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Takes another try at echo cancellation, uses a circular buffer to improve the latency (a teeny-weeny bit).

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cusax-fix
Lyubomir Marinov 16 years ago
parent 6ae5819685
commit e23a9b617b
12 changed files with 452 additions and 1381 deletions
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lib/native/linux-64/libjportaudio.so
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src/native/portaudio/AudioQualityImprovement.c
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@ -12,118 +12,90 @@
#include <stdlib.h>
#include <string.h>
/**
* The minimum number of frames by which the playback is to be delayed before it
* is accessible to the echo cancellation/capture.
*/
#define MIN_PLAY_DELAY_IN_FRAMES 2
static void AudioQualityImprovement_cancelEchoFromPlay
(AudioQualityImprovement *aqi,
void *buffer, unsigned long length);
static void AudioQualityImprovement_free(AudioQualityImprovement *aqi);
(AudioQualityImprovement *audioQualityImprovement,
void *buffer,
unsigned long length);
static void AudioQualityImprovement_free
(AudioQualityImprovement *audioQualityImprovement);
static AudioQualityImprovement *AudioQualityImprovement_new
(const char *stringID, jlong longID, AudioQualityImprovement *next);
static void AudioQualityImprovement_popFromPlay
(AudioQualityImprovement *aqi, spx_uint32_t sampleCount);
static void AudioQualityImprovement_resampleInPlay
(AudioQualityImprovement *aqi,
double sampleRate, unsigned long sampleSizeInBits, int channels,
void *buffer, unsigned long length);
static void AudioQualityImprovement_retain(AudioQualityImprovement *aqi);
(AudioQualityImprovement *audioQualityImprovement,
double sampleRate,
unsigned long sampleSizeInBits,
int channels,
void *buffer,
unsigned long length);
static void AudioQualityImprovement_retain
(AudioQualityImprovement *audioQualityImprovement);
static void AudioQualityImprovement_setFrameSize
(AudioQualityImprovement *aqi, jint frameSize);
static void AudioQualityImprovement_setInputLatency
(AudioQualityImprovement *aqi, jlong inputLatency);
static void AudioQualityImprovement_setOutputLatency
(AudioQualityImprovement *aqi, jlong outputLatency);
/**
* Updates the indicator of the specified <tt>AudioQualityImprovement</tt> which
* determines whether <tt>AudioQualityImprovement#play</tt> delays the access to
* it from <tt>AudioQualityImprovement#echo</tt>.
*
* @param aqi the <tt>AudioQualityImprovement</tt> of which to update the
* indicator which determines whether <tt>AudioQualityImprovement#play</tt>
* delays the access to it from <tt>AudioQualityImprovement#echo</tt>
*/
static void AudioQualityImprovement_updatePlayDelay
(AudioQualityImprovement *aqi);
static void AudioQualityImprovement_updatePlayIsDelaying
(AudioQualityImprovement *aqi);
(AudioQualityImprovement *audioQualityImprovement, jint frameSize);
static void AudioQualityImprovement_updatePreprocess
(AudioQualityImprovement *aqi);
(AudioQualityImprovement *audioQualityImprovement);
static Mutex *AudioQualityImprovement_sharedInstancesMutex = NULL;
static AudioQualityImprovement *AudioQualityImprovement_sharedInstances = NULL;
#ifndef _WIN32
static pthread_mutex_t AudioQualityImprovement_sharedInstancesMutex
= PTHREAD_MUTEX_INITIALIZER;
#else /* Windows */
static CRITICAL_SECTION AudioQualityImprovement_sharedInstancesMutex = {(void*)-1, -1, 0, 0, 0, 0};
#endif
static AudioQualityImprovement *AudioQualityImprovement_sharedInstances
= NULL;
/**
*
* @param aqi
* @param buffer
* @param length the length of <tt>buffer</tt> in bytes
*/
static void
AudioQualityImprovement_cancelEchoFromPlay
(AudioQualityImprovement *aqi,
(AudioQualityImprovement *audioQualityImprovement,
void *buffer, unsigned long length)
{
spx_uint32_t sampleCount;
if (aqi->playIsDelaying == JNI_TRUE)
return;
sampleCount = length / sizeof(spx_int16_t);
if (aqi->playLength < sampleCount)
return;
/*
* Ensure that out exists and is large enough to receive the result of the
* echo cancellation.
*/
if (!(aqi->out) || (aqi->outCapacity < length))
if (!(audioQualityImprovement->out)
|| (audioQualityImprovement->outCapacity < length))
{
spx_int16_t *newOut = realloc(aqi->out, length);
spx_int16_t *newOut = realloc(audioQualityImprovement->out, length);
if (newOut)
{
aqi->out = newOut;
aqi->outCapacity = length;
audioQualityImprovement->out = newOut;
audioQualityImprovement->outCapacity = length;
}
else
return;
}
/* Perform the echo cancellation and return the result in buffer. */
speex_echo_cancellation(aqi->echo, buffer, aqi->play, aqi->out);
AudioQualityImprovement_popFromPlay(aqi, sampleCount);
memcpy(buffer, aqi->out, length);
speex_echo_cancellation(
audioQualityImprovement->echo,
buffer,
audioQualityImprovement->play,
audioQualityImprovement->out);
audioQualityImprovement->playSize = 0;
memcpy(buffer, audioQualityImprovement->out, length);
}
static void
AudioQualityImprovement_free(AudioQualityImprovement *aqi)
AudioQualityImprovement_free(AudioQualityImprovement *audioQualityImprovement)
{
/* mutex */
Mutex_free(aqi->mutex);
mutex_destroy(audioQualityImprovement->mutex);
free(audioQualityImprovement->mutex);
/* preprocess */
if (aqi->preprocess)
speex_preprocess_state_destroy(aqi->preprocess);
if (audioQualityImprovement->preprocess)
speex_preprocess_state_destroy(audioQualityImprovement->preprocess);
/* echo */
if (aqi->echo)
speex_echo_state_destroy(aqi->echo);
if (audioQualityImprovement->echo)
speex_echo_state_destroy(audioQualityImprovement->echo);
/* out */
if (aqi->out)
free(aqi->out);
if (audioQualityImprovement->out)
free(audioQualityImprovement->out);
/* play */
if (aqi->play)
free(aqi->play);
if (audioQualityImprovement->play)
free(audioQualityImprovement->play);
/* resampler */
if (aqi->resampler)
speex_resampler_destroy(aqi->resampler);
if (audioQualityImprovement->resampler)
speex_resampler_destroy(audioQualityImprovement->resampler);
/* stringID */
free(aqi->stringID);
free(audioQualityImprovement->stringID);
free(aqi);
free(audioQualityImprovement);
}
AudioQualityImprovement *
@ -131,7 +103,7 @@ AudioQualityImprovement_getSharedInstance(const char *stringID, jlong longID)
{
AudioQualityImprovement *theSharedInstance = NULL;
if (!Mutex_lock(AudioQualityImprovement_sharedInstancesMutex))
if (!mutex_lock(&AudioQualityImprovement_sharedInstancesMutex))
{
AudioQualityImprovement *aSharedInstance
= AudioQualityImprovement_sharedInstances;
@ -159,131 +131,125 @@ AudioQualityImprovement_getSharedInstance(const char *stringID, jlong longID)
if (theSharedInstance)
AudioQualityImprovement_sharedInstances = theSharedInstance;
}
Mutex_unlock(AudioQualityImprovement_sharedInstancesMutex);
mutex_unlock(&AudioQualityImprovement_sharedInstancesMutex);
}
return theSharedInstance;
}
/** Loads the <tt>AudioQualityImprovement</tt> class. */
void
AudioQualityImprovement_load()
{
AudioQualityImprovement_sharedInstancesMutex = Mutex_new(NULL);
}
static AudioQualityImprovement *
AudioQualityImprovement_new
(const char *stringID, jlong longID, AudioQualityImprovement *next)
{
AudioQualityImprovement *aqi = calloc(1, sizeof(AudioQualityImprovement));
AudioQualityImprovement *audioQualityImprovement
= malloc(sizeof(AudioQualityImprovement));
if (aqi)
if (audioQualityImprovement)
{
audioQualityImprovement->echo = NULL;
audioQualityImprovement->mutex = NULL;
audioQualityImprovement->out = NULL;
audioQualityImprovement->play = NULL;
audioQualityImprovement->preprocess = NULL;
audioQualityImprovement->resampler = NULL;
/* audioQualityImprovement->stringID = NULL; */
/* stringID */
aqi->stringID = strdup(stringID);
if (!(aqi->stringID))
audioQualityImprovement->stringID = strdup(stringID);
if (!(audioQualityImprovement->stringID))
{
AudioQualityImprovement_free(aqi);
AudioQualityImprovement_free(audioQualityImprovement);
return NULL;
}
/* mutex */
aqi->mutex = Mutex_new(NULL);
if (!(aqi->mutex))
audioQualityImprovement->mutex = malloc(sizeof(Mutex));
if (!(audioQualityImprovement->mutex)
|| mutex_init(audioQualityImprovement->mutex, NULL))
{
AudioQualityImprovement_free(aqi);
AudioQualityImprovement_free(audioQualityImprovement);
return NULL;
}
aqi->inputLatency = -1;
aqi->longID = longID;
aqi->next = next;
aqi->outputLatency = -1;
aqi->retainCount = 1;
audioQualityImprovement->denoise = JNI_FALSE;
audioQualityImprovement->echoFilterLengthInMillis = 0;
audioQualityImprovement->frameSize = 0;
audioQualityImprovement->longID = longID;
audioQualityImprovement->next = next;
audioQualityImprovement->retainCount = 1;
audioQualityImprovement->sampleRate = 0;
}
return aqi;
}
static void
AudioQualityImprovement_popFromPlay
(AudioQualityImprovement *aqi, spx_uint32_t sampleCount)
{
spx_uint32_t i;
spx_uint32_t sampleCountToMove = aqi->playLength - sampleCount;
spx_int16_t *playNew = aqi->play;
spx_int16_t *playOld = aqi->play + sampleCount;
for (i = 0; i < sampleCountToMove; i++)
*playNew++ = *playOld++;
aqi->playLength -= sampleCount;
return audioQualityImprovement;
}
/**
*
* @param aqi
* @param sampleOrigin
* @param sampleRate
* @param sampleSizeInBits
* @param channels
* @param latency the latency of the stream associated with <tt>buffer</tt> in
* milliseconds
* @param buffer
* @param length the length of <tt>buffer</tt> in bytes
*/
void
AudioQualityImprovement_process
(AudioQualityImprovement *aqi,
void AudioQualityImprovement_process
(AudioQualityImprovement *audioQualityImprovement,
AudioQualityImprovementSampleOrigin sampleOrigin,
double sampleRate, unsigned long sampleSizeInBits, int channels,
jlong latency,
void *buffer, unsigned long length)
double sampleRate,
unsigned long sampleSizeInBits,
int channels,
void *buffer,
unsigned long length)
{
if ((sampleSizeInBits == 16) && (channels == 1) && !Mutex_lock(aqi->mutex))
if ((sampleSizeInBits == 16)
&& (channels == 1)
&& !mutex_lock(audioQualityImprovement->mutex))
{
switch (sampleOrigin)
{
case AUDIO_QUALITY_IMPROVEMENT_SAMPLE_ORIGIN_INPUT:
if (sampleRate == aqi->sampleRate)
if (sampleRate == audioQualityImprovement->sampleRate)
{
AudioQualityImprovement_setFrameSize(aqi, length);
if (aqi->preprocess)
AudioQualityImprovement_setFrameSize(
audioQualityImprovement,
length);
if (audioQualityImprovement->preprocess)
{
AudioQualityImprovement_setInputLatency(aqi, latency);
if (aqi->echo && aqi->play && aqi->playLength)
if (audioQualityImprovement->echo
&& audioQualityImprovement->play
&& (audioQualityImprovement->playSize
== audioQualityImprovement->frameSize))
{
AudioQualityImprovement_cancelEchoFromPlay(
aqi,
buffer, length);
audioQualityImprovement,
buffer,
length);
}
speex_preprocess_run(aqi->preprocess, buffer);
speex_preprocess_run(
audioQualityImprovement->preprocess,
buffer);
}
}
break;
case AUDIO_QUALITY_IMPROVEMENT_SAMPLE_ORIGIN_OUTPUT:
if (aqi->preprocess && aqi->echo)
if (audioQualityImprovement->preprocess
&& audioQualityImprovement->echo)
{
AudioQualityImprovement_setOutputLatency(aqi, latency);
AudioQualityImprovement_resampleInPlay(
aqi,
sampleRate, sampleSizeInBits, channels,
buffer, length);
audioQualityImprovement,
sampleRate,
sampleSizeInBits,
channels,
buffer,
length);
}
break;
}
Mutex_unlock(aqi->mutex);
mutex_unlock(audioQualityImprovement->mutex);
}
}
void
AudioQualityImprovement_release(AudioQualityImprovement *aqi)
AudioQualityImprovement_release
(AudioQualityImprovement *audioQualityImprovement)
{
if (!Mutex_lock(AudioQualityImprovement_sharedInstancesMutex))
if (!mutex_lock(&AudioQualityImprovement_sharedInstancesMutex))
{
if (!Mutex_lock(aqi->mutex))
if (!mutex_lock(audioQualityImprovement->mutex))
{
--(aqi->retainCount);
if (aqi->retainCount < 1)
--(audioQualityImprovement->retainCount);
if (audioQualityImprovement->retainCount < 1)
{
if (aqi == AudioQualityImprovement_sharedInstances)
if (audioQualityImprovement
== AudioQualityImprovement_sharedInstances)
{
AudioQualityImprovement_sharedInstances
= AudioQualityImprovement_sharedInstances->next;
@ -298,435 +264,299 @@ AudioQualityImprovement_release(AudioQualityImprovement *aqi)
AudioQualityImprovement *nextSharedInstance
= prevSharedInstance->next;
if (aqi == nextSharedInstance)
if (audioQualityImprovement == nextSharedInstance)
{
prevSharedInstance->next = aqi->next;
prevSharedInstance->next
= audioQualityImprovement->next;
break;
}
prevSharedInstance = nextSharedInstance;
}
}
Mutex_unlock(aqi->mutex);
AudioQualityImprovement_free(aqi);
mutex_unlock(audioQualityImprovement->mutex);
AudioQualityImprovement_free(audioQualityImprovement);
}
else
Mutex_unlock(aqi->mutex);
mutex_unlock(audioQualityImprovement->mutex);
}
Mutex_unlock(AudioQualityImprovement_sharedInstancesMutex);
mutex_unlock(&AudioQualityImprovement_sharedInstancesMutex);
}
}
/**
*
* @param aqi
* @param sampleRate
* @param sampleSizeInBits
* @param channels
* @param buffer
* @param length the length of <tt>buffer</tt> in bytes
*/
static void
AudioQualityImprovement_resampleInPlay
(AudioQualityImprovement *aqi,
double sampleRate, unsigned long sampleSizeInBits, int channels,
void *buffer, unsigned long length)
(AudioQualityImprovement *audioQualityImprovement,
double sampleRate,
unsigned long sampleSizeInBits,
int channels,
void *buffer,
unsigned long length)
{
spx_uint32_t playSize;
spx_uint32_t playCapacity;
spx_uint32_t playLength;;
spx_int16_t *play;
if (sampleRate == aqi->sampleRate)
if (sampleRate == audioQualityImprovement->sampleRate)
playSize = length;
else if (length * aqi->sampleRate == aqi->frameSize * sampleRate)
else if (length * audioQualityImprovement->sampleRate
== audioQualityImprovement->frameSize * sampleRate)
{
if (aqi->resampler)
if (audioQualityImprovement->resampler)
{
speex_resampler_set_rate(
aqi->resampler,
(spx_uint32_t) sampleRate, (spx_uint32_t) (aqi->sampleRate));
playSize = aqi->frameSize;
audioQualityImprovement->resampler,
(spx_uint32_t) sampleRate,
(spx_uint32_t) (audioQualityImprovement->sampleRate));
playSize = audioQualityImprovement->frameSize;
}
else
{
aqi->resampler
audioQualityImprovement->resampler
= speex_resampler_init(
channels,
(spx_uint32_t) sampleRate, (spx_uint32_t) (aqi->sampleRate),
(spx_uint32_t) sampleRate,
(spx_uint32_t) (audioQualityImprovement->sampleRate),
SPEEX_RESAMPLER_QUALITY_VOIP,
NULL);
if (aqi->resampler)
playSize = aqi->frameSize;
else
if (!(audioQualityImprovement->resampler))
{
aqi->playIsDelaying = JNI_TRUE;
aqi->playLength = 0;
audioQualityImprovement->playSize = 0;
return;
}
}
}
else
{
/*
* The specified buffer neither is in the format of the audio capture
* nor can be resampled to it.
*/
aqi->playIsDelaying = JNI_TRUE;
aqi->playLength = 0;
audioQualityImprovement->playSize = 0;
return;
}
/* Ensure that play exists and is large enough. */
playCapacity
= ((1 + aqi->playDelay) + 1) * (aqi->frameSize / sizeof(spx_int16_t));
playLength = playSize / sizeof(spx_int16_t);
if (playCapacity < playLength)
playCapacity = playLength;
if (!(aqi->play) || (aqi->playCapacity < playCapacity))
if (!(audioQualityImprovement->play)
|| (audioQualityImprovement->playCapacity < playSize))
{
spx_int16_t *newPlay;
spx_int16_t *newPlay = realloc(audioQualityImprovement->play, playSize);
newPlay = realloc(aqi->play, playCapacity * sizeof(spx_int16_t));
if (newPlay)
{
if (!(aqi->play))
{
aqi->playIsDelaying = JNI_TRUE;
aqi->playLength = 0;
}
aqi->play = newPlay;
aqi->playCapacity = playCapacity;
audioQualityImprovement->play = newPlay;
audioQualityImprovement->playCapacity = playSize;
}
else
{
aqi->playIsDelaying = JNI_TRUE;
aqi->playLength = 0;
audioQualityImprovement->playSize = 0;
return;
}
}
/* Ensure that there is room for buffer in play. */
if (aqi->playLength + playLength > aqi->playCapacity)
if (length == audioQualityImprovement->frameSize)
{
aqi->playIsDelaying = JNI_TRUE;
aqi->playLength = 0;
/*
* We don't have enough room in play for buffer which means that we'll
* have to throw some samples away. But it'll effectively mean that
* we'll enlarge the drift which will disrupt the echo cancellation. So
* it seems the least of two evils to just reset the echo cancellation.
*/
speex_echo_state_reset(aqi->echo);
memcpy(audioQualityImprovement->play, buffer, playSize);
audioQualityImprovement->playSize = playSize;
}
/* Place buffer in play. */
play = aqi->play + aqi->playLength;
if (length == aqi->frameSize)
memcpy(play, buffer, playSize);
else
{
unsigned long sampleSizeInBytes = sampleSizeInBits / 8;
spx_uint32_t bufferSampleCount = length / sampleSizeInBytes;
spx_uint32_t playSampleCount = playSize / sampleSizeInBytes;
speex_resampler_process_interleaved_int(
aqi->resampler,
buffer, &bufferSampleCount, play, &playLength);
audioQualityImprovement->resampler,
buffer,
&bufferSampleCount,
audioQualityImprovement->play,
&playSampleCount);
audioQualityImprovement->playSize = playSampleCount * sampleSizeInBytes;
}
aqi->playLength += playLength;
/* Take into account the latency. */
if (aqi->playIsDelaying == JNI_TRUE)
AudioQualityImprovement_updatePlayIsDelaying(aqi);
}
static void
AudioQualityImprovement_retain(AudioQualityImprovement *aqi)
AudioQualityImprovement_retain
(AudioQualityImprovement *audioQualityImprovement)
{
if (!Mutex_lock(aqi->mutex))
if (!mutex_lock(audioQualityImprovement->mutex))
{
++(aqi->retainCount);
Mutex_unlock(aqi->mutex);
++(audioQualityImprovement->retainCount);
mutex_unlock(audioQualityImprovement->mutex);
}
}
/**
* Sets the indicator which determines whether noise suppression is to be
* performed by the specified <tt>AudioQualityImprovement</tt> (for captured
* audio).
*
* @param aqi the <tt>AudioQualityImprovement</tt> on which to set the indicator
* which determines whether it is to perform noise suppression (for captured audio)
* @param denoise <tt>JNI_TRUE</tt> if the specified <tt>aqi</tt> is to perform
* noise suppression (for captured audio); otherwise, <tt>JNI_FALSE</tt>
*/
void
AudioQualityImprovement_setDenoise
(AudioQualityImprovement *aqi, jboolean denoise)
(AudioQualityImprovement *audioQualityImprovement, jboolean denoise)
{
if (!Mutex_lock(aqi->mutex))
if (!mutex_lock(audioQualityImprovement->mutex))
{
if (aqi->denoise != denoise)
if (audioQualityImprovement->denoise != denoise)
{
aqi->denoise = denoise;
AudioQualityImprovement_updatePreprocess(aqi);
audioQualityImprovement->denoise = denoise;
AudioQualityImprovement_updatePreprocess(audioQualityImprovement);
}
Mutex_unlock(aqi->mutex);
mutex_unlock(audioQualityImprovement->mutex);
}
}
/**
* Sets the filter length in milliseconds of the echo cancellation
* implementation of the specified <tt>AudioQualityImprovement</tt>. The
* recommended filter length is approximately the third of the room
* reverberation time. For example, in a small room, reverberation time is in
* the order of 300 ms, so a filter length of 100 ms is a good choice (800
* samples at 8000 Hz sampling rate).
*
* @param aqi the <tt>AudioQualityImprovement</tt> to set the filter length of
* @param echoFilterLengthInMillis the filter length in milliseconds of the echo
* cancellation of <tt>aqi</tt>
*/
void
AudioQualityImprovement_setEchoFilterLengthInMillis
(AudioQualityImprovement *aqi, jlong echoFilterLengthInMillis)
(AudioQualityImprovement *audioQualityImprovement,
jlong echoFilterLengthInMillis)
{
if (echoFilterLengthInMillis < 0)
echoFilterLengthInMillis = 0;
if (!Mutex_lock(aqi->mutex))
if (!mutex_lock(audioQualityImprovement->mutex))
{
if (aqi->echoFilterLengthInMillis != echoFilterLengthInMillis)
if (audioQualityImprovement->echoFilterLengthInMillis
!= echoFilterLengthInMillis)
{
aqi->echoFilterLengthInMillis = echoFilterLengthInMillis;
AudioQualityImprovement_updatePreprocess(aqi);
audioQualityImprovement->echoFilterLengthInMillis
= echoFilterLengthInMillis;
AudioQualityImprovement_updatePreprocess(audioQualityImprovement);
}
Mutex_unlock(aqi->mutex);
mutex_unlock(audioQualityImprovement->mutex);
}
}
static void
AudioQualityImprovement_setFrameSize
(AudioQualityImprovement *aqi, jint frameSize)
(AudioQualityImprovement *audioQualityImprovement, jint frameSize)
{
if (aqi->frameSize != frameSize)
if (audioQualityImprovement->frameSize != frameSize)
{
aqi->frameSize = frameSize;
AudioQualityImprovement_updatePreprocess(aqi);
}
}
static void
AudioQualityImprovement_setInputLatency
(AudioQualityImprovement *aqi, jlong inputLatency)
{
if (aqi->inputLatency != inputLatency)
{
aqi->inputLatency = inputLatency;
AudioQualityImprovement_updatePlayDelay(aqi);
}
}
static void
AudioQualityImprovement_setOutputLatency
(AudioQualityImprovement *aqi, jlong outputLatency)
{
if (aqi->outputLatency != outputLatency)
{
aqi->outputLatency = outputLatency;
AudioQualityImprovement_updatePlayDelay(aqi);
audioQualityImprovement->frameSize = frameSize;
AudioQualityImprovement_updatePreprocess(audioQualityImprovement);
}
}
void
AudioQualityImprovement_setSampleRate
(AudioQualityImprovement *aqi, int sampleRate)
(AudioQualityImprovement *audioQualityImprovement, int sampleRate)
{
if (!Mutex_lock(aqi->mutex))
if (!mutex_lock(audioQualityImprovement->mutex))
{
if (aqi->sampleRate != sampleRate)
if (audioQualityImprovement->sampleRate != sampleRate)
{
aqi->sampleRate = sampleRate;
AudioQualityImprovement_updatePlayDelay(aqi);
AudioQualityImprovement_updatePreprocess(aqi);
audioQualityImprovement->sampleRate = sampleRate;
AudioQualityImprovement_updatePreprocess(audioQualityImprovement);
}
Mutex_unlock(aqi->mutex);
}
}
/** Unloads the <tt>AudioQualityImprovement</tt> class. */
void
AudioQualityImprovement_unload()
{
if (AudioQualityImprovement_sharedInstancesMutex)
{
Mutex_free(AudioQualityImprovement_sharedInstancesMutex);
AudioQualityImprovement_sharedInstancesMutex = NULL;
}
}
static void
AudioQualityImprovement_updatePlayDelay(AudioQualityImprovement *aqi)
{
spx_uint32_t playDelay;
if ((aqi->inputLatency < 0)
|| (aqi->outputLatency < 0)
|| !(aqi->frameSize)
|| !(aqi->sampleRate))
{
playDelay = MIN_PLAY_DELAY_IN_FRAMES;
mutex_unlock(audioQualityImprovement->mutex);
}
else
{
playDelay
= (aqi->outputLatency * aqi->sampleRate)
/ ((aqi->frameSize / sizeof(spx_int16_t)) * 1000);
if (playDelay < MIN_PLAY_DELAY_IN_FRAMES)
playDelay = MIN_PLAY_DELAY_IN_FRAMES;
}
if (aqi->playDelay != playDelay)
{
aqi->playDelay = playDelay;
if (aqi->play && (aqi->playIsDelaying == JNI_TRUE))
AudioQualityImprovement_updatePlayIsDelaying(aqi);
}
}
/**
* Updates the indicator of the specified <tt>AudioQualityImprovement</tt> which
* determines whether <tt>AudioQualityImprovement#play</tt> delays the access to
* it from <tt>AudioQualityImprovement#echo</tt>.
*
* @param aqi the <tt>AudioQualityImprovement</tt> of which to update the
* indicator which determines whether <tt>AudioQualityImprovement#play</tt>
* delays the access to it from <tt>AudioQualityImprovement#echo</tt>
*/
static void
AudioQualityImprovement_updatePlayIsDelaying(AudioQualityImprovement *aqi)
{
spx_uint32_t playDelay
= aqi->playDelay * (aqi->frameSize / sizeof(spx_int16_t));
aqi->playIsDelaying
= ((aqi->playLength < playDelay) && (playDelay <= aqi->playCapacity))
? JNI_TRUE
: JNI_FALSE;
}
static void
AudioQualityImprovement_updatePreprocess(AudioQualityImprovement *aqi)
AudioQualityImprovement_updatePreprocess
(AudioQualityImprovement *audioQualityImprovement)
{
if (aqi->echo)
if (audioQualityImprovement->echo)
{
int frameSize = 0;
if ((aqi->echoFilterLengthInMillis > 0)
&& (aqi->sampleRate > 0)
if ((audioQualityImprovement->echoFilterLengthInMillis > 0)
&& (audioQualityImprovement->sampleRate > 0)
&& speex_echo_ctl(
aqi->echo,
SPEEX_ECHO_GET_FRAME_SIZE, &frameSize))
audioQualityImprovement->echo,
SPEEX_ECHO_GET_FRAME_SIZE,
&frameSize))
frameSize = 0;
if (frameSize
&& (aqi->frameSize
&& (audioQualityImprovement->frameSize
== (frameSize * (16 /* sampleSizeInBits */ / 8))))
{
int echoFilterLength
= (int)
((aqi->sampleRate * aqi->echoFilterLengthInMillis)
((audioQualityImprovement->sampleRate
* audioQualityImprovement->echoFilterLengthInMillis)
/ 1000);
if (aqi->filterLengthOfEcho != echoFilterLength)
if (audioQualityImprovement->filterLengthOfEcho != echoFilterLength)
frameSize = 0;
}
else
frameSize = 0;
if (frameSize < 1)
{
if (aqi->preprocess)
if (audioQualityImprovement->preprocess)
{
speex_preprocess_ctl(
aqi->preprocess,
SPEEX_PREPROCESS_SET_ECHO_STATE, NULL);
audioQualityImprovement->preprocess,
SPEEX_PREPROCESS_SET_ECHO_STATE,
NULL);
}
speex_echo_state_destroy(aqi->echo);
aqi->echo = NULL;
speex_echo_state_destroy(audioQualityImprovement->echo);
audioQualityImprovement->echo = NULL;
}
}
if (aqi->preprocess
&& ((aqi->frameSize != aqi->frameSizeOfPreprocess)
|| (aqi->sampleRate != aqi->sampleRateOfPreprocess)))
if (audioQualityImprovement->preprocess
&& ((audioQualityImprovement->frameSize
!= audioQualityImprovement->frameSizeOfPreprocess)
|| (audioQualityImprovement->sampleRate
!= audioQualityImprovement->sampleRateOfPreprocess)))
{
speex_preprocess_state_destroy(aqi->preprocess);
aqi->preprocess = NULL;
speex_preprocess_state_destroy(audioQualityImprovement->preprocess);
audioQualityImprovement->preprocess = NULL;
}
if ((aqi->frameSize > 0) && (aqi->sampleRate > 0))
if ((audioQualityImprovement->frameSize > 0)
&& (audioQualityImprovement->sampleRate > 0))
{
if (aqi->echoFilterLengthInMillis > 0)
if (audioQualityImprovement->echoFilterLengthInMillis > 0)
{
if (!(aqi->echo))
if (!(audioQualityImprovement->echo))
{
int echoFrameSize
= aqi->frameSize / (16 /* sampleSizeInBits */ / 8);
int echoFilterLength
= (int)
((aqi->sampleRate * aqi->echoFilterLengthInMillis)
((audioQualityImprovement->sampleRate
* audioQualityImprovement
->echoFilterLengthInMillis)
/ 1000);
aqi->echo
= speex_echo_state_init(echoFrameSize, echoFilterLength);
aqi->filterLengthOfEcho = echoFilterLength;
/*
* Since echo has just been (re)created, make sure that the
* delay in play will happen again taking into consideration the
* latest frameSize.
*/
if (aqi->play)
AudioQualityImprovement_updatePlayIsDelaying(aqi);
audioQualityImprovement->echo
= speex_echo_state_init(
audioQualityImprovement->frameSize
/ (16 /* sampleSizeInBits */ / 8),
echoFilterLength);
audioQualityImprovement->filterLengthOfEcho
= echoFilterLength;
}
if (aqi->echo)
if (audioQualityImprovement->echo)
{
speex_echo_ctl(
aqi->echo,
SPEEX_ECHO_SET_SAMPLING_RATE, &(aqi->sampleRate));
audioQualityImprovement->echo,
SPEEX_ECHO_SET_SAMPLING_RATE,
&(audioQualityImprovement->sampleRate));
}
}
if (aqi->denoise || aqi->echo)
if (audioQualityImprovement->denoise || audioQualityImprovement->echo)
{
if (!(aqi->preprocess))
if (!(audioQualityImprovement->preprocess))
{
aqi->preprocess
audioQualityImprovement->preprocess
= speex_preprocess_state_init(
aqi->frameSize / (16 /* sampleSizeInBits */ / 8),
aqi->sampleRate);
aqi->frameSizeOfPreprocess = aqi->frameSize;
aqi->sampleRateOfPreprocess = aqi->sampleRate;
if (aqi->preprocess)
audioQualityImprovement->frameSize
/ (16 /* sampleSizeInBits */ / 8),
audioQualityImprovement->sampleRate);
audioQualityImprovement->frameSizeOfPreprocess
= audioQualityImprovement->frameSize;
audioQualityImprovement->sampleRateOfPreprocess
= audioQualityImprovement->sampleRate;
if (audioQualityImprovement->preprocess)
{
int on = 1;
int vad = 1;
speex_preprocess_ctl(
aqi->preprocess,
SPEEX_PREPROCESS_SET_DEREVERB, &on);
speex_preprocess_ctl(
aqi->preprocess,
SPEEX_PREPROCESS_SET_VAD, &on);
audioQualityImprovement->preprocess,
SPEEX_PREPROCESS_SET_VAD,
&vad);
}
}
if (aqi->preprocess)
if (audioQualityImprovement->preprocess)
{
int denoise = (aqi->denoise == JNI_TRUE) ? 1 : 0;
int denoise
= (audioQualityImprovement->denoise == JNI_TRUE) ? 1 : 0;
speex_preprocess_ctl(
aqi->preprocess,
SPEEX_PREPROCESS_SET_DENOISE, &denoise);
if (aqi->echo)
audioQualityImprovement->preprocess,
SPEEX_PREPROCESS_SET_DENOISE,
&denoise);
if (audioQualityImprovement->echo)
{
speex_preprocess_ctl(
aqi->preprocess,
SPEEX_PREPROCESS_SET_ECHO_STATE, aqi->echo);
audioQualityImprovement->preprocess,
SPEEX_PREPROCESS_SET_ECHO_STATE,
audioQualityImprovement->echo);
}
}
}

152
src/native/portaudio/AudioQualityImprovement.h
Unescape View File

@ -14,7 +14,64 @@
typedef void *AudioQualityImprovement;
#else /* #ifndef AUDIO_QUALITY_IMPROVEMENT_IMPLEMENTATION */
#include "Mutex.h"
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
typedef CRITICAL_SECTION Mutex;
static inline int mutex_init(Mutex* mutex, void* arg)
{
InitializeCriticalSection(mutex);
arg = NULL; /* unused */
return 0;
}
static inline int mutex_destroy(Mutex* mutex)
{
DeleteCriticalSection(mutex);
return 0;
}
static inline int mutex_lock(Mutex* mutex)
{
EnterCriticalSection(mutex);
return 0;
}
static inline int mutex_unlock(Mutex* mutex)
{
LeaveCriticalSection(mutex);
return 0;
}
#else /* Unix */
#include <pthread.h>
typedef pthread_mutex_t Mutex;
static inline int mutex_init(Mutex* mutex, void* arg)
{
return pthread_mutex_init(mutex, arg);
}
static inline int mutex_destroy(Mutex* mutex)
{
return pthread_mutex_destroy(mutex);
}
static inline int mutex_lock(Mutex* mutex)
{
return pthread_mutex_lock(mutex);
}
static inline int mutex_unlock(Mutex* mutex)
{
return pthread_mutex_unlock(mutex);
return 0;
}
#endif
#include <speex/speex_echo.h>
#include <speex/speex_preprocess.h>
#include <speex/speex_resampler.h>
@ -24,48 +81,17 @@ typedef struct _AudioQualityImprovement
jboolean denoise;
SpeexEchoState *echo;
jlong echoFilterLengthInMillis;
/** The length of the echo cancelling filter of #echo in samples. */
int filterLengthOfEcho;
jint frameSize;
int frameSizeOfPreprocess;
/** The capture latency in milliseconds. */
jlong inputLatency;
jlong longID;
Mutex *mutex;
struct _AudioQualityImprovement *next;
/**
* The intermediate buffer into which the result of echo cancellation is
* written for a specific <tt>buffer</tt> of captured audio.
*/
spx_int16_t *out;
/** The capacity of #out in bytes. */
spx_uint32_t outCapacity;
/** The playback latency in milliseconds. */
jlong outputLatency;
spx_int16_t *play;
/**
* The number of samples allocated to #play regardless of whether they are
* valid or not.
*/
spx_uint32_t playCapacity;
/** The number of frames to delay playback with. */
spx_uint32_t playDelay;
/**
* The indicator which determines whether #play is currently delaying the
* access to it from #echo.
*/
jboolean playIsDelaying;
/** The number of valid samples written into #play. */
spx_uint32_t playLength;
spx_uint32_t playSize;
SpeexPreprocessState *preprocess;
SpeexResamplerState *resampler;
int retainCount;
@ -73,68 +99,32 @@ typedef struct _AudioQualityImprovement
int sampleRateOfPreprocess;
char *stringID;
} AudioQualityImprovement;
#endif /* #ifndef AUDIO_QUALITY_IMPROVEMENT_IMPLEMENTATION */
typedef enum
{
/**
* The constant which indicates that the associated samples have originated
* from an input stream i.e. capture.
*/
AUDIO_QUALITY_IMPROVEMENT_SAMPLE_ORIGIN_INPUT,
/**
* The constant which indicates that the associated samples have originated
* from an output stream i.e. playback.
*/
AUDIO_QUALITY_IMPROVEMENT_SAMPLE_ORIGIN_OUTPUT
} AudioQualityImprovementSampleOrigin;
AudioQualityImprovement *AudioQualityImprovement_getSharedInstance
(const char *stringID, jlong longID);
/** Loads the <tt>AudioQualityImprovement</tt> class. */
void AudioQualityImprovement_load();
void AudioQualityImprovement_process
(AudioQualityImprovement *aqi,
(AudioQualityImprovement *audioQualityImprovement,
AudioQualityImprovementSampleOrigin sampleOrigin,
double sampleRate, unsigned long sampleSizeInBits, int channels,
jlong latency,
void *buffer, unsigned long length);
void AudioQualityImprovement_release(AudioQualityImprovement *aqi);
/**
* Sets the indicator which determines whether noise suppression is to be
* performed by the specified <tt>AudioQualityImprovement</tt> (for captured
* audio).
*
* @param aqi the <tt>AudioQualityImprovement</tt> on which to set the indicator
* which determines whether it is to perform noise suppression (for captured audio)
* @param denoise <tt>JNI_TRUE</tt> if the specified <tt>aqi</tt> is to perform
* noise suppression (for captured audio); otherwise, <tt>JNI_FALSE</tt>
*/
double sampleRate,
unsigned long sampleSizeInBits,
int channels,
void *buffer,
unsigned long length);
void AudioQualityImprovement_release
(AudioQualityImprovement *audioQualityImprovement);
void AudioQualityImprovement_setDenoise
(AudioQualityImprovement *aqi, jboolean denoise);
/**
* Sets the filter length in milliseconds of the echo cancellation
* implementation of the specified <tt>AudioQualityImprovement</tt>. The
* recommended filter length is approximately the third of the room
* reverberation time. For example, in a small room, reverberation time is in
* the order of 300 ms, so a filter length of 100 ms is a good choice (800
* samples at 8000 Hz sampling rate).
*
* @param aqi the <tt>AudioQualityImprovement</tt> to set the filter length of
* @param echoFilterLengthInMillis the filter length in milliseconds of the echo
* cancellation of <tt>aqi</tt>
*/
(AudioQualityImprovement *audioQualityImprovement, jboolean denoise);
void AudioQualityImprovement_setEchoFilterLengthInMillis
(AudioQualityImprovement *aqi, jlong echoFilterLengthInMillis);
(AudioQualityImprovement *audioQualityImprovement,
jlong echoFilterLengthInMillis);
void AudioQualityImprovement_setSampleRate
(AudioQualityImprovement *aqi, int sampleRate);
/** Unloads the <tt>AudioQualityImprovement</tt> class. */
void AudioQualityImprovement_unload();
(AudioQualityImprovement *audioQualityImprovement, int sampleRate);
#endif /* #ifndef _NET_JAVA_SIP_COMMUNICATOR_IMPL_NEOMEDIA_PORTAUDIO_AUDIOQUALITYIMPROVEMENT_H_ */

95
src/native/portaudio/ConditionVariable.h
Unescape View File

@ -1,95 +0,0 @@
/*
* SIP Communicator, the OpenSource Java VoIP and Instant Messaging client.
*
* Distributable under LGPL license.
* See terms of license at gnu.org.
*/
#ifndef _NET_JAVA_SIP_COMMUNICATOR_IMPL_NEOMEDIA_CONDITIONVARIABLE_H_
#define _NET_JAVA_SIP_COMMUNICATOR_IMPL_NEOMEDIA_CONDITIONVARIABLE_H_
#include "Mutex.h"
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
typedef HANDLE ConditionVariable;
static inline void ConditionVariable_free(ConditionVariable *condVar)
{
if (CloseHandle(*condVar))
free(condVar);
}
static inline ConditionVariable *ConditionVariable_new(void *attr)
{
ConditionVariable *condVar = malloc(sizeof(ConditionVariable));
if (condVar)
{
HANDLE event = CreateEvent(NULL, FALSE, FALSE, NULL);
if (event)
*condVar = event;
else
{
free(condVar);
condVar = NULL;
}
}
return condVar;
}
static inline int ConditionVariable_notify(ConditionVariable *condVar)
{
return SetEvent(*condVar) ? 0 : GetLastError();
}
static inline int ConditionVariable_wait
(ConditionVariable *condVar, Mutex *mutex)
{
DWORD waitForSingleObject;
LeaveCriticalSection(mutex);
waitForSingleObject = WaitForSingleObject(*condVar, INFINITE);
EnterCriticalSection(mutex);
return waitForSingleObject;
}
#else /* #ifdef _WIN32 */
#include <pthread.h>
typedef pthread_cond_t ConditionVariable;
static inline void ConditionVariable_free(ConditionVariable *condVar)
{
if (!pthread_cond_destroy(condVar))
free(condVar);
}
static inline ConditionVariable *ConditionVariable_new(void *attr)
{
ConditionVariable *condVar = malloc(sizeof(ConditionVariable));
if (condVar && pthread_cond_init(condVar, attr))
{
free(condVar);
condVar = NULL;
}
return condVar;
}
static inline int ConditionVariable_notify(ConditionVariable *condVar)
{
return pthread_cond_signal(condVar);
}
static inline int ConditionVariable_wait
(ConditionVariable *condVar, Mutex *mutex)
{
return pthread_cond_wait(condVar, mutex);
}
#endif /* #ifdef _WIN32 */
#endif /* _NET_JAVA_SIP_COMMUNICATOR_IMPL_NEOMEDIA_CONDITIONVARIABLE_H_ */

82
src/native/portaudio/Mutex.h
Unescape View File

@ -1,82 +0,0 @@
/*
* SIP Communicator, the OpenSource Java VoIP and Instant Messaging client.
*
* Distributable under LGPL license.
* See terms of license at gnu.org.
*/
#ifndef _NET_JAVA_SIP_COMMUNICATOR_IMPL_NEOMEDIA_PORTAUDIO_MUTEX_H_
#define _NET_JAVA_SIP_COMMUNICATOR_IMPL_NEOMEDIA_PORTAUDIO_MUTEX_H_
#include <stdlib.h>
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
typedef CRITICAL_SECTION Mutex;
static inline void Mutex_free(Mutex* mutex)
{
DeleteCriticalSection(mutex);
free(mutex);
}
static inline int Mutex_lock(Mutex* mutex)
{
EnterCriticalSection(mutex);
return 0;
}
static inline Mutex *Mutex_new(void* attr)
{
Mutex *mutex = malloc(sizeof(Mutex));
(void) attr;
if (mutex)
InitializeCriticalSection(mutex);
return mutex;
}
static inline int Mutex_unlock(Mutex* mutex)
{
LeaveCriticalSection(mutex);
return 0;
}
#else /* #ifdef _WIN32 */
#include <pthread.h>
typedef pthread_mutex_t Mutex;
static inline void Mutex_free(Mutex* mutex)
{
if (!pthread_mutex_destroy(mutex))
free(mutex);
}
static inline int Mutex_lock(Mutex* mutex)
{
return pthread_mutex_lock(mutex);
}
static inline Mutex *Mutex_new(void* attr)
{
Mutex *mutex = malloc(sizeof(Mutex));
if (mutex && pthread_mutex_init(mutex, attr))
{
free(mutex);
mutex = NULL;
}
return mutex;
}
static inline int Mutex_unlock(Mutex* mutex)
{
return pthread_mutex_unlock(mutex);
}
#endif /* #ifdef _WIN32 */
#endif /* #ifndef _NET_JAVA_SIP_COMMUNICATOR_IMPL_NEOMEDIA_PORTAUDIO_MUTEX_H_ */

2
src/native/portaudio/README
Unescape View File

@ -10,7 +10,7 @@
- Windows
$./configure --enable-static --disable-shared && make
Note for Windows x64, edit Makefile (after ./configure) and modify
%.lo: %.c $(MAKEFILE) $(PAINC)
$(LIBTOOL) --mode=compile $(CC) -c $(CFLAGS) $< -o $@

798
src/native/portaudio/net_java_sip_communicator_impl_neomedia_portaudio_PortAudio.c
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File diff suppressed because it is too large Load Diff

8
src/native/portaudio/net_java_sip_communicator_impl_neomedia_portaudio_PortAudio.h
Unescape View File

@ -221,6 +221,14 @@ JNIEXPORT jint JNICALL Java_net_java_sip_communicator_impl_neomedia_portaudio_Po
JNIEXPORT jint JNICALL Java_net_java_sip_communicator_impl_neomedia_portaudio_PortAudio_PaDeviceInfo_1getMaxOutputChannels
(JNIEnv *, jclass, jlong);
/*
* Class: net_java_sip_communicator_impl_neomedia_portaudio_PortAudio
* Method: PaDeviceInfo_getName
* Signature: (J)Ljava/lang/String;
*/
JNIEXPORT jstring JNICALL Java_net_java_sip_communicator_impl_neomedia_portaudio_PortAudio_PaDeviceInfo_1getName
(JNIEnv *, jclass, jlong);
/*
* Class: net_java_sip_communicator_impl_neomedia_portaudio_PortAudio
* Method: PaDeviceInfo_getNameBytes

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