L+R buffer size (fixed)
int output_length; //!< Number of wished samples
int output_finished; //!< 0: not ready, 1: ready
float output_phasor;
pthread_mutex_t input_mutex;
pthread_cond_t input_ready;

42
dsd/dsd_symbol.c
View File

@ -102,22 +102,36 @@ int getSymbol(dsd_opts * opts, dsd_state * state, int have_sync)
{
int i;
state->input_length = 0; // states all samples have been consumed
state->input_offset = 0;
// debug ...
for (i = 0; i < state->output_length; i++)
{
float s = sin(state->output_phasor);
state->output_phasor = fmod(state->output_phasor + (M_PI / 48.0), 2.0 * M_PI);
state->output_samples[2*state->output_offset + 2*i] = s * 16368.0f;
state->output_samples[2*state->output_offset + 2*i+1] = s * 16368.0f;
}
state->output_offset += state->output_length;
state->output_num_samples = state->output_offset;
// ... debug
//ffprintf(stderr, stderr, "dsd::getSymbol: input processing has finished\n");
if (state->output_num_samples > state->output_length)
{
fprintf(stderr, "WARNING: audio buffer over-run! Truncating output\n");
state->output_num_samples = state->output_length;
}
for (i = 0; i < state->output_num_samples; i++)
{
state->output_samples[2*i] = state->output_buffer[i]; // L channel
state->output_samples[2*i+1] = state->output_buffer[i]; // R channel
}
// state->output_num_samples = state->output_offset;
//
// //ffprintf(stderr, stderr, "dsd::getSymbol: input processing has finished\n");
//
// if (state->output_num_samples > state->output_length)
// {
// fprintf(stderr, "WARNING: audio buffer over-run! Truncating output\n");
// state->output_num_samples = state->output_length;
// }
//
// for (i = 0; i < state->output_num_samples; i++)
// {
// state->output_samples[2*i] = state->output_buffer[i]; // L channel
// state->output_samples[2*i+1] = state->output_buffer[i]; // R channel
// }
//
state->output_finished = 1;
}
else

52
dsd/dsd_upsample.c
View File

@ -17,8 +17,56 @@
#include "dsd.h"
void
upsample (dsd_state * state, float invalue)
void upsample(dsd_state * state, float invalue)
{
int i, j, sum;
float *outbuf1, c, d;
outbuf1 = state->audio_out_float_buf_p;
outbuf1--;
c = *outbuf1;
d = invalue;
// basic triangle interpolation
outbuf1++;
*outbuf1 = ((invalue * (float) 0.166) + (c * (float) 0.834));
outbuf1++;
*outbuf1 = ((invalue * (float) 0.332) + (c * (float) 0.668));
outbuf1++;
*outbuf1 = ((invalue * (float) 0.5) + (c * (float) 0.5));
outbuf1++;
*outbuf1 = ((invalue * (float) 0.668) + (c * (float) 0.332));
outbuf1++;
*outbuf1 = ((invalue * (float) 0.834) + (c * (float) 0.166));
outbuf1++;
*outbuf1 = d;
outbuf1++;
if (state->audio_out_idx2 > 24)
{
// smoothing
outbuf1 -= 16;
for (j = 0; j < 4; j++)
{
for (i = 0; i < 6; i++)
{
sum = 0;
outbuf1 -= 2;
sum += *outbuf1;
outbuf1 += 2;
sum += *outbuf1;
outbuf1 += 2;
sum += *outbuf1;
outbuf1 -= 2;
*outbuf1 = (sum / (float) 3);
outbuf1++;
}
outbuf1 -= 8;
}
}
}
void upsample16(dsd_state * state, short invalue)
{
int i, j, sum;

45
plugins/channel/demoddsd/dsddecoder.cpp
View File

@ -18,6 +18,8 @@
#include <QtGlobal>
#include "dsddecoder.h"
#include "dsd_livescanner.h"
#include "audio/audiofifo.h"
DSDDecoder::DSDDecoder()
{
@ -25,6 +27,7 @@ DSDDecoder::DSDDecoder()
initState(&m_dsdParams.state);
m_dsdParams.opts.split = 1;
m_dsdParams.opts.upsample = 1; // force upsampling of audio to 48k
m_dsdParams.opts.playoffset = 0;
m_dsdParams.opts.delay = 0;
m_dsdParams.opts.audio_in_type = 0;
@ -54,6 +57,7 @@ DSDDecoder::DSDDecoder()
m_dsdParams.state.output_buffer = (short *) malloc(1<<18); // Raw output buffer with single S16LE samples @ 8k (max: 128 kS)
m_dsdParams.state.output_offset = 0;
m_dsdParams.state.output_finished = 0;
if (m_dsdParams.state.output_buffer == NULL)
{
@ -61,13 +65,17 @@ DSDDecoder::DSDDecoder()
}
m_dsdParams.state.output_samples = (short *) malloc(1<<19); // Audio output buffer with L+R S16LE samples (max: 128 kS)
m_dsdParams.state.output_length = 1<<19; // the buffer size fixed
m_dsdParams.state.output_buffers_size = 1<<17; // the buffers size in number of samples: 128 kS
if (m_dsdParams.state.output_samples == NULL)
{
qCritical("DSDDecoder::DSDDecoder: Unable to allocate audio L+R buffer.");
}
m_dsdParams.state.output_offset = 0;
m_zeroBuffer = new short[1<<18]; // 128 kS
memset(m_zeroBuffer, 0, sizeof(short) * (1<<18));
m_lastNbSamples = 0;
}
DSDDecoder::~DSDDecoder()
@ -82,14 +90,47 @@ void DSDDecoder::setInBuffer(const short *inBuffer)
}
void DSDDecoder::pushSamples(int nbSamples)
{
if (nbSamples == 0)
{
m_lastNbSamples = 0;
m_dsdParams.state.output_offset = 0; // reset output
}
else
{
m_dsdParams.state.input_offset = 0;
m_dsdParams.state.input_length = nbSamples;
m_dsdParams.state.output_finished = 0;
m_dsdParams.state.output_length = m_lastNbSamples;
m_lastNbSamples = nbSamples;
if (pthread_cond_signal(&m_dsdParams.state.input_ready)) {
printf("DSDDecoder::pushSamples: Unable to signal input ready");
}
}
}
void DSDDecoder::popAudioSamples(AudioFifo *audioFifo, bool audioMute)
{
if (audioMute)
{
uint res = audioFifo->write((const quint8*) m_zeroBuffer, m_dsdParams.state.output_num_samples, 10);
if (res != m_dsdParams.state.output_num_samples) {
qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_dsdParams.state.output_num_samples);
}
}
else if (m_dsdParams.state.output_finished)
{
uint res = audioFifo->write((const quint8*) m_dsdParams.state.output_samples, m_dsdParams.state.output_num_samples, 10);
if (res != m_dsdParams.state.output_num_samples) {
qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_dsdParams.state.output_num_samples);
}
}
m_dsdParams.state.output_finished = 0; // will be done by the next push anyway
}
void DSDDecoder::start()
{
@ -102,6 +143,8 @@ void DSDDecoder::start()
m_dsdParams.state.dsd_running = 0;
}
m_lastNbSamples = 0;
qDebug("DSDDecoder::start: started");
}

6
plugins/channel/demoddsd/dsddecoder.h
View File

@ -20,6 +20,8 @@
#include "dsd.h"
class AudioFifo;
class DSDDecoder
{
public:
@ -27,7 +29,9 @@ public:
~DSDDecoder();
void setInBuffer(const short *inBuffer);
void pushSample(short sample);
void pushSamples(int nbSamples); // Push this amount of samples to the DSD decoder thread
void popAudioSamples(AudioFifo *audioFifo, bool audioMute);
void start();
void stop();
@ -42,6 +46,8 @@ private:
static void* run_dsd(void *arg);
dsd_params m_dsdParams;
short *m_zeroBuffer;
int m_lastNbSamples;
};
#endif /* PLUGINS_CHANNEL_DEMODDSD_DSDDECODER_H_ */

73
plugins/channel/demoddsd/dsddemod.cpp
View File

@ -145,52 +145,51 @@ void DSDDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
m_scopeSampleBuffer.push_back(s);
m_dsdInBuffer[m_dsdInCount++] = sample;
if (m_running.m_audioMute)
{
m_audioBuffer[m_audioBufferFill].l = 0;
m_audioBuffer[m_audioBufferFill].r = 0;
}
else
{
m_audioBuffer[m_audioBufferFill].l = sample;
m_audioBuffer[m_audioBufferFill].r = sample;
}
++m_audioBufferFill;
if (m_audioBufferFill >= m_audioBuffer.size())
{
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
if (res != m_audioBufferFill)
{
qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_audioBufferFill);
}
m_audioBufferFill = 0;
}
// if (m_running.m_audioMute)
// {
// m_audioBuffer[m_audioBufferFill].l = 0;
// m_audioBuffer[m_audioBufferFill].r = 0;
// }
// else
// {
// m_audioBuffer[m_audioBufferFill].l = sample;
// m_audioBuffer[m_audioBufferFill].r = sample;
// }
//
// ++m_audioBufferFill;
//
// if (m_audioBufferFill >= m_audioBuffer.size())
// {
// uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
//
// if (res != m_audioBufferFill)
// {
// qDebug("DSDDemod::feed: %u/%u audio samples written", res, m_audioBufferFill);
// }
//
// m_audioBufferFill = 0;
// }
m_interpolatorDistanceRemain += m_interpolatorDistance;
}
}
}
if (m_audioBufferFill > 0)
{
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
if (res != m_audioBufferFill)
{
qDebug("NFMDemod::feed: %u/%u tail samples written", res, m_audioBufferFill);
}
// if (m_audioBufferFill > 0)
// {
// uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
//
// if (res != m_audioBufferFill)
// {
// qDebug("NFMDemod::feed: %u/%u tail samples written", res, m_audioBufferFill);
// }
//
// m_audioBufferFill = 0;
// }
m_audioBufferFill = 0;
}
if (m_dsdInCount > 0)
{
m_dsdDecoder.popAudioSamples(&m_audioFifo, m_running.m_audioMute);
m_dsdDecoder.pushSamples(m_dsdInCount);
}
if((m_scope != 0) && (m_scopeEnabled))
{

2
sdrbase/audio/audiooutput.cpp
View File

@ -179,7 +179,7 @@ qint64 AudioOutput::readData(char* data, qint64 maxLen)
if (samples != framesPerBuffer)
{
qDebug("AudioOutput::readData: read %d samples vs %d requested", samples, framesPerBuffer);
//qDebug("AudioOutput::readData: read %d samples vs %d requested", samples, framesPerBuffer);
}
for (uint i = 0; i < samples; i++)