diff --git a/audiohandler.cpp b/audiohandler.cpp
index 98d166d..6def10f 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -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()
}
}
+///
+/// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data
+///
+///
+///
+///
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(ulaw_decode[(quint8)packet->data[f+packet->sent]], data + (f * 2 + sentlen));
- else
- qToLittleEndian((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(data + ((f * multiplier) + sentlen));
- if (enc >= 0)
- outdata=ulaw_encode[enc];
- else
- outdata=0x7f & ulaw_encode[-enc];
- }
- else {
- outdata = (quint8)(((qFromLittleEndian((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(in + f);
+ if (enc >= 0)
+ outdata = ulaw_encode[enc];
+ else
+ outdata = 0x7f & ulaw_encode[-enc];
+ }
+ else {
+ outdata = (quint8)(((qFromLittleEndian(in + f) >> 8) ^ 0x80) & 0xff);
+ }
+ packet->dataout[f] = (char)outdata;
+ f++;
+ }
+ }
+ ret.append(packet->dataout);
packet = audioBuffer.erase(packet); // returns next packet
}
else {
diff --git a/audiohandler.h b/audiohandler.h
index 4dae53e..47e436f 100644
--- a/audiohandler.h
+++ b/audiohandler.h
@@ -15,17 +15,20 @@
#include
#include
#include
+#include "resampler/speex_resampler.h"
#include
//#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 audioBuffer;
+
+ SpeexResamplerState* resampler;
+ unsigned int ratioNum;
+ unsigned int ratioDen;
};
#endif // AUDIOHANDLER_H
diff --git a/packettypes.h b/packettypes.h
index 2fcb110..17e270b 100644
--- a/packettypes.h
+++ b/packettypes.h
@@ -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
};
};
};
diff --git a/resampler/arch.h b/resampler/arch.h
new file mode 100644
index 0000000..225d727
--- /dev/null
+++ b/resampler/arch.h
@@ -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
diff --git a/resampler/resample.c b/resampler/resample.c
new file mode 100644
index 0000000..6c58f9f
--- /dev/null
+++ b/resampler/resample.c
@@ -0,0 +1,1240 @@
+/* Copyright (C) 2007-2008 Jean-Marc Valin
+ Copyright (C) 2008 Thorvald Natvig
+
+ File: resample.c
+ Arbitrary resampling code
+
+ 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.
+*/
+
+/*
+ The design goals of this code are:
+ - Very fast algorithm
+ - SIMD-friendly algorithm
+ - Low memory requirement
+ - Good *perceptual* quality (and not best SNR)
+
+ Warning: This resampler is relatively new. Although I think I got rid of
+ all the major bugs and I don't expect the API to change anymore, there
+ may be something I've missed. So use with caution.
+
+ This algorithm is based on this original resampling algorithm:
+ Smith, Julius O. Digital Audio Resampling Home Page
+ Center for Computer Research in Music and Acoustics (CCRMA),
+ Stanford University, 2007.
+ Web published at https://ccrma.stanford.edu/~jos/resample/.
+
+ There is one main difference, though. This resampler uses cubic
+ interpolation instead of linear interpolation in the above paper. This
+ makes the table much smaller and makes it possible to compute that table
+ on a per-stream basis. In turn, being able to tweak the table for each
+ stream makes it possible to both reduce complexity on simple ratios
+ (e.g. 2/3), and get rid of the rounding operations in the inner loop.
+ The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
+*/
+
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#ifdef OUTSIDE_SPEEX
+#include
+static void *speex_alloc(int size) {return calloc(size,1);}
+static void *speex_realloc(void *ptr, int size) {return realloc(ptr, size);}
+static void speex_free(void *ptr) {free(ptr);}
+#ifndef EXPORT
+#define EXPORT
+#endif
+#include "speex_resampler.h"
+#include "arch.h"
+#else /* OUTSIDE_SPEEX */
+
+#include "resampler/speex_resampler.h"
+#include "resampler/arch.h"
+#include "resampler/os_support.h"
+#endif /* OUTSIDE_SPEEX */
+
+#include
+#include
+
+#ifndef M_PI
+#define M_PI 3.14159265358979323846
+#endif
+
+#define IMAX(a,b) ((a) > (b) ? (a) : (b))
+#define IMIN(a,b) ((a) < (b) ? (a) : (b))
+
+#ifndef NULL
+#define NULL 0
+#endif
+
+#ifndef UINT32_MAX
+#define UINT32_MAX 4294967295U
+#endif
+
+#if defined(__SSE__) && !defined(FIXED_POINT)
+#include "resample_sse.h"
+#endif
+
+#ifdef USE_NEON
+#include "resample_neon.h"
+#endif
+
+/* Numer of elements to allocate on the stack */
+#ifdef VAR_ARRAYS
+#define FIXED_STACK_ALLOC 8192
+#else
+#define FIXED_STACK_ALLOC 1024
+#endif
+
+typedef int (*resampler_basic_func)(SpeexResamplerState *, spx_uint32_t , const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
+
+struct SpeexResamplerState_ {
+ spx_uint32_t in_rate;
+ spx_uint32_t out_rate;
+ spx_uint32_t num_rate;
+ spx_uint32_t den_rate;
+
+ int quality;
+ spx_uint32_t nb_channels;
+ spx_uint32_t filt_len;
+ spx_uint32_t mem_alloc_size;
+ spx_uint32_t buffer_size;
+ int int_advance;
+ int frac_advance;
+ float cutoff;
+ spx_uint32_t oversample;
+ int initialised;
+ int started;
+
+ /* These are per-channel */
+ spx_int32_t *last_sample;
+ spx_uint32_t *samp_frac_num;
+ spx_uint32_t *magic_samples;
+
+ spx_word16_t *mem;
+ spx_word16_t *sinc_table;
+ spx_uint32_t sinc_table_length;
+ resampler_basic_func resampler_ptr;
+
+ int in_stride;
+ int out_stride;
+} ;
+
+static const double kaiser12_table[68] = {
+ 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
+ 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
+ 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
+ 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
+ 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
+ 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
+ 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
+ 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
+ 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
+ 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
+ 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
+ 0.00001000, 0.00000000};
+/*
+static const double kaiser12_table[36] = {
+ 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
+ 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
+ 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
+ 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
+ 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
+ 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
+*/
+static const double kaiser10_table[36] = {
+ 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
+ 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
+ 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
+ 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
+ 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
+ 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000};
+
+static const double kaiser8_table[36] = {
+ 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
+ 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
+ 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
+ 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
+ 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
+ 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000};
+
+static const double kaiser6_table[36] = {
+ 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
+ 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
+ 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
+ 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
+ 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
+ 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000};
+
+struct FuncDef {
+ const double *table;
+ int oversample;
+};
+
+static const struct FuncDef kaiser12_funcdef = {kaiser12_table, 64};
+#define KAISER12 (&kaiser12_funcdef)
+static const struct FuncDef kaiser10_funcdef = {kaiser10_table, 32};
+#define KAISER10 (&kaiser10_funcdef)
+static const struct FuncDef kaiser8_funcdef = {kaiser8_table, 32};
+#define KAISER8 (&kaiser8_funcdef)
+static const struct FuncDef kaiser6_funcdef = {kaiser6_table, 32};
+#define KAISER6 (&kaiser6_funcdef)
+
+struct QualityMapping {
+ int base_length;
+ int oversample;
+ float downsample_bandwidth;
+ float upsample_bandwidth;
+ const struct FuncDef *window_func;
+};
+
+
+/* This table maps conversion quality to internal parameters. There are two
+ reasons that explain why the up-sampling bandwidth is larger than the
+ down-sampling bandwidth:
+ 1) When up-sampling, we can assume that the spectrum is already attenuated
+ close to the Nyquist rate (from an A/D or a previous resampling filter)
+ 2) Any aliasing that occurs very close to the Nyquist rate will be masked
+ by the sinusoids/noise just below the Nyquist rate (guaranteed only for
+ up-sampling).
+*/
+static const struct QualityMapping quality_map[11] = {
+ { 8, 4, 0.830f, 0.860f, KAISER6 }, /* Q0 */
+ { 16, 4, 0.850f, 0.880f, KAISER6 }, /* Q1 */
+ { 32, 4, 0.882f, 0.910f, KAISER6 }, /* Q2 */ /* 82.3% cutoff ( ~60 dB stop) 6 */
+ { 48, 8, 0.895f, 0.917f, KAISER8 }, /* Q3 */ /* 84.9% cutoff ( ~80 dB stop) 8 */
+ { 64, 8, 0.921f, 0.940f, KAISER8 }, /* Q4 */ /* 88.7% cutoff ( ~80 dB stop) 8 */
+ { 80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 */ /* 89.1% cutoff (~100 dB stop) 10 */
+ { 96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 */ /* 91.5% cutoff (~100 dB stop) 10 */
+ {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */ /* 93.1% cutoff (~100 dB stop) 10 */
+ {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */ /* 94.5% cutoff (~100 dB stop) 10 */
+ {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 */ /* 95.5% cutoff (~100 dB stop) 10 */
+ {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
+};
+/*8,24,40,56,80,104,128,160,200,256,320*/
+static double compute_func(float x, const struct FuncDef *func)
+{
+ float y, frac;
+ double interp[4];
+ int ind;
+ y = x*func->oversample;
+ ind = (int)floor(y);
+ frac = (y-ind);
+ /* CSE with handle the repeated powers */
+ interp[3] = -0.1666666667*frac + 0.1666666667*(frac*frac*frac);
+ interp[2] = frac + 0.5*(frac*frac) - 0.5*(frac*frac*frac);
+ /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
+ interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac);
+ /* Just to make sure we don't have rounding problems */
+ interp[1] = 1.f-interp[3]-interp[2]-interp[0];
+
+ /*sum = frac*accum[1] + (1-frac)*accum[2];*/
+ return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3];
+}
+
+#if 0
+#include
+int main(int argc, char **argv)
+{
+ int i;
+ for (i=0;i<256;i++)
+ {
+ printf ("%f\n", compute_func(i/256., KAISER12));
+ }
+ return 0;
+}
+#endif
+
+#ifdef FIXED_POINT
+/* The slow way of computing a sinc for the table. Should improve that some day */
+static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func)
+{
+ /*fprintf (stderr, "%f ", x);*/
+ float xx = x * cutoff;
+ if (fabs(x)<1e-6f)
+ return WORD2INT(32768.*cutoff);
+ else if (fabs(x) > .5f*N)
+ return 0;
+ /*FIXME: Can it really be any slower than this? */
+ return WORD2INT(32768.*cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func));
+}
+#else
+/* The slow way of computing a sinc for the table. Should improve that some day */
+static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func)
+{
+ /*fprintf (stderr, "%f ", x);*/
+ float xx = x * cutoff;
+ if (fabs(x)<1e-6)
+ return cutoff;
+ else if (fabs(x) > .5*N)
+ return 0;
+ /*FIXME: Can it really be any slower than this? */
+ return cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func);
+}
+#endif
+
+#ifdef FIXED_POINT
+static void cubic_coef(spx_word16_t x, spx_word16_t interp[4])
+{
+ /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
+ but I know it's MMSE-optimal on a sinc */
+ spx_word16_t x2, x3;
+ x2 = MULT16_16_P15(x, x);
+ x3 = MULT16_16_P15(x, x2);
+ interp[0] = PSHR32(MULT16_16(QCONST16(-0.16667f, 15),x) + MULT16_16(QCONST16(0.16667f, 15),x3),15);
+ interp[1] = EXTRACT16(EXTEND32(x) + SHR32(SUB32(EXTEND32(x2),EXTEND32(x3)),1));
+ interp[3] = PSHR32(MULT16_16(QCONST16(-0.33333f, 15),x) + MULT16_16(QCONST16(.5f,15),x2) - MULT16_16(QCONST16(0.16667f, 15),x3),15);
+ /* Just to make sure we don't have rounding problems */
+ interp[2] = Q15_ONE-interp[0]-interp[1]-interp[3];
+ if (interp[2]<32767)
+ interp[2]+=1;
+}
+#else
+static void cubic_coef(spx_word16_t frac, spx_word16_t interp[4])
+{
+ /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
+ but I know it's MMSE-optimal on a sinc */
+ interp[0] = -0.16667f*frac + 0.16667f*frac*frac*frac;
+ interp[1] = frac + 0.5f*frac*frac - 0.5f*frac*frac*frac;
+ /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
+ interp[3] = -0.33333f*frac + 0.5f*frac*frac - 0.16667f*frac*frac*frac;
+ /* Just to make sure we don't have rounding problems */
+ interp[2] = 1.-interp[0]-interp[1]-interp[3];
+}
+#endif
+
+static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const spx_word16_t *sinc_table = st->sinc_table;
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ spx_word32_t sum;
+
+ while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+ {
+ const spx_word16_t *sinct = & sinc_table[samp_frac_num*N];
+ const spx_word16_t *iptr = & in[last_sample];
+
+#ifndef OVERRIDE_INNER_PRODUCT_SINGLE
+ int j;
+ sum = 0;
+ for(j=0;j= den_rate)
+ {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
+ }
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+
+#ifdef FIXED_POINT
+#else
+/* This is the same as the previous function, except with a double-precision accumulator */
+static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const spx_word16_t *sinc_table = st->sinc_table;
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ double sum;
+
+ while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+ {
+ const spx_word16_t *sinct = & sinc_table[samp_frac_num*N];
+ const spx_word16_t *iptr = & in[last_sample];
+
+#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
+ int j;
+ double accum[4] = {0,0,0,0};
+
+ for(j=0;j= den_rate)
+ {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
+ }
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+#endif
+
+static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ spx_word32_t sum;
+
+ while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+ {
+ const spx_word16_t *iptr = & in[last_sample];
+
+ const int offset = samp_frac_num*st->oversample/st->den_rate;
+#ifdef FIXED_POINT
+ const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
+#else
+ const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
+#endif
+ spx_word16_t interp[4];
+
+
+#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
+ int j;
+ spx_word32_t accum[4] = {0,0,0,0};
+
+ for(j=0;jsinc_table[4+(j+1)*st->oversample-offset-2]);
+ accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
+ accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
+ accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
+ }
+
+ cubic_coef(frac, interp);
+ sum = MULT16_32_Q15(interp[0],SHR32(accum[0], 1)) + MULT16_32_Q15(interp[1],SHR32(accum[1], 1)) + MULT16_32_Q15(interp[2],SHR32(accum[2], 1)) + MULT16_32_Q15(interp[3],SHR32(accum[3], 1));
+ sum = SATURATE32PSHR(sum, 15, 32767);
+#else
+ cubic_coef(frac, interp);
+ sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
+#endif
+
+ out[out_stride * out_sample++] = sum;
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate)
+ {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
+ }
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+
+#ifdef FIXED_POINT
+#else
+/* This is the same as the previous function, except with a double-precision accumulator */
+static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ spx_word32_t sum;
+
+ while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+ {
+ const spx_word16_t *iptr = & in[last_sample];
+
+ const int offset = samp_frac_num*st->oversample/st->den_rate;
+#ifdef FIXED_POINT
+ const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
+#else
+ const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
+#endif
+ spx_word16_t interp[4];
+
+
+#ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
+ int j;
+ double accum[4] = {0,0,0,0};
+
+ for(j=0;jsinc_table[4+(j+1)*st->oversample-offset-2]);
+ accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
+ accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
+ accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
+ }
+
+ cubic_coef(frac, interp);
+ sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
+#else
+ cubic_coef(frac, interp);
+ sum = interpolate_product_double(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
+#endif
+
+ out[out_stride * out_sample++] = PSHR32(sum,15);
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate)
+ {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
+ }
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+#endif
+
+/* This resampler is used to produce zero output in situations where memory
+ for the filter could not be allocated. The expected numbers of input and
+ output samples are still processed so that callers failing to check error
+ codes are not surprised, possibly getting into infinite loops. */
+static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+
+ (void)in;
+ while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+ {
+ out[out_stride * out_sample++] = 0;
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate)
+ {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
+ }
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+
+static int multiply_frac(spx_uint32_t *result, spx_uint32_t value, spx_uint32_t num, spx_uint32_t den)
+{
+ spx_uint32_t major = value / den;
+ spx_uint32_t remain = value % den;
+ /* TODO: Could use 64 bits operation to check for overflow. But only guaranteed in C99+ */
+ if (remain > UINT32_MAX / num || major > UINT32_MAX / num
+ || major * num > UINT32_MAX - remain * num / den)
+ return RESAMPLER_ERR_OVERFLOW;
+ *result = remain * num / den + major * num;
+ return RESAMPLER_ERR_SUCCESS;
+}
+
+static int update_filter(SpeexResamplerState *st)
+{
+ spx_uint32_t old_length = st->filt_len;
+ spx_uint32_t old_alloc_size = st->mem_alloc_size;
+ int use_direct;
+ spx_uint32_t min_sinc_table_length;
+ spx_uint32_t min_alloc_size;
+
+ st->int_advance = st->num_rate/st->den_rate;
+ st->frac_advance = st->num_rate%st->den_rate;
+ st->oversample = quality_map[st->quality].oversample;
+ st->filt_len = quality_map[st->quality].base_length;
+
+ if (st->num_rate > st->den_rate)
+ {
+ /* down-sampling */
+ st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
+ if (multiply_frac(&st->filt_len,st->filt_len,st->num_rate,st->den_rate) != RESAMPLER_ERR_SUCCESS)
+ goto fail;
+ /* Round up to make sure we have a multiple of 8 for SSE */
+ st->filt_len = ((st->filt_len-1)&(~0x7))+8;
+ if (2*st->den_rate < st->num_rate)
+ st->oversample >>= 1;
+ if (4*st->den_rate < st->num_rate)
+ st->oversample >>= 1;
+ if (8*st->den_rate < st->num_rate)
+ st->oversample >>= 1;
+ if (16*st->den_rate < st->num_rate)
+ st->oversample >>= 1;
+ if (st->oversample < 1)
+ st->oversample = 1;
+ } else {
+ /* up-sampling */
+ st->cutoff = quality_map[st->quality].upsample_bandwidth;
+ }
+
+#ifdef RESAMPLE_FULL_SINC_TABLE
+ use_direct = 1;
+ if (INT_MAX/sizeof(spx_word16_t)/st->den_rate < st->filt_len)
+ goto fail;
+#else
+ /* Choose the resampling type that requires the least amount of memory */
+ use_direct = st->filt_len*st->den_rate <= st->filt_len*st->oversample+8
+ && INT_MAX/sizeof(spx_word16_t)/st->den_rate >= st->filt_len;
+#endif
+ if (use_direct)
+ {
+ min_sinc_table_length = st->filt_len*st->den_rate;
+ } else {
+ if ((INT_MAX/sizeof(spx_word16_t)-8)/st->oversample < st->filt_len)
+ goto fail;
+
+ min_sinc_table_length = st->filt_len*st->oversample+8;
+ }
+ if (st->sinc_table_length < min_sinc_table_length)
+ {
+ spx_word16_t *sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,min_sinc_table_length*sizeof(spx_word16_t));
+ if (!sinc_table)
+ goto fail;
+
+ st->sinc_table = sinc_table;
+ st->sinc_table_length = min_sinc_table_length;
+ }
+ if (use_direct)
+ {
+ spx_uint32_t i;
+ for (i=0;iden_rate;i++)
+ {
+ spx_int32_t j;
+ for (j=0;jfilt_len;j++)
+ {
+ st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-(spx_int32_t)st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func);
+ }
+ }
+#ifdef FIXED_POINT
+ st->resampler_ptr = resampler_basic_direct_single;
+#else
+ if (st->quality>8)
+ st->resampler_ptr = resampler_basic_direct_double;
+ else
+ st->resampler_ptr = resampler_basic_direct_single;
+#endif
+ /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/
+ } else {
+ spx_int32_t i;
+ for (i=-4;i<(spx_int32_t)(st->oversample*st->filt_len+4);i++)
+ st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func);
+#ifdef FIXED_POINT
+ st->resampler_ptr = resampler_basic_interpolate_single;
+#else
+ if (st->quality>8)
+ st->resampler_ptr = resampler_basic_interpolate_double;
+ else
+ st->resampler_ptr = resampler_basic_interpolate_single;
+#endif
+ /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/
+ }
+
+ /* Here's the place where we update the filter memory to take into account
+ the change in filter length. It's probably the messiest part of the code
+ due to handling of lots of corner cases. */
+
+ /* Adding buffer_size to filt_len won't overflow here because filt_len
+ could be multiplied by sizeof(spx_word16_t) above. */
+ min_alloc_size = st->filt_len-1 + st->buffer_size;
+ if (min_alloc_size > st->mem_alloc_size)
+ {
+ spx_word16_t *mem;
+ if (INT_MAX/sizeof(spx_word16_t)/st->nb_channels < min_alloc_size)
+ goto fail;
+ else if (!(mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*min_alloc_size * sizeof(*mem))))
+ goto fail;
+
+ st->mem = mem;
+ st->mem_alloc_size = min_alloc_size;
+ }
+ if (!st->started)
+ {
+ spx_uint32_t i;
+ for (i=0;inb_channels*st->mem_alloc_size;i++)
+ st->mem[i] = 0;
+ /*speex_warning("reinit filter");*/
+ } else if (st->filt_len > old_length)
+ {
+ spx_uint32_t i;
+ /* Increase the filter length */
+ /*speex_warning("increase filter size");*/
+ for (i=st->nb_channels;i--;)
+ {
+ spx_uint32_t j;
+ spx_uint32_t olen = old_length;
+ /*if (st->magic_samples[i])*/
+ {
+ /* Try and remove the magic samples as if nothing had happened */
+
+ /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
+ olen = old_length + 2*st->magic_samples[i];
+ for (j=old_length-1+st->magic_samples[i];j--;)
+ st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j];
+ for (j=0;jmagic_samples[i];j++)
+ st->mem[i*st->mem_alloc_size+j] = 0;
+ st->magic_samples[i] = 0;
+ }
+ if (st->filt_len > olen)
+ {
+ /* If the new filter length is still bigger than the "augmented" length */
+ /* Copy data going backward */
+ for (j=0;jmem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*st->mem_alloc_size+(olen-2-j)];
+ /* Then put zeros for lack of anything better */
+ for (;jfilt_len-1;j++)
+ st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = 0;
+ /* Adjust last_sample */
+ st->last_sample[i] += (st->filt_len - olen)/2;
+ } else {
+ /* Put back some of the magic! */
+ st->magic_samples[i] = (olen - st->filt_len)/2;
+ for (j=0;jfilt_len-1+st->magic_samples[i];j++)
+ st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
+ }
+ }
+ } else if (st->filt_len < old_length)
+ {
+ spx_uint32_t i;
+ /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
+ samples so they can be used directly as input the next time(s) */
+ for (i=0;inb_channels;i++)
+ {
+ spx_uint32_t j;
+ spx_uint32_t old_magic = st->magic_samples[i];
+ st->magic_samples[i] = (old_length - st->filt_len)/2;
+ /* We must copy some of the memory that's no longer used */
+ /* Copy data going backward */
+ for (j=0;jfilt_len-1+st->magic_samples[i]+old_magic;j++)
+ st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
+ st->magic_samples[i] += old_magic;
+ }
+ }
+ return RESAMPLER_ERR_SUCCESS;
+
+fail:
+ st->resampler_ptr = resampler_basic_zero;
+ /* st->mem may still contain consumed input samples for the filter.
+ Restore filt_len so that filt_len - 1 still points to the position after
+ the last of these samples. */
+ st->filt_len = old_length;
+ return RESAMPLER_ERR_ALLOC_FAILED;
+}
+
+EXPORT SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
+{
+ return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality, err);
+}
+
+EXPORT 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)
+{
+ SpeexResamplerState *st;
+ int filter_err;
+
+ if (nb_channels == 0 || ratio_num == 0 || ratio_den == 0 || quality > 10 || quality < 0)
+ {
+ if (err)
+ *err = RESAMPLER_ERR_INVALID_ARG;
+ return NULL;
+ }
+ st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState));
+ if (!st)
+ {
+ if (err)
+ *err = RESAMPLER_ERR_ALLOC_FAILED;
+ return NULL;
+ }
+ st->initialised = 0;
+ st->started = 0;
+ st->in_rate = 0;
+ st->out_rate = 0;
+ st->num_rate = 0;
+ st->den_rate = 0;
+ st->quality = -1;
+ st->sinc_table_length = 0;
+ st->mem_alloc_size = 0;
+ st->filt_len = 0;
+ st->mem = 0;
+ st->resampler_ptr = 0;
+
+ st->cutoff = 1.f;
+ st->nb_channels = nb_channels;
+ st->in_stride = 1;
+ st->out_stride = 1;
+
+ st->buffer_size = 160;
+
+ /* Per channel data */
+ if (!(st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t))))
+ goto fail;
+ if (!(st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t))))
+ goto fail;
+ if (!(st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t))))
+ goto fail;
+
+ speex_resampler_set_quality(st, quality);
+ speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);
+
+ filter_err = update_filter(st);
+ if (filter_err == RESAMPLER_ERR_SUCCESS)
+ {
+ st->initialised = 1;
+ } else {
+ speex_resampler_destroy(st);
+ st = NULL;
+ }
+ if (err)
+ *err = filter_err;
+
+ return st;
+
+fail:
+ if (err)
+ *err = RESAMPLER_ERR_ALLOC_FAILED;
+ speex_resampler_destroy(st);
+ return NULL;
+}
+
+EXPORT void speex_resampler_destroy(SpeexResamplerState *st)
+{
+ speex_free(st->mem);
+ speex_free(st->sinc_table);
+ speex_free(st->last_sample);
+ speex_free(st->magic_samples);
+ speex_free(st->samp_frac_num);
+ speex_free(st);
+}
+
+static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t channel_index, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+ int j=0;
+ const int N = st->filt_len;
+ int out_sample = 0;
+ spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
+ spx_uint32_t ilen;
+
+ st->started = 1;
+
+ /* Call the right resampler through the function ptr */
+ out_sample = st->resampler_ptr(st, channel_index, mem, in_len, out, out_len);
+
+ if (st->last_sample[channel_index] < (spx_int32_t)*in_len)
+ *in_len = st->last_sample[channel_index];
+ *out_len = out_sample;
+ st->last_sample[channel_index] -= *in_len;
+
+ ilen = *in_len;
+
+ for(j=0;jmagic_samples[channel_index];
+ spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
+ const int N = st->filt_len;
+
+ speex_resampler_process_native(st, channel_index, &tmp_in_len, *out, &out_len);
+
+ st->magic_samples[channel_index] -= tmp_in_len;
+
+ /* If we couldn't process all "magic" input samples, save the rest for next time */
+ if (st->magic_samples[channel_index])
+ {
+ spx_uint32_t i;
+ for (i=0;imagic_samples[channel_index];i++)
+ mem[N-1+i]=mem[N-1+i+tmp_in_len];
+ }
+ *out += out_len*st->out_stride;
+ return out_len;
+}
+
+#ifdef FIXED_POINT
+EXPORT 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)
+#else
+EXPORT 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)
+#endif
+{
+ int j;
+ spx_uint32_t ilen = *in_len;
+ spx_uint32_t olen = *out_len;
+ spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
+ const int filt_offs = st->filt_len - 1;
+ const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
+ const int istride = st->in_stride;
+
+ if (st->magic_samples[channel_index])
+ olen -= speex_resampler_magic(st, channel_index, &out, olen);
+ if (! st->magic_samples[channel_index]) {
+ while (ilen && olen) {
+ spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
+ spx_uint32_t ochunk = olen;
+
+ if (in) {
+ for(j=0;jout_stride;
+ if (in)
+ in += ichunk * istride;
+ }
+ }
+ *in_len -= ilen;
+ *out_len -= olen;
+ return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
+}
+
+#ifdef FIXED_POINT
+EXPORT 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)
+#else
+EXPORT 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)
+#endif
+{
+ int j;
+ const int istride_save = st->in_stride;
+ const int ostride_save = st->out_stride;
+ spx_uint32_t ilen = *in_len;
+ spx_uint32_t olen = *out_len;
+ spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
+ const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1);
+#ifdef VAR_ARRAYS
+ const unsigned int ylen = (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC;
+ spx_word16_t ystack[ylen];
+#else
+ const unsigned int ylen = FIXED_STACK_ALLOC;
+ spx_word16_t ystack[FIXED_STACK_ALLOC];
+#endif
+
+ st->out_stride = 1;
+
+ while (ilen && olen) {
+ spx_word16_t *y = ystack;
+ spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
+ spx_uint32_t ochunk = (olen > ylen) ? ylen : olen;
+ spx_uint32_t omagic = 0;
+
+ if (st->magic_samples[channel_index]) {
+ omagic = speex_resampler_magic(st, channel_index, &y, ochunk);
+ ochunk -= omagic;
+ olen -= omagic;
+ }
+ if (! st->magic_samples[channel_index]) {
+ if (in) {
+ for(j=0;jfilt_len-1]=WORD2INT(in[j*istride_save]);
+#else
+ x[j+st->filt_len-1]=in[j*istride_save];
+#endif
+ } else {
+ for(j=0;jfilt_len-1]=0;
+ }
+
+ speex_resampler_process_native(st, channel_index, &ichunk, y, &ochunk);
+ } else {
+ ichunk = 0;
+ ochunk = 0;
+ }
+
+ for (j=0;jout_stride = ostride_save;
+ *in_len -= ilen;
+ *out_len -= olen;
+
+ return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
+}
+
+EXPORT int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
+{
+ spx_uint32_t i;
+ int istride_save, ostride_save;
+ spx_uint32_t bak_out_len = *out_len;
+ spx_uint32_t bak_in_len = *in_len;
+ istride_save = st->in_stride;
+ ostride_save = st->out_stride;
+ st->in_stride = st->out_stride = st->nb_channels;
+ for (i=0;inb_channels;i++)
+ {
+ *out_len = bak_out_len;
+ *in_len = bak_in_len;
+ if (in != NULL)
+ speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len);
+ else
+ speex_resampler_process_float(st, i, NULL, in_len, out+i, out_len);
+ }
+ st->in_stride = istride_save;
+ st->out_stride = ostride_save;
+ return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
+}
+
+EXPORT 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)
+{
+ spx_uint32_t i;
+ int istride_save, ostride_save;
+ spx_uint32_t bak_out_len = *out_len;
+ spx_uint32_t bak_in_len = *in_len;
+ istride_save = st->in_stride;
+ ostride_save = st->out_stride;
+ st->in_stride = st->out_stride = st->nb_channels;
+ for (i=0;inb_channels;i++)
+ {
+ *out_len = bak_out_len;
+ *in_len = bak_in_len;
+ if (in != NULL)
+ speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len);
+ else
+ speex_resampler_process_int(st, i, NULL, in_len, out+i, out_len);
+ }
+ st->in_stride = istride_save;
+ st->out_stride = ostride_save;
+ return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
+}
+
+EXPORT int speex_resampler_set_rate(SpeexResamplerState *st, spx_uint32_t in_rate, spx_uint32_t out_rate)
+{
+ return speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate);
+}
+
+EXPORT void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_rate, spx_uint32_t *out_rate)
+{
+ *in_rate = st->in_rate;
+ *out_rate = st->out_rate;
+}
+
+static inline spx_uint32_t compute_gcd(spx_uint32_t a, spx_uint32_t b)
+{
+ while (b != 0)
+ {
+ spx_uint32_t temp = a;
+
+ a = b;
+ b = temp % b;
+ }
+ return a;
+}
+
+EXPORT 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)
+{
+ spx_uint32_t fact;
+ spx_uint32_t old_den;
+ spx_uint32_t i;
+
+ if (ratio_num == 0 || ratio_den == 0)
+ return RESAMPLER_ERR_INVALID_ARG;
+
+ if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den)
+ return RESAMPLER_ERR_SUCCESS;
+
+ old_den = st->den_rate;
+ st->in_rate = in_rate;
+ st->out_rate = out_rate;
+ st->num_rate = ratio_num;
+ st->den_rate = ratio_den;
+
+ fact = compute_gcd(st->num_rate, st->den_rate);
+
+ st->num_rate /= fact;
+ st->den_rate /= fact;
+
+ if (old_den > 0)
+ {
+ for (i=0;inb_channels;i++)
+ {
+ if (multiply_frac(&st->samp_frac_num[i],st->samp_frac_num[i],st->den_rate,old_den) != RESAMPLER_ERR_SUCCESS)
+ return RESAMPLER_ERR_OVERFLOW;
+ /* Safety net */
+ if (st->samp_frac_num[i] >= st->den_rate)
+ st->samp_frac_num[i] = st->den_rate-1;
+ }
+ }
+
+ if (st->initialised)
+ return update_filter(st);
+ return RESAMPLER_ERR_SUCCESS;
+}
+
+EXPORT void speex_resampler_get_ratio(SpeexResamplerState *st, spx_uint32_t *ratio_num, spx_uint32_t *ratio_den)
+{
+ *ratio_num = st->num_rate;
+ *ratio_den = st->den_rate;
+}
+
+EXPORT int speex_resampler_set_quality(SpeexResamplerState *st, int quality)
+{
+ if (quality > 10 || quality < 0)
+ return RESAMPLER_ERR_INVALID_ARG;
+ if (st->quality == quality)
+ return RESAMPLER_ERR_SUCCESS;
+ st->quality = quality;
+ if (st->initialised)
+ return update_filter(st);
+ return RESAMPLER_ERR_SUCCESS;
+}
+
+EXPORT void speex_resampler_get_quality(SpeexResamplerState *st, int *quality)
+{
+ *quality = st->quality;
+}
+
+EXPORT void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride)
+{
+ st->in_stride = stride;
+}
+
+EXPORT void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride)
+{
+ *stride = st->in_stride;
+}
+
+EXPORT void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride)
+{
+ st->out_stride = stride;
+}
+
+EXPORT void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride)
+{
+ *stride = st->out_stride;
+}
+
+EXPORT int speex_resampler_get_input_latency(SpeexResamplerState *st)
+{
+ return st->filt_len / 2;
+}
+
+EXPORT int speex_resampler_get_output_latency(SpeexResamplerState *st)
+{
+ return ((st->filt_len / 2) * st->den_rate + (st->num_rate >> 1)) / st->num_rate;
+}
+
+EXPORT int speex_resampler_skip_zeros(SpeexResamplerState *st)
+{
+ spx_uint32_t i;
+ for (i=0;inb_channels;i++)
+ st->last_sample[i] = st->filt_len/2;
+ return RESAMPLER_ERR_SUCCESS;
+}
+
+EXPORT int speex_resampler_reset_mem(SpeexResamplerState *st)
+{
+ spx_uint32_t i;
+ for (i=0;inb_channels;i++)
+ {
+ st->last_sample[i] = 0;
+ st->magic_samples[i] = 0;
+ st->samp_frac_num[i] = 0;
+ }
+ for (i=0;inb_channels*(st->filt_len-1);i++)
+ st->mem[i] = 0;
+ return RESAMPLER_ERR_SUCCESS;
+}
+
+EXPORT const char *speex_resampler_strerror(int err)
+{
+ switch (err)
+ {
+ case RESAMPLER_ERR_SUCCESS:
+ return "Success.";
+ case RESAMPLER_ERR_ALLOC_FAILED:
+ return "Memory allocation failed.";
+ case RESAMPLER_ERR_BAD_STATE:
+ return "Bad resampler state.";
+ case RESAMPLER_ERR_INVALID_ARG:
+ return "Invalid argument.";
+ case RESAMPLER_ERR_PTR_OVERLAP:
+ return "Input and output buffers overlap.";
+ default:
+ return "Unknown error. Bad error code or strange version mismatch.";
+ }
+}
diff --git a/resampler/resample_sse.h b/resampler/resample_sse.h
new file mode 100644
index 0000000..a0c7a20
--- /dev/null
+++ b/resampler/resample_sse.h
@@ -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
+
+#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
+#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;iport = 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: " <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();
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;
diff --git a/udphandler.h b/udphandler.h
index 00010ef..a511b25 100644
--- a/udphandler.h
+++ b/udphandler.h
@@ -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;
diff --git a/udpserver.cpp b/udpserver.cpp
index f2d7fe6..609739b 100644
--- a/udpserver.cpp
+++ b/udpserver.cpp
@@ -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;
diff --git a/wfmain.cpp b/wfmain.cpp
index ddab534..fcd6fad 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -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
diff --git a/wfview.pro b/wfview.pro
index 96e2251..fc1825a 100644
--- a/wfview.pro
+++ b/wfview.pro
@@ -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 \
diff --git a/wfview.sln b/wfview.sln
index 477a4b4..9148927 100644
--- a/wfview.sln
+++ b/wfview.sln
@@ -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
diff --git a/wfview.vcxproj b/wfview.vcxproj
index cf67190..1a65ae5 100644
--- a/wfview.vcxproj
+++ b/wfview.vcxproj
@@ -152,7 +152,7 @@
Sync
debug\
Disabled
- _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;%(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)
false
MultiThreadedDebugDLL
true
@@ -210,6 +210,7 @@
+
@@ -237,6 +238,7 @@
+
diff --git a/wfview.vcxproj.filters b/wfview.vcxproj.filters
index 158fab7..8114315 100644
--- a/wfview.vcxproj.filters
+++ b/wfview.vcxproj.filters
@@ -108,6 +108,9 @@
Source Files
+
+ Source Files
+
@@ -323,6 +326,10 @@
+
+
+
+
@@ -339,5 +346,8 @@
Header Files
+
+ Header Files
+
\ No newline at end of file