musicdsp/audio-orchestra
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
b21b2f7413
audio-orchestra/airtaudio/api/Jack.cpp

793 lines
25 KiB
C++
Raw Blame History

/**
* @author Gary P. SCAVONE
*
* @copyright 2001-2013 Gary P. Scavone, all right reserved
*
* @license like MIT (see license file)
*/
#if defined(__UNIX_JACK__)
#include <unistd.h>
#include <limits.h>
#include <iostream>
#include <airtaudio/Interface.h>
#include <string.h>
// JACK is a low-latency audio server, originally written for the
// GNU/Linux operating system and now also ported to OS-X. It can
// connect a number of different applications to an audio device, as
// well as allowing them to share audio between themselves.
//
// When using JACK with RtAudio, "devices" refer to JACK clients that
// have ports connected to the server. The JACK server is typically
// started in a terminal as follows:
//
// .jackd -d alsa -d hw:0
//
// or through an interface program such as qjackctl. Many of the
// parameters normally set for a stream are fixed by the JACK server
// and can be specified when the JACK server is started. In
// particular,
//
// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
//
// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
// frames, and number of buffers = 4. Once the server is running, it
// is not possible to override these values. If the values are not
// specified in the command-line, the JACK server uses default values.
//
// The JACK server does not have to be running when an instance of
// RtApiJack is created, though the function getDeviceCount() will
// report 0 devices found until JACK has been started. When no
// devices are available (i.e., the JACK server is not running), a
// stream cannot be opened.
#include <jack/jack.h>
#include <unistd.h>
#include <cstdio>
// A structure to hold various information related to the Jack API
// implementation.
struct JackHandle {
jack_client_t *client;
jack_port_t **ports[2];
std::string deviceName[2];
bool xrun[2];
std::condition_variable condition;
int32_t drainCounter; // Tracks callback counts when draining
bool internalDrain; // Indicates if stop is initiated from callback or not.
JackHandle(void) :
client(0),
drainCounter(0),
internalDrain(false) {
ports[0] = 0;
ports[1] = 0;
xrun[0] = false;
xrun[1] = false;
}
};
static void jackSilentError(const char *) {};
airtaudio::api::Jack::Jack(void) {
// Nothing to do here.
#if !defined(__RTAUDIO_DEBUG__)
// Turn off Jack's internal error reporting.
jack_set_error_function(&jackSilentError);
#endif
}
airtaudio::api::Jack::~Jack(void) {
if (m_stream.state != STREAM_CLOSED) {
closeStream();
}
}
uint32_t airtaudio::api::Jack::getDeviceCount(void) {
// See if we can become a jack client.
jack_options_t options = (jack_options_t) (JackNoStartServer); //JackNullOption;
jack_status_t *status = NULL;
jack_client_t *client = jack_client_open("RtApiJackCount", options, status);
if (client == NULL) {
return 0;
}
const char **ports;
std::string port, previousPort;
uint32_t nChannels = 0, nDevices = 0;
ports = jack_get_ports(client, NULL, NULL, 0);
if (ports) {
// Parse the port names up to the first colon (:).
size_t iColon = 0;
do {
port = (char *) ports[ nChannels ];
iColon = port.find(":");
if (iColon != std::string::npos) {
port = port.substr(0, iColon + 1);
if (port != previousPort) {
nDevices++;
previousPort = port;
}
}
} while (ports[++nChannels]);
free(ports);
}
jack_client_close(client);
return nDevices;
}
airtaudio::DeviceInfo airtaudio::api::Jack::getDeviceInfo(uint32_t _device)
{
airtaudio::DeviceInfo info;
info.probed = false;
jack_options_t options = (jack_options_t) (JackNoStartServer); //JackNullOption
jack_status_t *status = NULL;
jack_client_t *client = jack_client_open("RtApiJackInfo", options, status);
if (client == NULL) {
m_errorText = "airtaudio::api::Jack::getDeviceInfo: Jack server not found or connection error!";
error(airtaudio::errorWarning);
return info;
}
const char **ports;
std::string port, previousPort;
uint32_t nPorts = 0, nDevices = 0;
ports = jack_get_ports(client, NULL, NULL, 0);
if (ports) {
// Parse the port names up to the first colon (:).
size_t iColon = 0;
do {
port = (char *) ports[ nPorts ];
iColon = port.find(":");
if (iColon != std::string::npos) {
port = port.substr(0, iColon);
if (port != previousPort) {
if (nDevices == _device) {
info.name = port;
}
nDevices++;
previousPort = port;
}
}
} while (ports[++nPorts]);
free(ports);
}
if (_device >= nDevices) {
jack_client_close(client);
m_errorText = "airtaudio::api::Jack::getDeviceInfo: device ID is invalid!";
error(airtaudio::errorInvalidUse);
return info;
}
// Get the current jack server sample rate.
info.sampleRates.clear();
info.sampleRates.push_back(jack_get_sample_rate(client));
// Count the available ports containing the client name as device
// channels. Jack "input ports" equal RtAudio output channels.
uint32_t nChannels = 0;
ports = jack_get_ports(client, info.name.c_str(), NULL, JackPortIsInput);
if (ports) {
while (ports[ nChannels ]) {
nChannels++;
}
free(ports);
info.outputChannels = nChannels;
}
// Jack "output ports" equal RtAudio input channels.
nChannels = 0;
ports = jack_get_ports(client, info.name.c_str(), NULL, JackPortIsOutput);
if (ports) {
while (ports[ nChannels ]) {
nChannels++;
}
free(ports);
info.inputChannels = nChannels;
}
if (info.outputChannels == 0 && info.inputChannels == 0) {
jack_client_close(client);
m_errorText = "airtaudio::api::Jack::getDeviceInfo: error determining Jack input/output channels!";
error(airtaudio::errorWarning);
return info;
}
// If device opens for both playback and capture, we determine the channels.
if (info.outputChannels > 0 && info.inputChannels > 0) {
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
}
// Jack always uses 32-bit floats.
info.nativeFormats = airtaudio::FLOAT32;
// Jack doesn't provide default devices so we'll use the first available one.
if ( _device == 0
&& info.outputChannels > 0) {
info.isDefaultOutput = true;
}
if ( _device == 0
&& info.inputChannels > 0) {
info.isDefaultInput = true;
}
jack_client_close(client);
info.probed = true;
return info;
}
static int32_t jackCallbackHandler(jack_nframes_t _nframes, void *_infoPointer) {
airtaudio::CallbackInfo* info = (airtaudio::CallbackInfo*)_infoPointer;
airtaudio::api::Jack* object = (airtaudio::api::Jack*)info->object;
if (object->callbackEvent((uint64_t)_nframes) == false) {
return 1;
}
return 0;
}
// This function will be called by a spawned thread when the Jack
// server signals that it is shutting down. It is necessary to handle
// it this way because the jackShutdown() function must return before
// the jack_deactivate() function (in closeStream()) will return.
static void *jackCloseStream(void *_ptr) {
airtaudio::CallbackInfo* info = (airtaudio::CallbackInfo*)_ptr;
airtaudio::api::Jack* object = (airtaudio::api::Jack*)info->object;
object->closeStream();
pthread_exit(NULL);
}
static void jackShutdown(void* _infoPointer) {
airtaudio::CallbackInfo* info = (airtaudio::CallbackInfo*)_infoPointer;
airtaudio::api::Jack* object = (airtaudio::api::Jack*)info->object;
// Check current stream state. If stopped, then we'll assume this
// was called as a result of a call to airtaudio::api::Jack::stopStream (the
// deactivation of a client handle causes this function to be called).
// If not, we'll assume the Jack server is shutting down or some
// other problem occurred and we should close the stream.
if (object->isStreamRunning() == false) {
return;
}
pthread_t threadId;
pthread_create(&threadId, NULL, jackCloseStream, info);
std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
}
static int32_t jackXrun(void* _infoPointer) {
JackHandle* handle = (JackHandle*)_infoPointer;
if (handle->ports[0]) {
handle->xrun[0] = true;
}
if (handle->ports[1]) {
handle->xrun[1] = true;
}
return 0;
}
bool airtaudio::api::Jack::probeDeviceOpen(uint32_t device,
airtaudio::api::StreamMode mode,
uint32_t channels,
uint32_t firstChannel,
uint32_t sampleRate,
airtaudio::format format,
uint32_t *bufferSize,
airtaudio::StreamOptions *options) {
JackHandle *handle = (JackHandle *) m_stream.apiHandle;
// Look for jack server and try to become a client (only do once per stream).
jack_client_t *client = 0;
if (mode == OUTPUT || (mode == INPUT && m_stream.mode != OUTPUT)) {
jack_options_t jackoptions = (jack_options_t) (JackNoStartServer); //JackNullOption;
jack_status_t *status = NULL;
if (options && !options->streamName.empty()) {
client = jack_client_open(options->streamName.c_str(), jackoptions, status);
} else {
client = jack_client_open("RtApiJack", jackoptions, status);
}
if (client == 0) {
m_errorText = "airtaudio::api::Jack::probeDeviceOpen: Jack server not found or connection error!";
error(airtaudio::errorWarning);
return FAILURE;
}
}
else {
// The handle must have been created on an earlier pass.
client = handle->client;
}
const char **ports;
std::string port, previousPort, deviceName;
uint32_t nPorts = 0, nDevices = 0;
ports = jack_get_ports(client, NULL, NULL, 0);
if (ports) {
// Parse the port names up to the first colon (:).
size_t iColon = 0;
do {
port = (char *) ports[ nPorts ];
iColon = port.find(":");
if (iColon != std::string::npos) {
port = port.substr(0, iColon);
if (port != previousPort) {
if (nDevices == device) deviceName = port;
nDevices++;
previousPort = port;
}
}
} while (ports[++nPorts]);
free(ports);
}
if (device >= nDevices) {
m_errorText = "airtaudio::api::Jack::probeDeviceOpen: device ID is invalid!";
return FAILURE;
}
// Count the available ports containing the client name as device
// channels. Jack "input ports" equal RtAudio output channels.
uint32_t nChannels = 0;
uint64_t flag = JackPortIsInput;
if (mode == INPUT) flag = JackPortIsOutput;
ports = jack_get_ports(client, deviceName.c_str(), NULL, flag);
if (ports) {
while (ports[ nChannels ]) nChannels++;
free(ports);
}
// Compare the jack ports for specified client to the requested number of channels.
if (nChannels < (channels + firstChannel)) {
m_errorStream << "airtaudio::api::Jack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
m_errorText = m_errorStream.str();
return FAILURE;
}
// Check the jack server sample rate.
uint32_t jackRate = jack_get_sample_rate(client);
if (sampleRate != jackRate) {
jack_client_close(client);
m_errorStream << "airtaudio::api::Jack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
m_errorText = m_errorStream.str();
return FAILURE;
}
m_stream.sampleRate = jackRate;
// Get the latency of the JACK port.
ports = jack_get_ports(client, deviceName.c_str(), NULL, flag);
if (ports[ firstChannel ]) {
// Added by Ge Wang
jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
// the range (usually the min and max are equal)
jack_latency_range_t latrange; latrange.min = latrange.max = 0;
// get the latency range
jack_port_get_latency_range(jack_port_by_name(client, ports[firstChannel]), cbmode, &latrange);
// be optimistic, use the min!
m_stream.latency[mode] = latrange.min;
//m_stream.latency[mode] = jack_port_get_latency(jack_port_by_name(client, ports[ firstChannel ]));
}
free(ports);
// The jack server always uses 32-bit floating-point data.
m_stream.deviceFormat[mode] = FLOAT32;
m_stream.userFormat = format;
if (options && options->flags & NONINTERLEAVED) m_stream.userInterleaved = false;
else m_stream.userInterleaved = true;
// Jack always uses non-interleaved buffers.
m_stream.deviceInterleaved[mode] = false;
// Jack always provides host byte-ordered data.
m_stream.doByteSwap[mode] = false;
// Get the buffer size. The buffer size and number of buffers
// (periods) is set when the jack server is started.
m_stream.bufferSize = (int) jack_get_buffer_size(client);
*bufferSize = m_stream.bufferSize;
m_stream.nDeviceChannels[mode] = channels;
m_stream.nUserChannels[mode] = channels;
// Set flags for buffer conversion.
m_stream.doConvertBuffer[mode] = false;
if (m_stream.userFormat != m_stream.deviceFormat[mode])
m_stream.doConvertBuffer[mode] = true;
if (m_stream.userInterleaved != m_stream.deviceInterleaved[mode] &&
m_stream.nUserChannels[mode] > 1)
m_stream.doConvertBuffer[mode] = true;
// Allocate our JackHandle structure for the stream.
if (handle == 0) {
try {
handle = new JackHandle;
}
catch (std::bad_alloc&) {
m_errorText = "airtaudio::api::Jack::probeDeviceOpen: error allocating JackHandle memory.";
goto error;
}
m_stream.apiHandle = (void *) handle;
handle->client = client;
}
handle->deviceName[mode] = deviceName;
// Allocate necessary internal buffers.
uint64_t bufferBytes;
bufferBytes = m_stream.nUserChannels[mode] * *bufferSize * formatBytes(m_stream.userFormat);
m_stream.userBuffer[mode] = (char *) calloc(bufferBytes, 1);
if (m_stream.userBuffer[mode] == NULL) {
m_errorText = "airtaudio::api::Jack::probeDeviceOpen: error allocating user buffer memory.";
goto error;
}
if (m_stream.doConvertBuffer[mode]) {
bool makeBuffer = true;
if (mode == OUTPUT)
bufferBytes = m_stream.nDeviceChannels[0] * formatBytes(m_stream.deviceFormat[0]);
else { // mode == INPUT
bufferBytes = m_stream.nDeviceChannels[1] * formatBytes(m_stream.deviceFormat[1]);
if (m_stream.mode == OUTPUT && m_stream.deviceBuffer) {
uint64_t bytesOut = m_stream.nDeviceChannels[0] * formatBytes(m_stream.deviceFormat[0]);
if (bufferBytes < bytesOut) makeBuffer = false;
}
}
if (makeBuffer) {
bufferBytes *= *bufferSize;
if (m_stream.deviceBuffer) free(m_stream.deviceBuffer);
m_stream.deviceBuffer = (char *) calloc(bufferBytes, 1);
if (m_stream.deviceBuffer == NULL) {
m_errorText = "airtaudio::api::Jack::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
// Allocate memory for the Jack ports (channels) identifiers.
handle->ports[mode] = (jack_port_t **) malloc (sizeof (jack_port_t *) * channels);
if (handle->ports[mode] == NULL) {
m_errorText = "airtaudio::api::Jack::probeDeviceOpen: error allocating port memory.";
goto error;
}
m_stream.device[mode] = device;
m_stream.channelOffset[mode] = firstChannel;
m_stream.state = STREAM_STOPPED;
m_stream.callbackInfo.object = (void *) this;
if (m_stream.mode == OUTPUT && mode == INPUT)
// We had already set up the stream for output.
m_stream.mode = DUPLEX;
else {
m_stream.mode = mode;
jack_set_process_callback(handle->client, jackCallbackHandler, (void *) &m_stream.callbackInfo);
jack_set_xrun_callback(handle->client, jackXrun, (void *) &handle);
jack_on_shutdown(handle->client, jackShutdown, (void *) &m_stream.callbackInfo);
}
// Register our ports.
char label[64];
if (mode == OUTPUT) {
for (uint32_t i=0; i<m_stream.nUserChannels[0]; i++) {
snprintf(label, 64, "outport %d", i);
handle->ports[0][i] = jack_port_register(handle->client,
(const char *)label,
JACK_DEFAULT_AUDIO_TYPE,
JackPortIsOutput,
0);
}
} else {
for (uint32_t i=0; i<m_stream.nUserChannels[1]; i++) {
snprintf(label, 64, "inport %d", i);
handle->ports[1][i] = jack_port_register(handle->client,
(const char *)label,
JACK_DEFAULT_AUDIO_TYPE,
JackPortIsInput,
0);
}
}
// Setup the buffer conversion information structure. We don't use
// buffers to do channel offsets, so we override that parameter
// here.
if (m_stream.doConvertBuffer[mode]) {
setConvertInfo(mode, 0);
}
return SUCCESS;
error:
if (handle) {
jack_client_close(handle->client);
if (handle->ports[0]) {
free(handle->ports[0]);
}
if (handle->ports[1]) {
free(handle->ports[1]);
}
delete handle;
m_stream.apiHandle = 0;
}
for (int32_t iii=0; iii<2; ++iii) {
if (m_stream.userBuffer[iii]) {
free(m_stream.userBuffer[iii]);
m_stream.userBuffer[iii] = 0;
}
}
if (m_stream.deviceBuffer) {
free(m_stream.deviceBuffer);
m_stream.deviceBuffer = 0;
}
return FAILURE;
}
void airtaudio::api::Jack::closeStream(void)
{
if (m_stream.state == STREAM_CLOSED) {
m_errorText = "airtaudio::api::Jack::closeStream(): no open stream to close!";
error(airtaudio::errorWarning);
return;
}
JackHandle *handle = (JackHandle *) m_stream.apiHandle;
if (handle) {
if (m_stream.state == STREAM_RUNNING)
jack_deactivate(handle->client);
jack_client_close(handle->client);
}
if (handle) {
if (handle->ports[0]) {
free(handle->ports[0]);
}
if (handle->ports[1]) {
free(handle->ports[1]);
}
delete handle;
m_stream.apiHandle = 0;
}
for (int32_t i=0; i<2; i++) {
if (m_stream.userBuffer[i]) {
free(m_stream.userBuffer[i]);
m_stream.userBuffer[i] = 0;
}
}
if (m_stream.deviceBuffer) {
free(m_stream.deviceBuffer);
m_stream.deviceBuffer = 0;
}
m_stream.mode = UNINITIALIZED;
m_stream.state = STREAM_CLOSED;
}
void airtaudio::api::Jack::startStream(void)
{
verifyStream();
if (m_stream.state == STREAM_RUNNING) {
m_errorText = "airtaudio::api::Jack::startStream(): the stream is already running!";
error(airtaudio::errorWarning);
return;
}
JackHandle *handle = (JackHandle *) m_stream.apiHandle;
int32_t result = jack_activate(handle->client);
if (result) {
m_errorText = "airtaudio::api::Jack::startStream(): unable to activate JACK client!";
goto unlock;
}
const char **ports;
// Get the list of available ports.
if (m_stream.mode == OUTPUT || m_stream.mode == DUPLEX) {
result = 1;
ports = jack_get_ports(handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
if (ports == NULL) {
m_errorText = "airtaudio::api::Jack::startStream(): error determining available JACK input ports!";
goto unlock;
}
// Now make the port connections. Since RtAudio wasn't designed to
// allow the user to select particular channels of a device, we'll
// just open the first "nChannels" ports with offset.
for (uint32_t i=0; i<m_stream.nUserChannels[0]; i++) {
result = 1;
if (ports[ m_stream.channelOffset[0] + i ])
result = jack_connect(handle->client, jack_port_name(handle->ports[0][i]), ports[ m_stream.channelOffset[0] + i ]);
if (result) {
free(ports);
m_errorText = "airtaudio::api::Jack::startStream(): error connecting output ports!";
goto unlock;
}
}
free(ports);
}
if (m_stream.mode == INPUT || m_stream.mode == DUPLEX) {
result = 1;
ports = jack_get_ports(handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput);
if (ports == NULL) {
m_errorText = "airtaudio::api::Jack::startStream(): error determining available JACK output ports!";
goto unlock;
}
// Now make the port connections. See note above.
for (uint32_t i=0; i<m_stream.nUserChannels[1]; i++) {
result = 1;
if (ports[ m_stream.channelOffset[1] + i ])
result = jack_connect(handle->client, ports[ m_stream.channelOffset[1] + i ], jack_port_name(handle->ports[1][i]));
if (result) {
free(ports);
m_errorText = "airtaudio::api::Jack::startStream(): error connecting input ports!";
goto unlock;
}
}
free(ports);
}
handle->drainCounter = 0;
handle->internalDrain = false;
m_stream.state = STREAM_RUNNING;
unlock:
if (result == 0) return;
error(airtaudio::errorSystemError);
}
void airtaudio::api::Jack::stopStream(void) {
verifyStream();
if (m_stream.state == STREAM_STOPPED) {
m_errorText = "airtaudio::api::Jack::stopStream(): the stream is already stopped!";
error(airtaudio::errorWarning);
return;
}
JackHandle *handle = (JackHandle *) m_stream.apiHandle;
if ( m_stream.mode == OUTPUT
|| m_stream.mode == DUPLEX) {
if (handle->drainCounter == 0) {
handle->drainCounter = 2;
std::unique_lock<std::mutex> lck(m_stream.mutex);
handle->condition.wait(lck);
}
}
jack_deactivate(handle->client);
m_stream.state = STREAM_STOPPED;
}
void airtaudio::api::Jack::abortStream(void)
{
verifyStream();
if (m_stream.state == STREAM_STOPPED) {
m_errorText = "airtaudio::api::Jack::abortStream(): the stream is already stopped!";
error(airtaudio::errorWarning);
return;
}
JackHandle *handle = (JackHandle *) m_stream.apiHandle;
handle->drainCounter = 2;
stopStream();
}
// This function will be called by a spawned thread when the user
// callback function signals that the stream should be stopped or
// aborted. It is necessary to handle it this way because the
// callbackEvent() function must return before the jack_deactivate()
// function will return.
static void jackStopStream(void *_ptr) {
airtaudio::CallbackInfo *info = (airtaudio::CallbackInfo *) _ptr;
airtaudio::api::Jack *object = (airtaudio::api::Jack *) info->object;
object->stopStream();
}
bool airtaudio::api::Jack::callbackEvent(uint64_t nframes)
{
if (m_stream.state == STREAM_STOPPED || m_stream.state == STREAM_STOPPING) return SUCCESS;
if (m_stream.state == STREAM_CLOSED) {
m_errorText = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
error(airtaudio::errorWarning);
return FAILURE;
}
if (m_stream.bufferSize != nframes) {
m_errorText = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
error(airtaudio::errorWarning);
return FAILURE;
}
CallbackInfo *info = (CallbackInfo *) &m_stream.callbackInfo;
JackHandle *handle = (JackHandle *) m_stream.apiHandle;
// Check if we were draining the stream and signal is finished.
if (handle->drainCounter > 3) {
m_stream.state = STREAM_STOPPING;
if (handle->internalDrain == true) {
new std::thread(jackStopStream, info);
} else {
handle->condition.notify_one();
}
return SUCCESS;
}
// Invoke user callback first, to get fresh output data.
if (handle->drainCounter == 0) {
airtaudio::AirTAudioCallback callback = (airtaudio::AirTAudioCallback) info->callback;
double streamTime = getStreamTime();
airtaudio::streamStatus status = 0;
if (m_stream.mode != INPUT && handle->xrun[0] == true) {
status |= OUTPUT_UNDERFLOW;
handle->xrun[0] = false;
}
if (m_stream.mode != OUTPUT && handle->xrun[1] == true) {
status |= INPUT_OVERFLOW;
handle->xrun[1] = false;
}
int32_t cbReturnValue = callback(m_stream.userBuffer[0], m_stream.userBuffer[1],
m_stream.bufferSize, streamTime, status, info->userData);
if (cbReturnValue == 2) {
m_stream.state = STREAM_STOPPING;
handle->drainCounter = 2;
new std::thread(jackStopStream, info);
return SUCCESS;
}
else if (cbReturnValue == 1) {
handle->drainCounter = 1;
handle->internalDrain = true;
}
}
jack_default_audio_sample_t *jackbuffer;
uint64_t bufferBytes = nframes * sizeof(jack_default_audio_sample_t);
if (m_stream.mode == OUTPUT || m_stream.mode == DUPLEX) {
if (handle->drainCounter > 1) { // write zeros to the output stream
for (uint32_t i=0; i<m_stream.nDeviceChannels[0]; i++) {
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[0][i], (jack_nframes_t) nframes);
memset(jackbuffer, 0, bufferBytes);
}
}
else if (m_stream.doConvertBuffer[0]) {
convertBuffer(m_stream.deviceBuffer, m_stream.userBuffer[0], m_stream.convertInfo[0]);
for (uint32_t i=0; i<m_stream.nDeviceChannels[0]; i++) {
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[0][i], (jack_nframes_t) nframes);
memcpy(jackbuffer, &m_stream.deviceBuffer[i*bufferBytes], bufferBytes);
}
}
else { // no buffer conversion
for (uint32_t i=0; i<m_stream.nUserChannels[0]; i++) {
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[0][i], (jack_nframes_t) nframes);
memcpy(jackbuffer, &m_stream.userBuffer[0][i*bufferBytes], bufferBytes);
}
}
if (handle->drainCounter) {
handle->drainCounter++;
goto unlock;
}
}
if ( m_stream.mode == INPUT
|| m_stream.mode == DUPLEX) {
if (m_stream.doConvertBuffer[1]) {
for (uint32_t i=0; i<m_stream.nDeviceChannels[1]; i++) {
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[1][i], (jack_nframes_t) nframes);
memcpy(&m_stream.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes);
}
convertBuffer(m_stream.userBuffer[1], m_stream.deviceBuffer, m_stream.convertInfo[1]);
} else {
// no buffer conversion
for (uint32_t i=0; i<m_stream.nUserChannels[1]; i++) {
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[1][i], (jack_nframes_t) nframes);
memcpy(&m_stream.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes);
}
}
}
unlock:
airtaudio::Api::tickStreamTime();
return SUCCESS;
}
//******************** End of __UNIX_JACK__ *********************//
#endif
Reference in New Issue View Git Blame Copy Permalink