jackaudio
/
jack2
mirror of https://github.com/jackaudio/jack2
1
0
Fork 0
Code Releases 45 Activity
Browse Source

Address review comments from shaun-tierney 

Removed a number of commented sections
Made a few sample types explicitly sized
Removed debugging statements
pull/244/head
Andrew Cooper 10 years ago
parent
17a87a5f33
commit
33000e7bc6
7 changed files with 43 additions and 407 deletions
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. +0
    -1
      qnx/cycles.h
  2. +31
    -49
      qnx/ioaudio/JackIoAudioAdapter.cpp
  3. +3
    -41
      qnx/ioaudio/JackIoAudioAdapter.h
  4. +8
    -215
      qnx/ioaudio/ioaudio_driver.c
  5. +0
    -56
      tests/qnx/TestIoAudioAdapter.cpp
  6. +0
    -44
      tests/qnx/TestIoAudioInterface.cpp
  7. +1
    -1
      tests/test.cpp

+ 0
- 1
qnx/cycles.h View File

@@ -23,7 +23,6 @@
#define __jack_cycles_h__

#include <sys/neutrino.h>
//#include <intttypes.h>

typedef uint64_t cycles_t;
#define get_cycles() ClockCycles()


+ 31
- 49
qnx/ioaudio/JackIoAudioAdapter.cpp View File

@@ -166,7 +166,7 @@ namespace Jack
//get hardware input parameters
fInputParams->mode = SND_PCM_MODE_BLOCK;
fInputParams->channel = SND_PCM_CHANNEL_CAPTURE;
//fInputParams->sync = Not Supported
//fInputParams->sync = Not Supported by io-audio

// Check supported formats, preferring in order: float, sint32, sint16, plugin converted
if( inputInfo.formats & SND_PCM_FMT_FLOAT )
@@ -192,11 +192,11 @@ namespace Jack
fInputParams->format.interleave = 1;
fInputParams->format.rate = fParams.fFrequency;
fInputParams->format.voices = fNumInputPorts;
//fInputParams->digital = Not Implemented
//fInputParams->digital = Not currently implemented in io-audio
fInputParams->start_mode = SND_PCM_START_DATA;
fInputParams->stop_mode = SND_PCM_STOP_STOP;
//fInputParams->time = 1;
//fInputParams->ust_time
//fInputParams->time = 1; // If set, the driver offers the time when the transfer began (gettimeofday() format)
//fInputParams->ust_time = 1; // If set, the driver offers the time when the transfer began (UST format)
fInputParams->buf.block.frag_size =
snd_pcm_format_size( fInputParams->format.format,
fParams.fBuffering * fNumInputPorts );
@@ -219,9 +219,6 @@ namespace Jack
return err;
}

//set nonblocking mode
//check_error(snd_pcm_nonblock_mode(fInputDevice, 1));

//get params record with actual values
fInputSetup->channel = SND_PCM_CHANNEL_CAPTURE;
check_error( snd_pcm_channel_setup( fInputDevice,
@@ -289,7 +286,7 @@ namespace Jack
check_error( snd_pcm_malloc_struct( &fOutputParams ) );
fOutputParams->channel = SND_PCM_CHANNEL_PLAYBACK;
fOutputParams->mode = SND_PCM_MODE_BLOCK;
//fOutputParams->sync = Not Supported
//fOutputParams->sync = Not Supported by io-audio

// Check supported formats, preferring in order: float, sint32, sint16, plugin converted
if( outputInfo.formats & SND_PCM_FMT_FLOAT )
@@ -315,11 +312,11 @@ namespace Jack
fOutputParams->format.interleave = 1;
fOutputParams->format.rate = fParams.fFrequency;
fOutputParams->format.voices = fNumOutputPorts;
//fOutputParams->digital = Not Implemented
//fOutputParams->digital = Not currently implemented by io-audio
fOutputParams->start_mode = SND_PCM_START_DATA;
fOutputParams->stop_mode = SND_PCM_STOP_STOP;
fOutputParams->time = 1;
//fOutputParams->ust_time
//fOutputParams->time = 1; // If set, the driver offers the time when the transfer began (gettimeofday() format)
//fOutputParams->ust_time = 1; // If set, the driver offers the time when the transfer began (UST Format)
fOutputParams->buf.block.frag_size =
snd_pcm_format_size( fOutputParams->format.format,
fParams.fBuffering * fNumOutputPorts );
@@ -341,9 +338,6 @@ namespace Jack
return err;
}

// //set nonblocking mode
// check_error(snd_pcm_nonblock_mode(fOutputDevice, 1));

//get params record with actual values
check_error( snd_pcm_channel_setup( fOutputDevice,
fOutputSetup ) );
@@ -392,14 +386,6 @@ namespace Jack
if( fJackOutputBuffers[i] )
free( fJackOutputBuffers[i] );

// for (unsigned int i = 0; i < fNumInputPorts; i++)
// if (fInputCardBuffer[i])
// free(fInputCardBuffer[i]);
//
// for (unsigned int i = 0; i < fNumOutputPorts; i++)
// if (fOutputCardBuffer[i])
// free(fOutputCardBuffer[i]);

if( fInputCardBuffer )
free( fInputCardBuffer );
if( fOutputCardBuffer )
@@ -450,7 +436,7 @@ namespace Jack
count );
if( fInputFormat.format == SND_PCM_SFMT_S16 )
{
short* buffer16b = (short*)fInputCardBuffer;
int16_t* buffer16b = (int16_t*)fInputCardBuffer;
for( s = 0; s < fInputBufferFrames; s++ )
for( c = 0; c < fNumInputPorts; c++ )
fJackInputBuffers[c][s] =
@@ -493,10 +479,10 @@ namespace Jack
count );
if( fInputFormat.format == SND_PCM_SFMT_S16 )
{
short* buffer16b;
int16_t* buffer16b;
for( c = 0; c < fNumInputPorts; c++ )
{
buffer16b = (short*)fInputCardBuffer;
buffer16b = (int16_t*)fInputCardBuffer;
for( s = 0; s < fInputBufferFrames; s++ )
fJackInputBuffers[c][s] =
jack_default_audio_sample_t( buffer16b[s] )
@@ -547,7 +533,7 @@ namespace Jack
case 1:
if( fOutputFormat.format == SND_PCM_SFMT_S16 )
{
short* buffer16b = (short*)fOutputCardBuffer;
int16_t* buffer16b = (int16_t*)fOutputCardBuffer;
for( f = 0; f < fOutputBufferFrames; f++ )
{
for( c = 0; c < fNumOutputPorts; c++ )
@@ -602,7 +588,7 @@ namespace Jack
case 0:
if( fOutputFormat.format == SND_PCM_SFMT_S16 )
{
short* buffer16b = (short*)fOutputCardBuffer;
int16_t* buffer16b = (int16_t*)fOutputCardBuffer;
for( c = 0; c < fNumOutputPorts; c++ )
{
for( f = 0; f < fOutputBufferFrames; f++ )
@@ -617,7 +603,7 @@ namespace Jack
}
}
}
else
else // SND_PCM_FORMAT_S32
{
int32_t* buffer32b = (int32_t*)fOutputCardBuffer;
for( c = 0; c < fNumOutputPorts; c++ )
@@ -700,7 +686,7 @@ namespace Jack
snd_ctl_hw_info_t card_info;
snd_ctl_t* ctl_handle;

//display info
//display info
jack_info( "Audio Interface Description :" );
jack_info(
"Sampling Frequency : %d, Sample Format : %s, buffering : %d, nperiod : %d",
@@ -715,7 +701,7 @@ namespace Jack
fNumInputPorts,
fNumOutputPorts );

//get audio card info and display
//get audio card info and display
int card = snd_card_name( fParams.fInputCardName );
check_error( snd_ctl_open( &ctl_handle,
card ) );
@@ -723,7 +709,7 @@ namespace Jack
&card_info ) );
printCardInfo( &card_info );

//display input/output streams info
//display input/output streams info
if( fNumInputPorts > 0 )
printHWParams( fInputParams );
if( fNumOutputPorts > 0 )
@@ -832,9 +818,6 @@ namespace Jack
param->value.ui );
fAudioInterface.fParams.fPeriod = param->value.ui;
break;
// case 'd':
// fAudioInterface.fParams.fInputCardName = strdup(param->value.str);
// break;
case 'r':
jack_info( "fFrequency = %d",
param->value.ui );
@@ -860,28 +843,25 @@ namespace Jack
break;
}
}

// fAudioInterface.setInputs(fCaptureChannels);
// fAudioInterface.setOutputs(fPlaybackChannels);
}

int JackIoAudioAdapter::Open()
{
//open audio interface
//open audio interface
if( fAudioInterface.open() )
return -1;

//start adapter thread
//start adapter thread
if( fThread.StartSync() < 0 )
{
jack_error( "Cannot start audioadapter thread" );
return -1;
}

//display card info
//display card info
fAudioInterface.longinfo();

//turn the thread realtime
//turn the thread realtime
fThread.AcquireRealTime( GetEngineControl()->fClientPriority );
return 0;
}
@@ -894,9 +874,9 @@ namespace Jack
switch( fThread.GetStatus() )
{

// Kill the thread in Init phase
// Kill the thread in Init phase
case JackThread::kStarting:
case JackThread::kIniting:
case JackThread::kIniting:
if( fThread.Kill() < 0 )
{
jack_error( "Cannot kill thread" );
@@ -904,7 +884,7 @@ namespace Jack
}
break;

// Stop when the thread cycle is finished
// Stop when the thread cycle is finished
case JackThread::kRunning:
if( fThread.Stop() < 0 )
{
@@ -945,6 +925,7 @@ namespace Jack
&channel_info );
snd_pcm_close( device );

// Determine number of capture channels from card or parameter
if( 0 == fAudioInterface.fParams.fCardInputVoices )
{
fCaptureChannels = channel_info.max_voices;
@@ -960,7 +941,7 @@ namespace Jack

fAudioInterface.fNumInputPorts = fCaptureChannels;

//ringbuffers
// Create capture channel ringbuffers
fCaptureRingBuffer = new JackResampler*[fCaptureChannels];
for( int i = 0; i < fCaptureChannels; i++ )
{
@@ -996,6 +977,7 @@ namespace Jack
&channel_info );
snd_pcm_close( device );

// Determine number of playback channels from card or parameter
if( 0 == fAudioInterface.fParams.fCardOutputVoices )
{
fPlaybackChannels = channel_info.max_voices;
@@ -1012,7 +994,7 @@ namespace Jack

fAudioInterface.fNumOutputPorts = fPlaybackChannels;

//ringbuffers
// Create playback channel ringbuffers
fPlaybackRingBuffer = new JackResampler*[fPlaybackChannels];
for( int i = 0; i < fPlaybackChannels; i++ )
{
@@ -1047,7 +1029,7 @@ namespace Jack

bool JackIoAudioAdapter::Init()
{
//fill the hardware buffers
//fill the hardware buffers
for( unsigned int i = 0; i < fAudioInterface.fParams.fPeriod; i++ )
fAudioInterface.write();
return true;
@@ -1055,7 +1037,7 @@ namespace Jack

bool JackIoAudioAdapter::Execute()
{
//read data from audio interface
//read data from audio interface
if( fAudioInterface.read() < 0 )
return false;

@@ -1063,7 +1045,7 @@ namespace Jack
fAudioInterface.fJackOutputBuffers,
fAdaptedBufferSize );

//write data to audio interface
//write data to audio interface
if( fAudioInterface.write() < 0 )
return false;



+ 3
- 41
qnx/ioaudio/JackIoAudioAdapter.h View File

@@ -120,7 +120,7 @@ namespace Jack
};

/**
* An ALSA audio interface
* An io-audio client interface
*/
class AudioInterface
{
@@ -138,11 +138,6 @@ namespace Jack
//samples info
snd_pcm_format_t fInputFormat;
snd_pcm_format_t fOutputFormat;
// uint32_t fSampleAccess;

//channels
// const char* fCaptureName;
// const char* fPlaybackName;

unsigned int fNumInputPorts;
unsigned int fNumOutputPorts;
@@ -151,49 +146,16 @@ namespace Jack
jack_nframes_t fInputBufferFrames;
jack_nframes_t fOutputBufferFrames;

//interleaved mode audiocard buffers
// audiocard buffers
void* fInputCardBuffer;
void* fOutputCardBuffer;

// //non-interleaved mode audiocard buffers
// void* fCardInputBuffers[NUM_BUFFERS];
// void* fCardOutputBuffers[NUM_BUFFERS];
//
// //non-interleaved mod, floating point software buffers
// jack_default_audio_sample_t* fJackInputBuffers[NUM_BUFFERS];
// jack_default_audio_sample_t* fJackOutputBuffers[NUM_BUFFERS];

//non-interleaved mod, floating point software buffers
// floating point JACK buffers
jack_default_audio_sample_t** fJackInputBuffers;
jack_default_audio_sample_t** fJackOutputBuffers;

//public methods ---------------------------------------------------------

// const char* cardName()
// {
// return fInputCardName;
// }
//
// int frequency()
// {
// return fFrequency;
// }
//
// int buffering()
// {
// return fBuffering;
// }

// jack_default_audio_sample_t** inputSoftChannels()
// {
// return fJackInputBuffers;
// }
//
// jack_default_audio_sample_t** outputSoftChannels()
// {
// return fJackOutputBuffers;
// }

AudioInterface(const AudioParam& ap = AudioParam());

AudioInterface(jack_nframes_t buffer_size, jack_nframes_t sample_rate);


+ 8
- 215
qnx/ioaudio/ioaudio_driver.c View File

@@ -132,10 +132,6 @@ static int parse_commandline(

if( opt >= 0 )
{
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: Parsed option %d (%s)",
opt,
opts[opt] );
switch( opt )
{
case 0: /* server: JACK server name */
@@ -148,9 +144,6 @@ static int parse_commandline(
{
jack_card->server = ado_strdup( value );
}
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: argument 'server' parsed as %s",
jack_card->server );
break;
case 1: /* name: JACK client name */
if( NULL == value )
@@ -162,10 +155,6 @@ static int parse_commandline(
{
jack_card->name = ado_strdup( value );
}
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: argument 'name' parsed as %s",
jack_card->name );

break;
case 2: /* type: io-audio channel type */
if( NULL != value )
@@ -185,9 +174,6 @@ static int parse_commandline(
"type option value not 'c' or 'p'; defaulting to 'p'" );
jack_card->channel_type = SND_PCM_CHANNEL_PLAYBACK;
}
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: argument 'type' parsed as %d",
jack_card->channel_type );
break;
case 3: /* voices: number of voices in io-audio channel and JACK ports */
if( NULL != value )
@@ -222,10 +208,6 @@ static int parse_commandline(
ado_error( "voices option given without value; failing" );
return EINVAL;
}
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: argument 'voices' with value '%s' parsed as %d",
value ? value : "NULL",
jack_card->voices );
break;
default:
break;
@@ -238,10 +220,6 @@ static int parse_commandline(
value );
}
}

ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: finished parsing arguments" );

ado_free( argdup );
return EOK;
}
@@ -270,86 +248,6 @@ int32_t cb_aquire(
*why_failed = SND_PCM_PARAMS_NO_CHANNEL;
return ( EAGAIN );
}
ado_debug( DB_LVL_DRIVER,
"config:" );
ado_debug( DB_LVL_DRIVER,
" format:" );
ado_debug( DB_LVL_DRIVER,
" interleave: %d",
config->format.interleave );
ado_debug( DB_LVL_DRIVER,
" format: {id: %d, name: '%s'}",
config->format.format,
snd_pcm_get_format_name( config->format.format ) );
ado_debug( DB_LVL_DRIVER,
" rate: %d",
config->format.rate );
ado_debug( DB_LVL_DRIVER,
" voices: %d",
config->format.voices );
ado_debug( DB_LVL_DRIVER,
" mode.block:" );
ado_debug( DB_LVL_DRIVER,
" frag_size: %d",
config->mode.block.frag_size );
ado_debug( DB_LVL_DRIVER,
" frags_min: %d",
config->mode.block.frags_min );
ado_debug( DB_LVL_DRIVER,
" frags_max: %d",
config->mode.block.frags_max );
ado_debug( DB_LVL_DRIVER,
" frags_total: %d",
config->mode.block.frags_total );
ado_debug( DB_LVL_DRIVER,
" dmabuf:" );
ado_debug( DB_LVL_DRIVER,
" addr: %x",
config->dmabuf.addr );
ado_debug( DB_LVL_DRIVER,
" phys_addr: %x",
config->dmabuf.phys_addr );
ado_debug( DB_LVL_DRIVER,
" size: %d",
config->dmabuf.size );
ado_debug( DB_LVL_DRIVER,
" name: '%s'",
config->dmabuf.name );
ado_debug( DB_LVL_DRIVER,
" mixer_device: %d",
config->mixer_device );
ado_debug( DB_LVL_DRIVER,
" mixer_eid:" );
ado_debug( DB_LVL_DRIVER,
" type: %d",
config->mixer_eid.type );
ado_debug( DB_LVL_DRIVER,
" name: '%s'",
config->mixer_eid.name );
ado_debug( DB_LVL_DRIVER,
" index: %d",
config->mixer_eid.index );
ado_debug( DB_LVL_DRIVER,
" weight: %d",
config->mixer_eid.weight );
ado_debug( DB_LVL_DRIVER,
" mixer_gid:" );
ado_debug( DB_LVL_DRIVER,
" type: %d",
config->mixer_gid.type );
ado_debug( DB_LVL_DRIVER,
" name: '%s'",
config->mixer_gid.name );
ado_debug( DB_LVL_DRIVER,
" index: %d",
config->mixer_gid.index );
ado_debug( DB_LVL_DRIVER,
" weight: %d",
config->mixer_gid.weight );

ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: creating ringbuffer of %u bytes",
config->dmabuf.size );

/*
* Create shared memory region for ringbuffer
@@ -358,22 +256,6 @@ int32_t cb_aquire(
config->dmabuf.name,
ADO_SHM_DMA_SAFE,
&config->dmabuf.phys_addr );
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: ado_shm_alloc() has finished. " );
ado_debug( DB_LVL_DRIVER,
" dmabuf:" );
ado_debug( DB_LVL_DRIVER,
" addr: %x",
config->dmabuf.addr );
ado_debug( DB_LVL_DRIVER,
" phys_addr: %x",
config->dmabuf.phys_addr );
ado_debug( DB_LVL_DRIVER,
" size: %d",
config->dmabuf.size );
ado_debug( DB_LVL_DRIVER,
" name: '%s'",
config->dmabuf.name );

/*
* Set up JACK ringbuffer structure to use SHM region instead of own
@@ -403,10 +285,11 @@ int32_t cb_release(
PCM_SUBCHN_CONTEXT_T *pc,
ado_pcm_config_t *config )
{
pthread_mutex_lock( &jack_card->process_lock );

ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: release()" );

pthread_mutex_lock( &jack_card->process_lock );

jack_card->subchn.go = 0;

jack_ringbuffer_reset( &jack_card->ringbuffer );
@@ -517,22 +400,8 @@ int jack_process(
if( jack_card->subchn.go
&& ( 0 == pthread_mutex_trylock( &jack_card->process_lock ) ) )
{
ado_debug( DB_LVL_PCM,
"deva-ctrl-jack: jack_process( %d )",
nframes );
ado_debug( DB_LVL_PCM,
" total_size: %d",
total_size );

int voices = jack_card->subchn.pcm_config->format.voices;
ado_debug( DB_LVL_PCM,
" voices: %d",
voices );

size_t size_per_voice = total_size / voices;
ado_debug( DB_LVL_PCM,
" size_per_voice: %d",
size_per_voice );

void* jack_buf[voices];
for( v = 0; v < voices; ++v )
@@ -544,14 +413,8 @@ int jack_process(
for( size_completed = 0; size_completed < total_size; size_completed +=
size_per_voice )
{
ado_debug( DB_LVL_PCM,
" size_completed: %d",
size_completed );
for( v = 0; v < voices; ++v )
{
ado_debug( DB_LVL_PCM,
" voice: %d",
v );
/*
* Advance ringbuffer write pointer on the assumption that io-audio has filled in nframes of data
*/
@@ -565,16 +428,6 @@ int jack_process(
if( SND_PCM_SFMT_FLOAT_LE
== jack_card->subchn.pcm_config->format.format )
{
ado_debug( DB_LVL_PCM,
" ringbuffer:" );
ado_debug( DB_LVL_PCM,
" - { buf: %x, len: %d }",
read_buf[0].buf,
read_buf[0].len );
ado_debug( DB_LVL_PCM,
" - { buf: %x, len: %d }",
read_buf[1].buf,
read_buf[1].len );
jack_ringbuffer_read( &jack_card->ringbuffer,
jack_buf[v],
size_per_voice );
@@ -586,19 +439,6 @@ int jack_process(
size_t remaining = size_per_voice / sizeof(sample_t);
read_buf[0].len /= sizeof(sample_t);
read_buf[1].len /= sizeof(sample_t);
ado_debug( DB_LVL_PCM,
" samples_remaining: %d",
remaining );
ado_debug( DB_LVL_PCM,
" ringbuffer:" );
ado_debug( DB_LVL_PCM,
" - { buf: %x, len: %d }",
read_buf[0].buf,
read_buf[0].len );
ado_debug( DB_LVL_PCM,
" - { buf: %x, len: %d }",
read_buf[1].buf,
read_buf[1].len );

int32_t* src = (sample_t*)read_buf[0].buf;
sample_t* dest = (sample_t*)jack_buf[v];
@@ -608,22 +448,8 @@ int jack_process(
{
dest[s] = ( (sample_t)src[s] ) * ( (sample_t)1.0 )
/ ( (sample_t)INT_MAX );
// ado_debug(DB_LVL_PCM, " %x : %d : %f", &src[s], src[s], dest[s]);
}
remaining -= amt;
ado_debug( DB_LVL_PCM,
" samples_remaining: %d",
remaining );
ado_debug( DB_LVL_PCM,
" ringbuffer:" );
ado_debug( DB_LVL_PCM,
" - { buf: %x, len: %d }",
read_buf[0].buf,
read_buf[0].len );
ado_debug( DB_LVL_PCM,
" - { buf: %x, len: %d }",
read_buf[1].buf,
read_buf[1].len );
if( remaining > 0 )
{
src = (int32_t*)read_buf[1].buf;
@@ -645,13 +471,14 @@ int jack_process(
}
dma_interrupt( jack_card->subchn.pcm_subchn );
}
ado_debug( DB_LVL_PCM,
" size_completed: %d",
size_completed );
pthread_mutex_unlock( &jack_card->process_lock );
}
else
{
/*
* In the case where there is no subchannel or we can't aquire the ringbuffer lock,
* write zeroes to the JACK buffers rather than let stale data sit in them.
*/
for( v = 0; v < jack_card->voices; ++v )
{
void* jack_buf = jack_port_get_buffer( jack_card->ports[v],
@@ -684,6 +511,7 @@ void jack_shutdown(
jack_card_t* card = (jack_card_t*)arg;

/*
* TODO
* Find a way to trigger io-audio to call ctrl_destroy() to clean up
*/
}
@@ -701,8 +529,6 @@ int ctrl_init(
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: ctrl_init" );

ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: allocating card" );
jack_card = ado_calloc( 1,
sizeof(jack_card_t) );
if( NULL == jack_card )
@@ -718,21 +544,14 @@ int ctrl_init(
pthread_mutex_init( &jack_card->process_lock,
NULL );

ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: setting card name" );
ado_card_set_shortname( card,
"jack_card" );
ado_card_set_longname( card,
"jack_card",
0x1000 );

ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: parsing arguments '%s'",
args );
retval = parse_commandline( jack_card,
args );
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: returned from parsing arguments" );
if( EOK != retval )
{
ado_error(
@@ -753,10 +572,6 @@ int ctrl_init(
client_name_size );
return -1;
}
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: Opening jack client with name '%s' on server '%s'",
jack_card->name,
jack_card->server ? jack_card->server : "DEFAULT" );
jack_status_t open_status;
if( NULL != jack_card->server )
{
@@ -782,8 +597,6 @@ int ctrl_init(
/*
* Set JACK and ADO processing callbacks
*/
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: registering callbacks" );
jack_set_process_callback( jack_card->client,
jack_process,
jack_card );
@@ -795,9 +608,6 @@ int ctrl_init(
* Allocate jack_port_t array
*/
int voices = jack_card->voices;
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: allocating array of %d ports",
voices );
jack_card->ports = ado_calloc( voices,
sizeof(jack_port_t*) );

@@ -813,9 +623,6 @@ int ctrl_init(
port_name_size,
PORTNAME_FMT,
( i + 1 ) );
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: registering port %d",
( i + 1 ) );
jack_card->ports[i] = jack_port_register( jack_card->client,
portname,
JACK_DEFAULT_AUDIO_TYPE,
@@ -827,20 +634,13 @@ int ctrl_init(
/*
* Initialize Capabilities
*/
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: initializing driver capabilities" );
jack_card->caps.chn_flags = SND_PCM_CHNINFO_BLOCK
| SND_PCM_CHNINFO_NONINTERLEAVE;
jack_card->caps.formats = 0xFFFF;
jack_card->caps.formats = SND_PCM_FMT_FLOAT_LE | SND_PCM_FMT_S32_LE;
// jack_card->caps.formats = SND_PCM_FMT_FLOAT_LE;

jack_nframes_t rate = jack_get_sample_rate( jack_card->client );
uint32_t rateflag = ado_pcm_rate2flag( rate );
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: sample rate of JACK: %u; flag: 0x%x",
rate,
rateflag );
jack_card->caps.rates = rateflag;

jack_card->caps.min_voices = 1;
@@ -851,16 +651,11 @@ int ctrl_init(
*/
jack_card->fragsize = jack_get_buffer_size( jack_card->client )
* sizeof(sample_t);
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: fragment is %u bytes",
jack_card->fragsize );
jack_card->caps.min_fragsize = jack_card->fragsize;
jack_card->caps.max_fragsize = jack_card->fragsize;
jack_card->caps.max_dma_size = 0;
jack_card->caps.max_frags = 0;

ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: initializing driver callbacks" );
jack_card->funcs.capabilities2 = cb_capabilities;
jack_card->funcs.aquire = cb_aquire;
jack_card->funcs.release = cb_release;
@@ -871,8 +666,6 @@ int ctrl_init(
/*
* Create Audio PCM Device
*/
ado_debug( DB_LVL_DRIVER,
"deva-ctrl-jack: creating PCM device" );
retval = ado_pcm_create( card,
"JACK io-audio driver",
( jack_card->channel_type


+ 0
- 56
tests/qnx/TestIoAudioAdapter.cpp View File

@@ -1,56 +0,0 @@
#include <gtest/gtest.h>
#include "JackIoAudioAdapter.h"

namespace Jack
{
class TestIoAudioAdapter: public ::testing::Test
{
public:
TestIoAudioAdapter()
{

}

};

TEST_F( TestIoAudioAdapter, Open_Success )
{

}

TEST_F( TestIoAudioAdapter, Close_Success )
{

}

TEST_F( TestIoAudioAdapter, Create_Success )
{

}

TEST_F( TestIoAudioAdapter, Destroy_Success )
{

}

TEST_F( TestIoAudioAdapter, SetSampleRate_Success )
{

}

TEST_F( TestIoAudioAdapter, SetBufferSize_Success )
{

}

TEST_F( TestIoAudioAdapter, Init_Success )
{

}

TEST_F( TestIoAudioAdapter, Execute_Success )
{

}

}

+ 0
- 44
tests/qnx/TestIoAudioInterface.cpp View File

@@ -1,44 +0,0 @@
#include <gtest/gtest.h>
#include "JackIoAudioAdapter.h"

namespace Jack
{
class TestIoAudioInterface: public ::testing::Test
{
public:
TestIoAudioInterface()
{

}
};

TEST_F( TestIoAudioInterface, Open_Success )
{

}

TEST_F( TestIoAudioInterface, Close_Success )
{

}

TEST_F( TestIoAudioInterface, Read_Success )
{

}

TEST_F( TestIoAudioInterface, Write_Success )
{

}

TEST_F( TestIoAudioInterface, ShortInfo_Success )
{

}

TEST_F( TestIoAudioInterface, LongInfo_Success )
{

}
}

+ 1
- 1
tests/test.cpp View File

@@ -479,7 +479,7 @@ int process4(jack_nframes_t nframes, void *arg)
jack_nframes_t delta_time = cur_time - last_time;

Log("calling process4 callback : jack_frame_time = %ld delta_time = %ld\n", cur_time, delta_time);
if (delta_time > 0 && (jack_nframes_t)abs((long int)delta_time - (long int)cur_buffer_size) > tolerance) {
if (delta_time > 0 && (jack_nframes_t)abs((int64_t)delta_time - (int64_t)cur_buffer_size) > tolerance) {
printf("!!! ERROR !!! jack_frame_time seems to return incorrect values cur_buffer_size = %d, delta_time = %d tolerance %d\n", cur_buffer_size, delta_time, tolerance);
}



Write Preview
Loading…
Cancel
Save