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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 <gnuradio/omnithread.h>
#include <string.h>
//#define LOGGING 0 // define to 0 or 1
#define SAMPLE_FORMAT paFloat32
typedef float sample_t;
// Number of portaudio buffers in the ringbuffer
static const unsigned int N_BUFFERS = 4;
static std::string
default_device_name ()
{
return gr_prefs::singleton()->get_string("audio_portaudio", "default_output_device", "");
}
void
audio_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.
*/
int
portaudio_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_sptr
audio_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)));
}
bool
audio_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.
*/
int
audio_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
}
void
audio_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));
}
void
audio_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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