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📄 audio_portaudio_sink.cc

📁 这是用python语言写的一个数字广播的信号处理工具包。利用它
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/* -*- c++ -*- *//* * Copyright 2006 Free Software Foundation, Inc. *  * This file is part of GNU Radio *  * GNU Radio is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 3, or (at your option) * any later version. *  * GNU Radio is distributed in he hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the * GNU General Public License for more details. *  * You should have received a copy of the GNU General Public License * along with GNU Radio; see the file COPYING.  If not, write to * the Free Software Foundation, Inc., 51 Franklin Street, * Boston, MA 02110-1301, USA. */#ifdef HAVE_CONFIG_H#include "config.h"#endif#include <audio_portaudio_sink.h>#include <gr_io_signature.h>#include <gr_prefs.h>#include <stdio.h>#include <iostream>#include <unistd.h>#include <stdexcept>#include <gri_portaudio.h>#include <omnithread.h>#include <string.h>#define	LOGGING   0		// define to 0 or 1#define SAMPLE_FORMAT 		paFloat32typedef float sample_t;// Number of portaudio buffers in the ringbufferstatic const unsigned int N_BUFFERS = 4;static std::string default_device_name (){  return gr_prefs::singleton()->get_string("audio_portaudio", "default_output_device", "");}voidaudio_portaudio_sink::create_ringbuffer(void){  int bufsize_samples = d_portaudio_buffer_size_frames * d_output_parameters.channelCount;    if (d_verbose)    fprintf(stderr,"ring buffer size  = %d frames\n",	    N_BUFFERS*bufsize_samples/d_output_parameters.channelCount);  // FYI, the buffer indicies are in units of samples.  d_writer = gr_make_buffer(N_BUFFERS * bufsize_samples, sizeof(sample_t));  d_reader = gr_buffer_add_reader(d_writer, 0);}/* * This routine will be called by the PortAudio engine when audio is needed. * It may called at interrupt level on some machines so don't do anything * that could mess up the system like calling malloc() or free(). * * Our job is to write framesPerBuffer frames into outputBuffer. */intportaudio_sink_callback (const void *inputBuffer,			 void *outputBuffer,			 unsigned long framesPerBuffer,			 const PaStreamCallbackTimeInfo* timeInfo,			 PaStreamCallbackFlags statusFlags,			 void *arg){  audio_portaudio_sink *self = (audio_portaudio_sink *)arg;  int nreqd_samples =    framesPerBuffer * self->d_output_parameters.channelCount;  int navail_samples = self->d_reader->items_available();    if (nreqd_samples <= navail_samples){  // We've got enough data...    if (LOGGING)      self->d_log->printf("PAsink cb: f/b = %4ld\n", framesPerBuffer);    // copy from ringbuffer into output buffer    memcpy(outputBuffer,	   self->d_reader->read_pointer(),	   nreqd_samples * sizeof(sample_t));    self->d_reader->update_read_pointer(nreqd_samples);	     // Tell the sink thread there is new room in the ringbuffer.    self->d_ringbuffer_ready.post();    return paContinue;  }  else {			// underrun    if (LOGGING)      self->d_log->printf("PAsink cb: f/b = %4ld UNDERRUN\n", framesPerBuffer);    self->d_nunderuns++;    ::write(2, "aU", 2);	// FIXME change to non-blocking call    // FIXME we should transfer what we've got and pad the rest    memset(outputBuffer, 0, nreqd_samples * sizeof(sample_t));    self->d_ringbuffer_ready.post();  // Tell the sink to get going!    return paContinue;  }}// ----------------------------------------------------------------audio_portaudio_sink_sptraudio_portaudio_make_sink (int sampling_rate, const std::string dev, bool ok_to_block){  return audio_portaudio_sink_sptr (new audio_portaudio_sink (sampling_rate,							      dev, ok_to_block));}audio_portaudio_sink::audio_portaudio_sink(int sampling_rate,					   const std::string device_name,					   bool ok_to_block)  : gr_sync_block ("audio_portaudio_sink",		   gr_make_io_signature(0, 0, 0),		   gr_make_io_signature(0, 0, 0)),    d_sampling_rate(sampling_rate),    d_device_name(device_name.empty() ? default_device_name() : device_name),    d_ok_to_block(ok_to_block),    d_verbose(gr_prefs::singleton()->get_bool("audio_portaudio", "verbose", false)),    d_portaudio_buffer_size_frames(0),    d_stream(0),    d_ringbuffer_ready(1, 1),		// binary semaphore    d_nunderuns(0){  memset(&d_output_parameters, 0, sizeof(d_output_parameters));  if (LOGGING)    d_log = gri_logger::singleton();  PaError 	      err;  int     	      i, numDevices;  PaDeviceIndex	      device = 0;  const PaDeviceInfo *deviceInfo = NULL;  err = Pa_Initialize();  if (err != paNoError) {    bail ("Initialize failed", err);  }  if (d_verbose)    gri_print_devices();  numDevices = Pa_GetDeviceCount();  if (numDevices < 0)    bail("Pa Device count failed", 0);  if (numDevices == 0)    bail("no devices available", 0);  if (d_device_name.empty())   {    // FIXME Get smarter about picking something    fprintf(stderr,"\nUsing Default Device\n");    device = Pa_GetDefaultOutputDevice();    deviceInfo = Pa_GetDeviceInfo(device);    fprintf(stderr,"%s is the chosen device using %s as the host\n",	    deviceInfo->name, Pa_GetHostApiInfo(deviceInfo->hostApi)->name);  }  else  {    bool found = false;    fprintf(stderr,"\nTest Devices\n");    for (i=0;i<numDevices;i++) {      deviceInfo = Pa_GetDeviceInfo( i );      fprintf(stderr,"Testing device name: %s",deviceInfo->name);      if (deviceInfo->maxOutputChannels <= 0) {	fprintf(stderr,"\n");	continue;      }      if (strstr(deviceInfo->name, d_device_name.c_str())){	fprintf(stderr,"  Chosen!\n");	device = i;	fprintf(stderr,"%s using %s as the host\n",d_device_name.c_str(),		Pa_GetHostApiInfo(deviceInfo->hostApi)->name), fflush(stderr);	found = true;	deviceInfo = Pa_GetDeviceInfo(device);	i = numDevices; 	// force loop exit      }      else	fprintf(stderr,"\n"),fflush(stderr);    }    if (!found){      bail("Failed to find specified device name", 0);      exit(1);    }  }  d_output_parameters.device = device;  d_output_parameters.channelCount     = deviceInfo->maxOutputChannels;  d_output_parameters.sampleFormat     = SAMPLE_FORMAT;  d_output_parameters.suggestedLatency = deviceInfo->defaultLowOutputLatency;  d_output_parameters.hostApiSpecificStreamInfo = NULL;  // We fill in the real channelCount in check_topology when we know  // how many inputs are connected to us.  // Now that we know the maximum number of channels (allegedly)  // supported by the h/w, we can compute a reasonable input  // signature.  The portaudio specs say that they'll accept any  // number of channels from 1 to max.  set_input_signature(gr_make_io_signature(1, deviceInfo->maxOutputChannels,					   sizeof (sample_t)));}boolaudio_portaudio_sink::check_topology (int ninputs, int noutputs){  PaError err;  if (Pa_IsStreamActive(d_stream))  {      Pa_CloseStream(d_stream);      d_stream = 0;      d_reader.reset();		// boost::shared_ptr for d_reader = 0      d_writer.reset();		// boost::shared_ptr for d_write = 0  }  d_output_parameters.channelCount = ninputs;	// # of channels we're really using#if 1  d_portaudio_buffer_size_frames = (int)(0.0213333333  * d_sampling_rate + 0.5);  // Force 1024 frame buffers at 48000  fprintf(stderr, "Latency = %8.5f, requested sampling_rate = %g\n", // Force latency to 21.3333333.. ms	  0.0213333333, (double)d_sampling_rate);#endif  err = Pa_OpenStream(&d_stream,		      NULL,			// No input		      &d_output_parameters,		      d_sampling_rate,		      d_portaudio_buffer_size_frames,		      paClipOff,		      &portaudio_sink_callback,		      (void*)this);  if (err != paNoError) {    output_error_msg ("OpenStream failed", err);    return false;  }#if 0    const PaStreamInfo *psi = Pa_GetStreamInfo(d_stream);  d_portaudio_buffer_size_frames = (int)(d_output_parameters.suggestedLatency  * psi->sampleRate);  fprintf(stderr, "Latency = %7.4f, psi->sampleRate = %g\n",	  d_output_parameters.suggestedLatency, psi->sampleRate);#endif  fprintf(stderr, "d_portaudio_buffer_size_frames = %d\n", d_portaudio_buffer_size_frames);  assert(d_portaudio_buffer_size_frames != 0);  create_ringbuffer();  err = Pa_StartStream(d_stream);  if (err != paNoError) {    output_error_msg ("StartStream failed", err);    return false;  }  return true;}audio_portaudio_sink::~audio_portaudio_sink (){  Pa_StopStream(d_stream);	// wait for output to drain  Pa_CloseStream(d_stream);  Pa_Terminate();}/* * This version consumes everything sent to it, blocking if required. * I think this will allow us better control of the total buffering/latency * in the audio path. */intaudio_portaudio_sink::work (int noutput_items,			    gr_vector_const_void_star &input_items,			    gr_vector_void_star &output_items){  const float **in = (const float **) &input_items[0];  const unsigned nchan = d_output_parameters.channelCount; // # of channels == samples/frame  int k;  for (k = 0; k < noutput_items; ){    int nframes = d_writer->space_available() / nchan;	// How much space in ringbuffer    if (nframes == 0){			// no room...      if (d_ok_to_block){	d_ringbuffer_ready.wait();	// block here, then try again	continue;      }      else {	// There's no room and we're not allowed to block.	// (A USRP is most likely controlling the pacing through the pipeline.)	// We drop the samples on the ground, and say we processed them all ;)	//	// FIXME, there's probably room for a bit more finesse here.	return noutput_items;      }    }    // We can write the smaller of the request and the room we've got    int nf = std::min(noutput_items - k, nframes);    float *p = (float *) d_writer->write_pointer();    for (int i = 0; i < nf; i++){      for (unsigned int c = 0; c < nchan; c++){	*p++ = in[c][k + i];      }    }    d_writer->update_write_pointer(nf * nchan);    k += nf;  }  return k;  // tell how many we actually did}voidaudio_portaudio_sink::output_error_msg (const char *msg, int err){  fprintf (stderr, "audio_portaudio_sink[%s]: %s: %s\n",	   d_device_name.c_str (), msg, Pa_GetErrorText(err));}voidaudio_portaudio_sink::bail (const char *msg, int err) throw (std::runtime_error){  output_error_msg (msg, err);  throw std::runtime_error ("audio_portaudio_sink");}

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