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Viewing changes to linux/alsa/JackAlsaAdapter.h

  • Committer: Bazaar Package Importer
  • Author(s): Adrian Knoth
  • Date: 2011-03-31 13:54:50 UTC
  • mfrom: (1.1.3 upstream) (2.1.4 experimental)
  • Revision ID: james.westby@ubuntu.com-20110331135450-zafc1di024kzeu31
Tags: 1.9.7~dfsg-1
* New upstream version 1.9.7 (ALSA resume, new latency API)
* Build with --mixed on i386 to be compatible with amd64.
* Don't patch jack_connect for fast consecutive calls anymore, it's now
  using the same code as in jackd1 and waits for the port connection to
  appear.

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Lines of Context:
linux/alsa/JackAlsaAdapter.h
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            void* fOutputCardChannels[256];
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            //non-interleaved mod, floating point software buffers
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            float* fInputSoftChannels[256];
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            float* fOutputSoftChannels[256];
 
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            jack_default_audio_sample_t* fInputSoftChannels[256];
 
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            jack_default_audio_sample_t* fOutputSoftChannels[256];
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            //public methods ---------------------------------------------------------
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                return fBuffering;
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            }
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            float** inputSoftChannels()
 
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            jack_default_audio_sample_t** inputSoftChannels()
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            {
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                return fInputSoftChannels;
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            }
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            float** outputSoftChannels()
 
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            jack_default_audio_sample_t** outputSoftChannels()
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            {
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                return fOutputSoftChannels;
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            }
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                fInputParams    = 0;
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                fOutputParams   = 0;
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                fPeriod = 2;
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                fInputCardBuffer = 0;
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                fOutputCardBuffer = 0;
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                for ( int i = 0; i < 256; i++ )
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                {
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                    fInputCardChannels[i] = 0;
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                //get hardware input parameters
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                check_error ( snd_pcm_hw_params_malloc ( &fInputParams ) );
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                setAudioParams ( fInputDevice, fInputParams );
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                //get hardware output parameters
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                check_error ( snd_pcm_hw_params_malloc ( &fOutputParams ) )
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                setAudioParams ( fOutputDevice, fOutputParams );
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                // set the number of physical input and output channels close to what we need
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                fCardInputs     = fSoftInputs;
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                fCardOutputs    = fSoftOutputs;
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                snd_pcm_hw_params_set_channels_near(fInputDevice, fInputParams, &fCardInputs);
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                snd_pcm_hw_params_set_channels_near(fOutputDevice, fOutputParams, &fCardOutputs);
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                //set input/output param
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                check_error ( snd_pcm_hw_params ( fInputDevice,  fInputParams ) );
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                check_error ( snd_pcm_hw_params ( fOutputDevice, fOutputParams ) );
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                for ( unsigned int i = 0; i < fSoftInputs; i++ )
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                {
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                    fInputSoftChannels[i] = ( float* ) aligned_calloc ( fBuffering, sizeof ( float ) );
 
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                    fInputSoftChannels[i] = ( jack_default_audio_sample_t* ) aligned_calloc ( fBuffering, sizeof ( jack_default_audio_sample_t ) );
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                    for ( int j = 0; j < fBuffering; j++ )
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                        fInputSoftChannels[i][j] = 0.0;
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                }
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                for ( unsigned int i = 0; i < fSoftOutputs; i++ )
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                {
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                    fOutputSoftChannels[i] = ( float* ) aligned_calloc ( fBuffering, sizeof ( float ) );
 
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                    fOutputSoftChannels[i] = ( jack_default_audio_sample_t* ) aligned_calloc ( fBuffering, sizeof ( jack_default_audio_sample_t ) );
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                    for ( int j = 0; j < fBuffering; j++ )
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                        fOutputSoftChannels[i][j] = 0.0;
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                }
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                            short* buffer16b = ( short* ) fInputCardBuffer;
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                            for ( s = 0; s < fBuffering; s++ )
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                                for ( c = 0; c < fCardInputs; c++ )
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                                    fInputSoftChannels[c][s] = float ( buffer16b[c + s*fCardInputs] ) * ( 1.0/float ( SHRT_MAX ) );
 
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                                    fInputSoftChannels[c][s] = jack_default_audio_sample_t(buffer16b[c + s*fCardInputs]) * (jack_default_audio_sample_t(1.0)/jack_default_audio_sample_t(SHRT_MAX));
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                        }
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                        else   // SND_PCM_FORMAT_S32
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                        {
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                            int32_t* buffer32b = ( int32_t* ) fInputCardBuffer;
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                            for ( s = 0; s < fBuffering; s++ )
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                                for ( c = 0; c < fCardInputs; c++ )
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                                    fInputSoftChannels[c][s] = float ( buffer32b[c + s*fCardInputs] ) * ( 1.0/float ( INT_MAX ) );
 
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                                    fInputSoftChannels[c][s] = jack_default_audio_sample_t(buffer32b[c + s*fCardInputs]) * (jack_default_audio_sample_t(1.0)/jack_default_audio_sample_t(INT_MAX));
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                        }
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                        break;
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                    case SND_PCM_ACCESS_RW_NONINTERLEAVED :
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                            {
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                                chan16b = ( short* ) fInputCardChannels[c];
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                                for ( s = 0; s < fBuffering; s++ )
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                                    fInputSoftChannels[c][s] = float ( chan16b[s] ) * ( 1.0/float ( SHRT_MAX ) );
 
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                                    fInputSoftChannels[c][s] = jack_default_audio_sample_t(chan16b[s]) * (jack_default_audio_sample_t(1.0)/jack_default_audio_sample_t(SHRT_MAX));
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                            }
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                        }
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                        else   // SND_PCM_FORMAT_S32
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                            {
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                                chan32b = ( int32_t* ) fInputCardChannels[c];
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                                for ( s = 0; s < fBuffering; s++ )
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                                    fInputSoftChannels[c][s] = float ( chan32b[s] ) * ( 1.0/float ( INT_MAX ) );
 
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                                    fInputSoftChannels[c][s] = jack_default_audio_sample_t(chan32b[s]) * (jack_default_audio_sample_t(1.0)/jack_default_audio_sample_t(INT_MAX));
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                            }
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                        }
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                        break;
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                            {
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                                for ( unsigned int c = 0; c < fCardOutputs; c++ )
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                                {
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                                    float x = fOutputSoftChannels[c][f];
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                                    buffer16b[c + f * fCardOutputs] = short ( max ( min ( x, 1.0 ), -1.0 ) * float ( SHRT_MAX ) );
 
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                                    jack_default_audio_sample_t x = fOutputSoftChannels[c][f];
 
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                                    buffer16b[c + f * fCardOutputs] = short(max(min (x, jack_default_audio_sample_t(1.0)), jack_default_audio_sample_t(-1.0)) * jack_default_audio_sample_t(SHRT_MAX));
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                                }
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                            }
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                        }
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                            {
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                                for ( unsigned int c = 0; c < fCardOutputs; c++ )
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                                {
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                                    float x = fOutputSoftChannels[c][f];
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                                    buffer32b[c + f * fCardOutputs] = int32_t ( max ( min ( x, 1.0 ), -1.0 ) * float ( INT_MAX ) );
 
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                                    jack_default_audio_sample_t x = fOutputSoftChannels[c][f];
 
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                                    buffer32b[c + f * fCardOutputs] = int32_t(max(min(x, jack_default_audio_sample_t(1.0)), jack_default_audio_sample_t(-1.0)) * jack_default_audio_sample_t(INT_MAX));
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                                }
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                            }
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                        }
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                                short* chan16b = ( short* ) fOutputCardChannels[c];
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                                for ( f = 0; f < fBuffering; f++ )
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                                {
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                                    float x = fOutputSoftChannels[c][f];
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                                    chan16b[f] = short ( max ( min ( x,1.0 ), -1.0 ) * float ( SHRT_MAX ) ) ;
 
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                                    jack_default_audio_sample_t x = fOutputSoftChannels[c][f];
 
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                                    chan16b[f] = short(max(min (x, jack_default_audio_sample_t(1.0)), jack_default_audio_sample_t(-1.0)) * jack_default_audio_sample_t(SHRT_MAX));
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                                }
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                            }
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                        }
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                                int32_t* chan32b = ( int32_t* ) fOutputCardChannels[c];
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                                for ( f = 0; f < fBuffering; f++ )
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                                {
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                                    float x = fOutputSoftChannels[c][f];
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                                    chan32b[f] = int32_t ( max ( min ( x,1.0 ),-1.0 ) * float ( INT_MAX ) ) ;
 
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                                    jack_default_audio_sample_t x = fOutputSoftChannels[c][f];
 
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                                    chan32b[f] = int32_t(max(min(x, jack_default_audio_sample_t(1.0)), jack_default_audio_sample_t(-1.0)) * jack_default_audio_sample_t(INT_MAX));
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                                }
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                            }
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                        }