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Add audio resampler

merge-requests/2/head
Phil Taylor 2021-03-09 17:22:16 +00:00
rodzic 998381ac00
commit a0f4a4deeb
15 zmienionych plików z 2171 dodań i 112 usunięć
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223
audiohandler.cpp
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@ -750,9 +750,13 @@ audioHandler::~audioHandler()
if (audioInput != Q_NULLPTR) {
delete audioInput;
}
if (resampler) {
speex_resampler_destroy(resampler);
}
}
bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, QString port)
bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, QString port, quint8 resampleQuality)
{
if (isInitialized) {
return false;
@ -760,7 +764,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
/* Always use 16 bit 48K samples internally*/
format.setSampleSize(16);
format.setChannelCount(channels);
format.setSampleRate(48000);
format.setSampleRate(INTERNAL_SAMPLE_RATE);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::SignedInt);
@ -770,7 +774,27 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
this->isInput = isinput;
this->radioSampleBits = bits;
this->radioSampleRate = samplerate;
this->chunkSize = this->radioSampleBits * 120;
this->radioChannels = channels;
//this->chunkSize = (INTERNAL_SAMPLE_RATE / 25) * (radioSampleBits / 8)/2;
this->chunkSize = 1920*radioChannels;
qDebug(logAudio()) << "Audio chunkSize: " << this->chunkSize;
int resample_error=0;
if (isinput) {
resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error);
}
else
{
resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error);
}
wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
qDebug(logAudio()) << "wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen << " input " << isinput;
qDebug(logAudio()) << "Got audio port name: " << port;
@ -869,8 +893,8 @@ void audioHandler::reinit()
delete audioOutput;
audioOutput = Q_NULLPTR;
audioOutput = new QAudioOutput(deviceInfo, format, this);
audioOutput->setBufferSize((radioSampleRate/25)*(radioSampleBits/8)*2);
connect(audioOutput, SIGNAL(notify()), SLOT(notified()));
audioOutput->setBufferSize((radioSampleRate / 25) * (radioSampleBits / 8) * 2);
connect(audioOutput, SIGNAL(notify()), SLOT(notified()));
connect(audioOutput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
}
@ -929,11 +953,18 @@ void audioHandler::stop()
}
}
/// <summary>
/// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data
/// </summary>
/// <param name="data"></param>
/// <param name="maxlen"></param>
/// <returns></returns>
qint64 audioHandler::readData(char* data, qint64 maxlen)
{
// Calculate output length, always full samples
int sentlen = 0;
//qDebug(logAudio()) << "Looking for: " << maxlen << " bytes";
// We must lock the mutex for the entire time that the buffer may be modified.
@ -941,11 +972,8 @@ qint64 audioHandler::readData(char* data, qint64 maxlen)
// Get next packet from buffer.
if (!audioBuffer.isEmpty())
{
// Output buffer is ALWAYS 16 bit.
int divisor = 16 / radioSampleBits;
auto packet = audioBuffer.begin();
while (packet != audioBuffer.end() && sentlen < maxlen)
{
@ -956,44 +984,18 @@ qint64 audioHandler::readData(char* data, qint64 maxlen)
}
else if (packet->seq == lastSeq+1 || packet->seq <= lastSeq)
{
int send = qMin((int)maxlen-sentlen, packet->dataout.length() - packet->sent);
lastSeq = packet->seq;
//qDebug(logAudio()) << "Packet " << hex << packet->seq << " arrived on time " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms";
// Will this packet fit in the current buffer?
int send = qMin((int)((maxlen/divisor) - (sentlen/divisor)), packet->data.length() - packet->sent);
if (divisor == 2)
{
// Input buffer is 8bit and output buffer is 16bit
for (int f = 0; f < send; f++)
{
if (isUlaw)
qToLittleEndian<qint16>(ulaw_decode[(quint8)packet->data[f+packet->sent]], data + (f * 2 + sentlen));
else
qToLittleEndian<qint16>((qint16)(packet->data[f+packet->sent] << 8) - 32640, data + (f * 2 + sentlen));
}
}
else if (divisor == 1)
{
// 16 bit audio so just copy it in place.
//qDebug(logAudio()) << "Adding packet to buffer:" << (*packet).seq << ": " << (*packet).data.length()-(*packet).sent;
memcpy(data+sentlen, packet->data.constData()+packet->sent, send);
}
else
{
//qDebug(logAudio()) << "Invalid number of bits in audio " << radioSampleBits;
break;
}
memcpy(data + sentlen, packet->dataout.constData() + packet->sent, send);
sentlen = sentlen + (send * divisor);
sentlen = sentlen + send;
if (send == packet->data.length())
if (send == packet->dataout.length())
{
lastSeq = packet->seq;
//qDebug(logAudio()) << "Get next packet";
packet = audioBuffer.erase(packet); // returns next packet
if (maxlen - sentlen == 0)
{
break;
}
}
else if (send == 0)
{
@ -1019,9 +1021,7 @@ qint64 audioHandler::readData(char* data, qint64 maxlen)
qint64 audioHandler::writeData(const char* data, qint64 len)
{
int multiplier = (int)16 / radioSampleBits;
qint64 sentlen = 0;
int tosend = 0;
QMutexLocker locker(&mutex);
audioPacket *current;
@ -1041,36 +1041,15 @@ qint64 audioHandler::writeData(const char* data, qint64 len)
}
current = &audioBuffer.last();
tosend = qMin((int)((len - sentlen)/multiplier), (int)chunkSize-current->sent);
int send = qMin((int)(len - sentlen), (int)chunkSize-current->sent);
if (radioSampleBits == 8) {
int f = 0;
while (f < tosend)
{
quint8 outdata=0;
if (isUlaw) {
qint16 enc = qFromLittleEndian<quint16>(data + ((f * multiplier) + sentlen));
if (enc >= 0)
outdata=ulaw_encode[enc];
else
outdata=0x7f & ulaw_encode[-enc];
}
else {
outdata = (quint8)(((qFromLittleEndian<qint16>((data + ((f * multiplier) + sentlen))) >> 8) ^ 0x80) & 0xff);
}
current->data.append((char)outdata);
f++;
}
}
else if (radioSampleBits == 16)
{
current->data.append(QByteArray::fromRawData(data + sentlen, tosend ));
}
current->datain.append(QByteArray::fromRawData(data + sentlen, send ));
sentlen = sentlen + send;
sentlen = sentlen + (tosend * multiplier);
current->seq = 0; // Not used in TX
current->time = QTime::currentTime();
current->sent = current->data.length();
current->sent = current->datain.length();
if (current->sent == chunkSize)
{
@ -1082,7 +1061,6 @@ qint64 audioHandler::writeData(const char* data, qint64 len)
}
return (sentlen); // Always return the same number as we received
}
@ -1139,10 +1117,59 @@ void audioHandler::stateChanged(QAudio::State state)
void audioHandler::incomingAudio(const audioPacket data)
void audioHandler::incomingAudio(audioPacket data)
{
if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) {
QMutexLocker locker(&mutex);
// Incoming data is 8bits?
if (radioSampleBits == 8)
{
QByteArray inPacket((int)data.datain.length() * 2, (char)0xff);
qint16* in = (qint16*)inPacket.data();
for (int f = 0; f < data.datain.length(); f++)
{
if (isUlaw)
{
in[f] = ulaw_decode[(quint8)data.datain[f]];
}
else
{
// Convert 8-bit sample to 16-bit
in[f] = (qint16)(((quint8)data.datain[f] << 8) - 32640);
}
}
data.datain = inPacket; // Replace incoming data with converted.
}
//qDebug(logAudio()) << "Adding packet to buffer:" << (*packet).seq << ": " << inPacket.length();
/* We now have an array of 16bit samples in the NATIVE samplerate of the radio
If the radio sample rate is below 48000, we need to resample.
*/
if (ratioDen != 1) {
// We need to resample
quint32 outFrames = ((data.datain.length() / 2) * ratioDen) / radioChannels;
quint32 inFrames = (data.datain.length() / 2) / radioChannels;
data.dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size.
int err = 0;
if (this->radioChannels == 1) {
err = wf_resampler_process_int(resampler, 0, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames);
}
else {
err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames);
}
if (err) {
qDebug(logAudio()) << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
}
else {
data.dataout = data.datain;
}
audioBuffer.push_back(data);
// Sort the buffer by seq number. This is important and audio packets may have arrived out-of-order
@ -1193,9 +1220,63 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
packet = audioBuffer.erase(packet); // returns next packet
}
else {
if (packet->data.length() == chunkSize && ret.length() == 0)
if (packet->datain.length() == chunkSize && ret.length() == 0)
{
ret.append(packet->data);
/* We now have an array of samples in the computer native format (48000)
If the radio sample rate is below 48000, we need to resample.
*/
if (ratioNum != 1)
{
// We need to resample (we are STILL 16 bit!)
quint32 outFrames = ((packet->datain.length() / 2) / ratioNum) / radioChannels;
quint32 inFrames = (packet->datain.length() / 2) / radioChannels;
packet->dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size.
int err = 0;
if (this->radioChannels == 1) {
err = wf_resampler_process_int(resampler, 0, (const qint16*)packet->datain.constData(), &inFrames, (qint16*)packet->dataout.data(), &outFrames);
}
else {
err = wf_resampler_process_interleaved_int(resampler, (const qint16*)packet->datain.constData(), &inFrames, (qint16*)packet->dataout.data(), &outFrames);
}
if (err) {
qDebug(logAudio()) << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
//qDebug(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
//qDebug(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length();
if (radioSampleBits == 8)
{
packet->datain = packet->dataout; // Copy packet back to input buffer.
}
}
else if (radioSampleBits == 16 ){
// Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise.
packet->dataout = packet->datain;
}
// Do we need to convert 16-bit to 8-bit?
if (radioSampleBits == 8) {
packet->dataout.resize(packet->datain.length() / 2);
qint16* in = (qint16*)packet->datain.data();
for (int f = 0; f < packet->dataout.length(); f++)
{
quint8 outdata = 0;
if (isUlaw) {
qint16 enc = qFromLittleEndian<quint16>(in + f);
if (enc >= 0)
outdata = ulaw_encode[enc];
else
outdata = 0x7f & ulaw_encode[-enc];
}
else {
outdata = (quint8)(((qFromLittleEndian<qint16>(in + f) >> 8) ^ 0x80) & 0xff);
}
packet->dataout[f] = (char)outdata;
f++;
}
}
ret.append(packet->dataout);
packet = audioBuffer.erase(packet); // returns next packet
}
else {

12
audiohandler.h
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@ -15,17 +15,20 @@
#include <QThread>
#include <QTimer>
#include <QTime>
#include "resampler/speex_resampler.h"
#include <QDebug>
//#define BUFFER_SIZE (32*1024)
#define INTERNAL_SAMPLE_RATE 48000
struct audioPacket {
quint16 seq;
QTime time;
quint16 sent;
QByteArray data;
QByteArray datain;
QByteArray dataout;
};
@ -54,7 +57,7 @@ public:
void getNextAudioChunk(QByteArray &data);
bool isChunkAvailable();
public slots:
bool init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isulaw, const bool isinput, QString port);
bool init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isulaw, const bool isinput, QString port, quint8 resampleQuality);
void incomingAudio(const audioPacket data);
void changeLatency(const quint16 newSize);
@ -88,7 +91,12 @@ private:
QAudioDeviceInfo deviceInfo;
quint16 radioSampleRate;
quint8 radioSampleBits;
quint8 radioChannels;
QVector<audioPacket> audioBuffer;
SpeexResamplerState* resampler;
unsigned int ratioNum;
unsigned int ratioDen;
};
#endif // AUDIOHANDLER_H

3
packettypes.h
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@ -294,7 +294,8 @@ typedef union conninfo_packet {
quint32 civport; // 0x7c
quint32 audioport; // 0x80
quint32 txbuffer; // 0x84
char unusedl[8]; // 0x88
quint8 convert; // 0x88
char unusedl[7]; // 0x89
};
};
};

219
resampler/arch.h 100644
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@ -0,0 +1,219 @@
/* Copyright (C) 2003 Jean-Marc Valin */
/**
@file arch.h
@brief Various architecture definitions Speex
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef ARCH_H
#define ARCH_H
/* A couple test to catch stupid option combinations */
#ifdef FIXED_POINT
#if ((defined (ARM4_ASM)||defined (ARM4_ASM)) && defined(BFIN_ASM)) || (defined (ARM4_ASM)&&defined(ARM5E_ASM))
#error Make up your mind. What CPU do you have?
#endif
#else
#if defined (ARM4_ASM) || defined(ARM5E_ASM) || defined(BFIN_ASM)
#error I suppose you can have a [ARM4/ARM5E/Blackfin] that has float instructions?
#endif
#endif
#ifndef OUTSIDE_SPEEX
#include "speex/speexdsp_types.h"
#endif
#define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */
#define ABS16(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 16-bit value. */
#define MIN16(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 16-bit value. */
#define MAX16(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 16-bit value. */
#define ABS32(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 32-bit value. */
#define MIN32(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 32-bit value. */
#define MAX32(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 32-bit value. */
#ifdef FIXED_POINT
typedef spx_int16_t spx_word16_t;
typedef spx_int32_t spx_word32_t;
typedef spx_word32_t spx_mem_t;
typedef spx_word16_t spx_coef_t;
typedef spx_word16_t spx_lsp_t;
typedef spx_word32_t spx_sig_t;
#define Q15ONE 32767
#define LPC_SCALING 8192
#define SIG_SCALING 16384
#define LSP_SCALING 8192.
#define GAMMA_SCALING 32768.
#define GAIN_SCALING 64
#define GAIN_SCALING_1 0.015625
#define LPC_SHIFT 13
#define LSP_SHIFT 13
#define SIG_SHIFT 14
#define GAIN_SHIFT 6
#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
#define VERY_SMALL 0
#define VERY_LARGE32 ((spx_word32_t)2147483647)
#define VERY_LARGE16 ((spx_word16_t)32767)
#define Q15_ONE ((spx_word16_t)32767)
#ifdef FIXED_DEBUG
#include "fixed_debug.h"
#else
#include "fixed_generic.h"
#ifdef ARM5E_ASM
#include "fixed_arm5e.h"
#elif defined (ARM4_ASM)
#include "fixed_arm4.h"
#elif defined (BFIN_ASM)
#include "fixed_bfin.h"
#endif
#endif
#else
typedef float spx_mem_t;
typedef float spx_coef_t;
typedef float spx_lsp_t;
typedef float spx_sig_t;
typedef float spx_word16_t;
typedef float spx_word32_t;
#define Q15ONE 1.0f
#define LPC_SCALING 1.f
#define SIG_SCALING 1.f
#define LSP_SCALING 1.f
#define GAMMA_SCALING 1.f
#define GAIN_SCALING 1.f
#define GAIN_SCALING_1 1.f
#define VERY_SMALL 1e-15f
#define VERY_LARGE32 1e15f
#define VERY_LARGE16 1e15f
#define Q15_ONE ((spx_word16_t)1.f)
#define QCONST16(x,bits) (x)
#define QCONST32(x,bits) (x)
#define NEG16(x) (-(x))
#define NEG32(x) (-(x))
#define EXTRACT16(x) (x)
#define EXTEND32(x) (x)
#define SHR16(a,shift) (a)
#define SHL16(a,shift) (a)
#define SHR32(a,shift) (a)
#define SHL32(a,shift) (a)
#define PSHR16(a,shift) (a)
#define PSHR32(a,shift) (a)
#define VSHR32(a,shift) (a)
#define SATURATE16(x,a) (x)
#define SATURATE32(x,a) (x)
#define SATURATE32PSHR(x,shift,a) (x)
#define PSHR(a,shift) (a)
#define SHR(a,shift) (a)
#define SHL(a,shift) (a)
#define SATURATE(x,a) (x)
#define ADD16(a,b) ((a)+(b))
#define SUB16(a,b) ((a)-(b))
#define ADD32(a,b) ((a)+(b))
#define SUB32(a,b) ((a)-(b))
#define MULT16_16_16(a,b) ((a)*(b))
#define MULT16_16(a,b) ((spx_word32_t)(a)*(spx_word32_t)(b))
#define MAC16_16(c,a,b) ((c)+(spx_word32_t)(a)*(spx_word32_t)(b))
#define MULT16_32_Q11(a,b) ((a)*(b))
#define MULT16_32_Q13(a,b) ((a)*(b))
#define MULT16_32_Q14(a,b) ((a)*(b))
#define MULT16_32_Q15(a,b) ((a)*(b))
#define MULT16_32_P15(a,b) ((a)*(b))
#define MAC16_32_Q11(c,a,b) ((c)+(a)*(b))
#define MAC16_32_Q15(c,a,b) ((c)+(a)*(b))
#define MAC16_16_Q11(c,a,b) ((c)+(a)*(b))
#define MAC16_16_Q13(c,a,b) ((c)+(a)*(b))
#define MAC16_16_P13(c,a,b) ((c)+(a)*(b))
#define MULT16_16_Q11_32(a,b) ((a)*(b))
#define MULT16_16_Q13(a,b) ((a)*(b))
#define MULT16_16_Q14(a,b) ((a)*(b))
#define MULT16_16_Q15(a,b) ((a)*(b))
#define MULT16_16_P15(a,b) ((a)*(b))
#define MULT16_16_P13(a,b) ((a)*(b))
#define MULT16_16_P14(a,b) ((a)*(b))
#define DIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b))
#define PDIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b))
#define DIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b))
#define PDIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b))
#define WORD2INT(x) ((x) < -32767.5f ? -32768 : \
((x) > 32766.5f ? 32767 : (spx_int16_t)floor(.5 + (x))))
#endif
#if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
/* 2 on TI C5x DSP */
#define BYTES_PER_CHAR 2
#define BITS_PER_CHAR 16
#define LOG2_BITS_PER_CHAR 4
#else
#define BYTES_PER_CHAR 1
#define BITS_PER_CHAR 8
#define LOG2_BITS_PER_CHAR 3
#endif
#ifdef FIXED_DEBUG
extern long long spx_mips;
#endif
#endif

1240
resampler/resample.c 100644
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128
resampler/resample_sse.h 100644
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@ -0,0 +1,128 @@
/* Copyright (C) 2007-2008 Jean-Marc Valin
* Copyright (C) 2008 Thorvald Natvig
*/
/**
@file resample_sse.h
@brief Resampler functions (SSE version)
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include <xmmintrin.h>
#define OVERRIDE_INNER_PRODUCT_SINGLE
static inline float inner_product_single(const float *a, const float *b, unsigned int len)
{
int i;
float ret;
__m128 sum = _mm_setzero_ps();
for (i=0;i<len;i+=8)
{
sum = _mm_add_ps(sum, _mm_mul_ps(_mm_loadu_ps(a+i), _mm_loadu_ps(b+i)));
sum = _mm_add_ps(sum, _mm_mul_ps(_mm_loadu_ps(a+i+4), _mm_loadu_ps(b+i+4)));
}
sum = _mm_add_ps(sum, _mm_movehl_ps(sum, sum));
sum = _mm_add_ss(sum, _mm_shuffle_ps(sum, sum, 0x55));
_mm_store_ss(&ret, sum);
return ret;
}
#define OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
static inline float interpolate_product_single(const float *a, const float *b, unsigned int len, const spx_uint32_t oversample, float *frac) {
int i;
float ret;
__m128 sum = _mm_setzero_ps();
__m128 f = _mm_loadu_ps(frac);
for(i=0;i<len;i+=2)
{
sum = _mm_add_ps(sum, _mm_mul_ps(_mm_load1_ps(a+i), _mm_loadu_ps(b+i*oversample)));
sum = _mm_add_ps(sum, _mm_mul_ps(_mm_load1_ps(a+i+1), _mm_loadu_ps(b+(i+1)*oversample)));
}
sum = _mm_mul_ps(f, sum);
sum = _mm_add_ps(sum, _mm_movehl_ps(sum, sum));
sum = _mm_add_ss(sum, _mm_shuffle_ps(sum, sum, 0x55));
_mm_store_ss(&ret, sum);
return ret;
}
#ifdef __SSE2__
#include <emmintrin.h>
#define OVERRIDE_INNER_PRODUCT_DOUBLE
static inline double inner_product_double(const float *a, const float *b, unsigned int len)
{
int i;
double ret;
__m128d sum = _mm_setzero_pd();
__m128 t;
for (i=0;i<len;i+=8)
{
t = _mm_mul_ps(_mm_loadu_ps(a+i), _mm_loadu_ps(b+i));
sum = _mm_add_pd(sum, _mm_cvtps_pd(t));
sum = _mm_add_pd(sum, _mm_cvtps_pd(_mm_movehl_ps(t, t)));
t = _mm_mul_ps(_mm_loadu_ps(a+i+4), _mm_loadu_ps(b+i+4));
sum = _mm_add_pd(sum, _mm_cvtps_pd(t));
sum = _mm_add_pd(sum, _mm_cvtps_pd(_mm_movehl_ps(t, t)));
}
sum = _mm_add_sd(sum, _mm_unpackhi_pd(sum, sum));
_mm_store_sd(&ret, sum);
return ret;
}
#define OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
static inline double interpolate_product_double(const float *a, const float *b, unsigned int len, const spx_uint32_t oversample, float *frac) {
int i;
double ret;
__m128d sum;
__m128d sum1 = _mm_setzero_pd();
__m128d sum2 = _mm_setzero_pd();
__m128 f = _mm_loadu_ps(frac);
__m128d f1 = _mm_cvtps_pd(f);
__m128d f2 = _mm_cvtps_pd(_mm_movehl_ps(f,f));
__m128 t;
for(i=0;i<len;i+=2)
{
t = _mm_mul_ps(_mm_load1_ps(a+i), _mm_loadu_ps(b+i*oversample));
sum1 = _mm_add_pd(sum1, _mm_cvtps_pd(t));
sum2 = _mm_add_pd(sum2, _mm_cvtps_pd(_mm_movehl_ps(t, t)));
t = _mm_mul_ps(_mm_load1_ps(a+i+1), _mm_loadu_ps(b+(i+1)*oversample));
sum1 = _mm_add_pd(sum1, _mm_cvtps_pd(t));
sum2 = _mm_add_pd(sum2, _mm_cvtps_pd(_mm_movehl_ps(t, t)));
}
sum1 = _mm_mul_pd(f1, sum1);
sum2 = _mm_mul_pd(f2, sum2);
sum = _mm_add_pd(sum1, sum2);
sum = _mm_add_sd(sum, _mm_unpackhi_pd(sum, sum));
_mm_store_sd(&ret, sum);
return ret;
}
#endif

344
resampler/speex_resampler.h 100644
Wyświetl plik

@ -0,0 +1,344 @@
/* Copyright (C) 2007 Jean-Marc Valin
File: speex_resampler.h
Resampling code
The design goals of this code are:
- Very fast algorithm
- Low memory requirement
- Good *perceptual* quality (and not best SNR)
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
1. Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
3. The name of the author may not be used to endorse or promote products
derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_RESAMPLER_H
#define SPEEX_RESAMPLER_H
#ifdef OUTSIDE_SPEEX
/********* WARNING: MENTAL SANITY ENDS HERE *************/
/* If the resampler is defined outside of Speex, we change the symbol names so that
there won't be any clash if linking with Speex later on. */
/* #define RANDOM_PREFIX your software name here */
#ifndef RANDOM_PREFIX
#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes"
#endif
#define CAT_PREFIX2(a,b) a ## b
#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b)
#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init)
#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac)
#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy)
#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float)
#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int)
#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float)
#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int)
#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate)
#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate)
#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac)
#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio)
#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality)
#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality)
#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride)
#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride)
#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride)
#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride)
#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency)
#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency)
#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros)
#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem)
#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror)
#define spx_int16_t short
#define spx_int32_t int
#define spx_uint16_t unsigned short
#define spx_uint32_t unsigned int
#define speex_assert(cond)
#else /* OUTSIDE_SPEEX */
#include "speexdsp_types.h"
#endif /* OUTSIDE_SPEEX */
#ifdef __cplusplus
extern "C" {
#endif
#define SPEEX_RESAMPLER_QUALITY_MAX 10
#define SPEEX_RESAMPLER_QUALITY_MIN 0
#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
#define SPEEX_RESAMPLER_QUALITY_VOIP 3
#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
enum {
RESAMPLER_ERR_SUCCESS = 0,
RESAMPLER_ERR_ALLOC_FAILED = 1,
RESAMPLER_ERR_BAD_STATE = 2,
RESAMPLER_ERR_INVALID_ARG = 3,
RESAMPLER_ERR_PTR_OVERLAP = 4,
RESAMPLER_ERR_OVERFLOW = 5,
RESAMPLER_ERR_MAX_ERROR
};
struct SpeexResamplerState_;
typedef struct SpeexResamplerState_ SpeexResamplerState;
/** Create a new resampler with integer input and output rates.
* @param nb_channels Number of channels to be processed
* @param in_rate Input sampling rate (integer number of Hz).
* @param out_rate Output sampling rate (integer number of Hz).
* @param quality Resampling quality between 0 and 10, where 0 has poor quality
* and 10 has very high quality.
* @return Newly created resampler state
* @retval NULL Error: not enough memory
*/
SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels,
spx_uint32_t in_rate,
spx_uint32_t out_rate,
int quality,
int *err);
/** Create a new resampler with fractional input/output rates. The sampling
* rate ratio is an arbitrary rational number with both the numerator and
* denominator being 32-bit integers.
* @param nb_channels Number of channels to be processed
* @param ratio_num Numerator of the sampling rate ratio
* @param ratio_den Denominator of the sampling rate ratio
* @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
* @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
* @param quality Resampling quality between 0 and 10, where 0 has poor quality
* and 10 has very high quality.
* @return Newly created resampler state
* @retval NULL Error: not enough memory
*/
SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
spx_uint32_t ratio_num,
spx_uint32_t ratio_den,
spx_uint32_t in_rate,
spx_uint32_t out_rate,
int quality,
int *err);
/** Destroy a resampler state.
* @param st Resampler state
*/
void speex_resampler_destroy(SpeexResamplerState *st);
/** Resample a float array. The input and output buffers must *not* overlap.
* @param st Resampler state
* @param channel_index Index of the channel to process for the multi-channel
* base (0 otherwise)
* @param in Input buffer
* @param in_len Number of input samples in the input buffer. Returns the
* number of samples processed
* @param out Output buffer
* @param out_len Size of the output buffer. Returns the number of samples written
*/
int speex_resampler_process_float(SpeexResamplerState *st,
spx_uint32_t channel_index,
const float *in,
spx_uint32_t *in_len,
float *out,
spx_uint32_t *out_len);
/** Resample an int array. The input and output buffers must *not* overlap.
* @param st Resampler state
* @param channel_index Index of the channel to process for the multi-channel
* base (0 otherwise)
* @param in Input buffer
* @param in_len Number of input samples in the input buffer. Returns the number
* of samples processed
* @param out Output buffer
* @param out_len Size of the output buffer. Returns the number of samples written
*/
int speex_resampler_process_int(SpeexResamplerState *st,
spx_uint32_t channel_index,
const spx_int16_t *in,
spx_uint32_t *in_len,
spx_int16_t *out,
spx_uint32_t *out_len);
/** Resample an interleaved float array. The input and output buffers must *not* overlap.
* @param st Resampler state
* @param in Input buffer
* @param in_len Number of input samples in the input buffer. Returns the number
* of samples processed. This is all per-channel.
* @param out Output buffer
* @param out_len Size of the output buffer. Returns the number of samples written.
* This is all per-channel.
*/
int speex_resampler_process_interleaved_float(SpeexResamplerState *st,
const float *in,
spx_uint32_t *in_len,
float *out,
spx_uint32_t *out_len);
/** Resample an interleaved int array. The input and output buffers must *not* overlap.
* @param st Resampler state
* @param in Input buffer
* @param in_len Number of input samples in the input buffer. Returns the number
* of samples processed. This is all per-channel.
* @param out Output buffer
* @param out_len Size of the output buffer. Returns the number of samples written.
* This is all per-channel.
*/
int speex_resampler_process_interleaved_int(SpeexResamplerState *st,
const spx_int16_t *in,
spx_uint32_t *in_len,
spx_int16_t *out,
spx_uint32_t *out_len);
/** Set (change) the input/output sampling rates (integer value).
* @param st Resampler state
* @param in_rate Input sampling rate (integer number of Hz).
* @param out_rate Output sampling rate (integer number of Hz).
*/
int speex_resampler_set_rate(SpeexResamplerState *st,
spx_uint32_t in_rate,
spx_uint32_t out_rate);
/** Get the current input/output sampling rates (integer value).
* @param st Resampler state
* @param in_rate Input sampling rate (integer number of Hz) copied.
* @param out_rate Output sampling rate (integer number of Hz) copied.
*/
void speex_resampler_get_rate(SpeexResamplerState *st,
spx_uint32_t *in_rate,
spx_uint32_t *out_rate);
/** Set (change) the input/output sampling rates and resampling ratio
* (fractional values in Hz supported).
* @param st Resampler state
* @param ratio_num Numerator of the sampling rate ratio
* @param ratio_den Denominator of the sampling rate ratio
* @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
* @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
*/
int speex_resampler_set_rate_frac(SpeexResamplerState *st,
spx_uint32_t ratio_num,
spx_uint32_t ratio_den,
spx_uint32_t in_rate,
spx_uint32_t out_rate);
/** Get the current resampling ratio. This will be reduced to the least
* common denominator.
* @param st Resampler state
* @param ratio_num Numerator of the sampling rate ratio copied
* @param ratio_den Denominator of the sampling rate ratio copied
*/
void speex_resampler_get_ratio(SpeexResamplerState *st,
spx_uint32_t *ratio_num,
spx_uint32_t *ratio_den);
/** Set (change) the conversion quality.
* @param st Resampler state
* @param quality Resampling quality between 0 and 10, where 0 has poor
* quality and 10 has very high quality.
*/
int speex_resampler_set_quality(SpeexResamplerState *st,
int quality);
/** Get the conversion quality.
* @param st Resampler state
* @param quality Resampling quality between 0 and 10, where 0 has poor
* quality and 10 has very high quality.
*/
void speex_resampler_get_quality(SpeexResamplerState *st,
int *quality);
/** Set (change) the input stride.
* @param st Resampler state
* @param stride Input stride
*/
void speex_resampler_set_input_stride(SpeexResamplerState *st,
spx_uint32_t stride);
/** Get the input stride.
* @param st Resampler state
* @param stride Input stride copied
*/
void speex_resampler_get_input_stride(SpeexResamplerState *st,
spx_uint32_t *stride);
/** Set (change) the output stride.
* @param st Resampler state
* @param stride Output stride
*/
void speex_resampler_set_output_stride(SpeexResamplerState *st,
spx_uint32_t stride);
/** Get the output stride.
* @param st Resampler state copied
* @param stride Output stride
*/
void speex_resampler_get_output_stride(SpeexResamplerState *st,
spx_uint32_t *stride);
/** Get the latency introduced by the resampler measured in input samples.
* @param st Resampler state
*/
int speex_resampler_get_input_latency(SpeexResamplerState *st);
/** Get the latency introduced by the resampler measured in output samples.
* @param st Resampler state
*/
int speex_resampler_get_output_latency(SpeexResamplerState *st);
/** Make sure that the first samples to go out of the resamplers don't have
* leading zeros. This is only useful before starting to use a newly created
* resampler. It is recommended to use that when resampling an audio file, as
* it will generate a file with the same length. For real-time processing,
* it is probably easier not to use this call (so that the output duration
* is the same for the first frame).
* @param st Resampler state
*/
int speex_resampler_skip_zeros(SpeexResamplerState *st);
/** Reset a resampler so a new (unrelated) stream can be processed.
* @param st Resampler state
*/
int speex_resampler_reset_mem(SpeexResamplerState *st);
/** Returns the English meaning for an error code
* @param err Error code
* @return English string
*/
const char *speex_resampler_strerror(int err);
#ifdef __cplusplus
}
#endif
#endif

57
udphandler.cpp
Wyświetl plik

@ -15,7 +15,8 @@ udpHandler::udpHandler(udpPreferences prefs) :
rxCodec(prefs.audioRXCodec),
txCodec(prefs.audioTXCodec),
audioInputPort(prefs.audioInput),
audioOutputPort(prefs.audioOutput)
audioOutputPort(prefs.audioOutput),
resampleQuality(prefs.resampleQuality)
{
this->port = this->controlPort;
@ -150,6 +151,8 @@ void udpHandler::dataReceived()
control_packet_t in = (control_packet_t)r.constData();
if (in->type == 0x04) {
// If timer is active, stop it as they are obviously there!
qDebug(logUdp()) << this->metaObject()->className() << ": Received I am here from: " <<datagram.senderAddress();
if (areYouThereTimer->isActive()) {
// send ping packets every second
areYouThereTimer->stop();
@ -305,7 +308,7 @@ void udpHandler::dataReceived()
}
else {
civ = new udpCivData(localIP, radioIP, civPort);
audio = new udpAudio(localIP, radioIP, audioPort, rxLatency, txLatency, rxSampleRate, rxCodec, txSampleRate, txCodec, audioOutputPort, audioInputPort);
audio = new udpAudio(localIP, radioIP, audioPort, rxLatency, txLatency, rxSampleRate, rxCodec, txSampleRate, txCodec, audioOutputPort, audioInputPort,resampleQuality);
QObject::connect(civ, SIGNAL(receive(QByteArray)), this, SLOT(receiveFromCivStream(QByteArray)));
QObject::connect(audio, SIGNAL(haveAudioData(audioPacket)), this, SLOT(receiveAudioData(audioPacket)));
@ -399,7 +402,7 @@ void udpHandler::sendRequestStream()
p.civport = qToBigEndian((quint32)civPort);
p.audioport = qToBigEndian((quint32)audioPort);
p.txbuffer = qToBigEndian((quint32)txLatency);
p.convert = 1;
sendTrackedPacket(QByteArray::fromRawData((const char*)p.packet, sizeof(p)));
return;
}
@ -642,7 +645,7 @@ void udpCivData::dataReceived()
// Audio stream
udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, QString outputPort, QString inputPort)
udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, QString outputPort, QString inputPort,quint8 resampleQuality)
{
qDebug(logUdp()) << "Starting udpAudio";
this->localIP = local;
@ -685,7 +688,7 @@ udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint
rxaudio->moveToThread(rxAudioThread);
connect(this, SIGNAL(setupRxAudio(quint8, quint8, quint16, quint16, bool, bool, QString)), rxaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool,QString)));
connect(this, SIGNAL(setupRxAudio(quint8, quint8, quint16, quint16, bool, bool, QString, quint8)), rxaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool,QString, quint8)));
qRegisterMetaType<audioPacket>();
connect(this, SIGNAL(haveAudioData(audioPacket)), rxaudio, SLOT(incomingAudio(audioPacket)));
@ -704,7 +707,7 @@ udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint
txaudio->moveToThread(txAudioThread);
connect(this, SIGNAL(setupTxAudio(quint8, quint8, quint16, quint16, bool, bool,QString)), txaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool,QString)));
connect(this, SIGNAL(setupTxAudio(quint8, quint8, quint16, quint16, bool, bool,QString,quint8)), txaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool,QString,quint8)));
connect(txAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater()));
rxAudioThread->start();
@ -717,8 +720,8 @@ udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint
connect(pingTimer, &QTimer::timeout, this, &udpBase::sendPing);
pingTimer->start(PING_PERIOD); // send ping packets every 100ms
emit setupTxAudio(txNumSamples, txChannelCount, txSampleRate, txLatency, txIsUlawCodec, true, inputPort);
emit setupRxAudio(rxNumSamples, rxChannelCount, rxSampleRate, txLatency, rxIsUlawCodec, false, outputPort);
emit setupTxAudio(txNumSamples, txChannelCount, txSampleRate, txLatency, txIsUlawCodec, true, inputPort,resampleQuality);
emit setupRxAudio(rxNumSamples, rxChannelCount, rxSampleRate, txLatency, rxIsUlawCodec, false, outputPort,resampleQuality);
watchdogTimer = new QTimer();
connect(watchdogTimer, &QTimer::timeout, this, &udpAudio::watchdog);
@ -788,7 +791,12 @@ void udpAudio::sendTxAudio()
p.len = sizeof(p) + partial.length();
p.sentid = myId;
p.rcvdid = remoteId;
p.ident = 0x0080; // TX audio is always this?
if (partial.length() == 0xa0) {
p.ident = 0x9781;
}
else {
p.ident = 0x0080; // TX audio is always this?
}
p.datalen = (quint16)qToBigEndian((quint16)partial.length());
p.sendseq = (quint16)qToBigEndian((quint16)sendAudioSeq); // THIS IS BIG ENDIAN!
QByteArray tx = QByteArray::fromRawData((const char*)p.packet, sizeof(p));
@ -838,23 +846,18 @@ void udpAudio::dataReceived()
*/
control_packet_t in = (control_packet_t)r.constData();
if (in->type != 0x01) {
if (r.mid(0, 2) == QByteArrayLiteral("\x6c\x05") ||
r.mid(0, 2) == QByteArrayLiteral("\x44\x02") ||
r.mid(0, 2) == QByteArrayLiteral("\xd8\x03") ||
r.mid(0, 2) == QByteArrayLiteral("\x70\x04"))
{
lastReceived = QTime::currentTime();
audioPacket tempAudio;
tempAudio.seq = in->seq;
tempAudio.time = lastReceived;
tempAudio.sent = 0;
tempAudio.data = r.mid(0x18);
// Prefer signal/slot to forward audio as it is thread/safe
// Need to do more testing but latency appears fine.
emit haveAudioData(tempAudio);
//rxaudio->incomingAudio(tempAudio);
}
if (in->type != 0x01 && in->len >= 0xAC) {
// 0xac is the smallest possible audio packet.
lastReceived = QTime::currentTime();
audioPacket tempAudio;
tempAudio.seq = in->seq;
tempAudio.time = lastReceived;
tempAudio.sent = 0;
tempAudio.datain = r.mid(0x18);
// Prefer signal/slot to forward audio as it is thread/safe
// Need to do more testing but latency appears fine.
emit haveAudioData(tempAudio);
//rxaudio->incomingAudio(tempAudio);
}
break;
}
@ -956,7 +959,7 @@ void udpBase::dataReceived(QByteArray r)
}
}
if (in->type == 0x04) {
qDebug(logUdp()) << this->metaObject()->className() << ": Received I am here";
qDebug(logUdp()) << this->metaObject()->className() << ": Received I am here ";
areYouThereCounter = 0;
// I don't think that we will ever receive an "I am here" other than in response to "Are you there?"
remoteId = in->sentid;

9
udphandler.h
Wyświetl plik

@ -48,6 +48,7 @@ struct udpPreferences {
quint8 audioRXCodec;
quint16 audioTXSampleRate;
quint8 audioTXCodec;
quint8 resampleQuality;
};
void passcode(QString in, QByteArray& out);
@ -166,14 +167,14 @@ class udpAudio : public udpBase
Q_OBJECT
public:
udpAudio(QHostAddress local, QHostAddress ip, quint16 aport, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, QString outputPort, QString inputPort);
udpAudio(QHostAddress local, QHostAddress ip, quint16 aport, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, QString outputPort, QString inputPort,quint8 resampleQuality);
~udpAudio();
signals:
void haveAudioData(audioPacket data);
void setupTxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port);
void setupRxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port);
void setupTxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port,quint8 resampleQuality);
void setupRxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port,quint8 resampleQuality);
void haveChangeLatency(quint16 value);
@ -276,6 +277,8 @@ private:
QString audioInputPort;
QString audioOutputPort;
quint8 resampleQuality;
quint16 reauthInterval = 60000;
QString devName;

6
udpserver.cpp
Wyświetl plik

@ -1062,14 +1062,14 @@ void udpServer::receiveAudioData(const audioPacket &d)
if (client != Q_NULLPTR && client->connected) {
audio_packet p;
memset(p.packet, 0x0, sizeof(p)); // We can't be sure it is initialized with 0x00!
p.len = sizeof(p) + d.data.length();
p.len = sizeof(p) + d.datain.length();
p.sentid = client->myId;
p.rcvdid = client->remoteId;
p.ident = 0x0080; // audio is always this?
p.datalen = (quint16)qToBigEndian((quint16)d.data.length());
p.datalen = (quint16)qToBigEndian((quint16)d.datain.length());
p.sendseq = (quint16)qToBigEndian((quint16)client->sendAudioSeq); // THIS IS BIG ENDIAN!
QByteArray t = QByteArray::fromRawData((const char*)p.packet, sizeof(p));
t.append(d.data);
t.append(d.datain);
QMutexLocker locker(&mutex);
client->txSeqBuf.append(SEQBUFENTRY());
client->txSeqBuf.last().seqNum = p.seq;

6
wfmain.cpp
Wyświetl plik

@ -761,8 +761,7 @@ void wfmain::setDefPrefs()
udpDefPrefs.audioRXCodec = 4;
udpDefPrefs.audioTXSampleRate = 48000;
udpDefPrefs.audioTXCodec = 4;
udpDefPrefs.resampleQuality = 4;
}
void wfmain::loadSettings()
@ -878,6 +877,8 @@ void wfmain::loadSettings()
ui->audioInputCombo->setCurrentIndex(audioInputIndex);
}
udpPrefs.resampleQuality = settings.value("ResampleQuality", udpDefPrefs.resampleQuality).toInt();
settings.endGroup();
settings.beginGroup("Server");
@ -985,6 +986,7 @@ void wfmain::saveSettings()
settings.setValue("AudioTXCodec", udpPrefs.audioTXCodec);
settings.setValue("AudioOutput", udpPrefs.audioOutput);
settings.setValue("AudioInput", udpPrefs.audioInput);
settings.setValue("ResampleQuality", udpPrefs.resampleQuality);
settings.endGroup();
// Memory channels

12
wfview.pro
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@ -31,6 +31,10 @@ QMAKE_LFLAGS += -O2 -march=native -s
DEFINES += QT_DEPRECATED_WARNINGS
DEFINES += QCUSTOMPLOT_COMPILE_LIBRARY
# These defines are used for the resampler
DEFINES += OUTSIDE_SPEEX
DEFINES += RANDOM_PREFIX=wf
linux:DEFINES += HOST=\\\"`hostname`\\\" UNAME=\\\"`whoami`\\\"
linux:DEFINES += GITSHORT="\\\"$(shell git -C $$PWD rev-parse --short HEAD)\\\""
@ -88,7 +92,8 @@ SOURCES += main.cpp\
udpserver.cpp \
meter.cpp \
qledlabel.cpp \
pttyhandler.cpp
pttyhandler.cpp \
resampler/resample.cpp
HEADERS += wfmain.h \
commhandler.h \
@ -105,7 +110,10 @@ HEADERS += wfmain.h \
packettypes.h \
meter.h \
qledlabel.h \
pttyhandler.h
pttyhandler.h \
resampler/speex_resampler.h \
resampler/arch.h \
resampler/resample_sse.h
FORMS += wfmain.ui \

10
wfview.sln
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@ -8,13 +8,23 @@ EndProject
Global
GlobalSection(SolutionConfigurationPlatforms) = preSolution
Debug|x64 = Debug|x64
Debug|x86 = Debug|x86
Release|x64 = Release|x64
Release|x86 = Release|x86
Template|x64 = Template|x64
Template|x86 = Template|x86
EndGlobalSection
GlobalSection(ProjectConfigurationPlatforms) = postSolution
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Debug|x64.ActiveCfg = Debug|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Debug|x64.Build.0 = Debug|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Debug|x86.ActiveCfg = Debug|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Release|x64.ActiveCfg = Release|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Release|x64.Build.0 = Release|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Release|x86.ActiveCfg = Release|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Template|x64.ActiveCfg = Release|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Template|x64.Build.0 = Release|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Template|x86.ActiveCfg = Release|x64
{326108AD-FA9D-3AAF-8D3E-062C4DDC34E2}.Template|x86.Build.0 = Release|x64
EndGlobalSection
GlobalSection(SolutionProperties) = preSolution
HideSolutionNode = FALSE

4
wfview.vcxproj
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@ -152,7 +152,7 @@
<ExceptionHandling>Sync</ExceptionHandling>
<ObjectFileName>debug\</ObjectFileName>
<Optimization>Disabled</Optimization>
<PreprocessorDefinitions>_WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;%(PreprocessorDefinitions)</PreprocessorDefinitions>
<PreprocessorDefinitions>_WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;%(PreprocessorDefinitions)</PreprocessorDefinitions>
<PreprocessToFile>false</PreprocessToFile>
<RuntimeLibrary>MultiThreadedDebugDLL</RuntimeLibrary>
<SuppressStartupBanner>true</SuppressStartupBanner>
@ -210,6 +210,7 @@
<ClCompile Include="main.cpp" />
<ClCompile Include="c:\qcustomplot\qcustomplot.cpp" />
<ClCompile Include="meter.cpp" />
<ClCompile Include="resampler\resample.c" />
<ClCompile Include="rigcommander.cpp" />
<ClCompile Include="rigidentities.cpp" />
<ClCompile Include="audiohandler.cpp" />
@ -237,6 +238,7 @@
<QtMoc Include="meter.h" />
<ClInclude Include="packettypes.h" />
<QtMoc Include="pttyhandler.h" />
<ClInclude Include="resampler\speex_resampler.h" />
<ClInclude Include="rigidentities.h" />
<QtMoc Include="udphandler.h">
</QtMoc>

10
wfview.vcxproj.filters
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@ -108,6 +108,9 @@
<ClCompile Include="pttyhandler.cpp">
<Filter>Source Files</Filter>
</ClCompile>
<ClCompile Include="resampler\resample.c">
<Filter>Source Files</Filter>
</ClCompile>
</ItemGroup>
<ItemGroup>
<QtMoc Include="commhandler.h">
@ -323,6 +326,10 @@
<None Include="resources\wfview.png" />
<None Include="resources\wfview.png" />
<None Include="resources\wfview.png" />
<None Include="resources\wfview.png" />
<None Include="resources\wfview.png" />
<None Include="resources\wfview.png" />
<None Include="resources\wfview.png" />
</ItemGroup>
<ItemGroup>
<None Include="resources\install.sh">
@ -339,5 +346,8 @@
<ClInclude Include="packettypes.h">
<Filter>Header Files</Filter>
</ClInclude>
<ClInclude Include="resampler\speex_resampler.h">
<Filter>Header Files</Filter>
</ClInclude>
</ItemGroup>
</Project>