/* -*- c++ -*- */
/*
* Copyright 2006-2011,2013-2014 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
#ifdef _MSC_VER
#include <io.h>
#endif
#include "../audio_registry.h"
#include "portaudio_impl.h"
#include "portaudio_source.h"
#include <gnuradio/io_signature.h>
#include <gnuradio/prefs.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <iostream>
#include <stdexcept>
#ifdef _MSC_VER
#include <io.h>
#endif
namespace gr {
namespace audio {
source::sptr
portaudio_source_fcn(int sampling_rate, const std::string& device_name, bool ok_to_block)
{
return source::sptr(new portaudio_source(sampling_rate, device_name, ok_to_block));
}
//#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 prefs::singleton()->get_string("audio_portaudio", "default_input_device", "");
}
void portaudio_source::create_ringbuffer(void)
{
int bufsize_samples =
d_portaudio_buffer_size_frames * d_input_parameters.channelCount;
if (d_verbose) {
fprintf(stderr,
"ring buffer size = %d frames\n",
N_BUFFERS * bufsize_samples / d_input_parameters.channelCount);
}
// FYI, the buffer indices 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 copy framesPerBuffer frames from inputBuffer.
*/
int portaudio_source_callback(const void* inputBuffer,
void* outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void* arg)
{
portaudio_source* self = (portaudio_source*)arg;
int nchan = self->d_input_parameters.channelCount;
int nframes_to_copy = framesPerBuffer;
int nframes_room = self->d_writer->space_available() / nchan;
if (nframes_to_copy <= nframes_room) { // We've got room for the data ..
// if (LOGGING)
// self->d_log->printf("PAsrc cb: f/b = %4ld\n", framesPerBuffer);
// copy from input buffer to ringbuffer
{
gr::thread::scoped_lock(d_ringbuffer_mutex);
memcpy(self->d_writer->write_pointer(),
inputBuffer,
nframes_to_copy * nchan * sizeof(sample_t));
self->d_writer->update_write_pointer(nframes_to_copy * nchan);
// Tell the source thread there is new data in the ringbuffer.
self->d_ringbuffer_ready = true;
}
self->d_ringbuffer_cond.notify_one();
return paContinue;
}
else { // overrun
self->d_noverruns++;
ssize_t r = ::write(2, "aO", 2); // FIXME change to non-blocking call
if (r == -1) {
perror("audio_portaudio_source::portaudio_source_callback write error to "
"stderr.");
}
self->d_ringbuffer_ready = false;
self->d_ringbuffer_cond.notify_one(); // Tell the sink to get going!
return paContinue;
}
}
// ----------------------------------------------------------------
portaudio_source::portaudio_source(int sampling_rate,
const std::string device_name,
bool ok_to_block)
: sync_block("audio_portaudio_source",
io_signature::make(0, 0, 0),
io_signature::make(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(prefs::singleton()->get_bool("audio_portaudio", "verbose", false)),
d_portaudio_buffer_size_frames(0),
d_stream(0),
d_ringbuffer_mutex(),
d_ringbuffer_cond(),
d_ringbuffer_ready(false),
d_noverruns(0)
{
memset(&d_input_parameters, 0, sizeof(d_input_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)
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
device = Pa_GetDefaultInputDevice();
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;
for (i = 0; i < numDevices; i++) {
deviceInfo = Pa_GetDeviceInfo(i);
fprintf(stderr, "Testing device name: %s", deviceInfo->name);
if (deviceInfo->maxInputChannels <= 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);
}
}
d_input_parameters.device = device;
d_input_parameters.channelCount = deviceInfo->maxInputChannels;
d_input_parameters.sampleFormat = SAMPLE_FORMAT;
d_input_parameters.suggestedLatency = deviceInfo->defaultLowInputLatency;
d_input_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 output
// signature. The portaudio specs say that they'll accept any
// number of channels from 1 to max.
set_output_signature(
io_signature::make(1, deviceInfo->maxInputChannels, sizeof(sample_t)));
}
bool portaudio_source::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_input_parameters.channelCount = noutputs; // # of channels we're really using
#if 1
d_portaudio_buffer_size_frames =
(int)(0.0213333333 * d_sampling_rate + 0.5); // Force 512 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,
&d_input_parameters,
NULL, // No output
d_sampling_rate,
d_portaudio_buffer_size_frames,
paClipOff,
&portaudio_source_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_input_parameters.suggestedLatency * psi->sampleRate);
fprintf(stderr, "Latency = %7.4f, psi->sampleRate = %g\n",
d_input_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;
}
portaudio_source::~portaudio_source()
{
Pa_StopStream(d_stream); // wait for output to drain
Pa_CloseStream(d_stream);
Pa_Terminate();
}
int portaudio_source::work(int noutput_items,
gr_vector_const_void_star& input_items,
gr_vector_void_star& output_items)
{
float** out = (float**)&output_items[0];
const unsigned nchan =
d_input_parameters.channelCount; // # of channels == samples/frame
int k;
for (k = 0; k < noutput_items;) {
int nframes = d_reader->items_available() / nchan; // # of frames in ringbuffer
if (nframes == 0) { // no data right now...
if (k > 0) // If we've produced anything so far, return that
return k;
if (d_ok_to_block) {
gr::thread::scoped_lock guard(d_ringbuffer_mutex);
while (d_ringbuffer_ready == false)
d_ringbuffer_cond.wait(guard); // block here, then try again
continue;
}
assert(k == 0);
// There's no data and we're not allowed to block.
// (A USRP is most likely controlling the pacing through the pipeline.)
// This is an underrun. The scheduler wouldn't have called us if it
// had anything better to do. Thus we really need to produce some amount
// of "fill".
//
// There are lots of options for comfort noise, etc.
// FIXME We'll fill with zeros for now. Yes, it will "click"...
// Fill with some frames of zeros
{
gr::thread::scoped_lock guard(d_ringbuffer_mutex);
int nf = std::min(noutput_items - k, (int)d_portaudio_buffer_size_frames);
for (int i = 0; i < nf; i++) {
for (unsigned int c = 0; c < nchan; c++) {
out[c][k + i] = 0;
}
}
k += nf;
d_ringbuffer_ready = false;
return k;
}
}
// We can read the smaller of the request and what's in the buffer.
{
gr::thread::scoped_lock guard(d_ringbuffer_mutex);
int nf = std::min(noutput_items - k, nframes);
const float* p = (const float*)d_reader->read_pointer();
for (int i = 0; i < nf; i++) {
for (unsigned int c = 0; c < nchan; c++) {
out[c][k + i] = *p++;
}
}
d_reader->update_read_pointer(nf * nchan);
k += nf;
d_ringbuffer_ready = false;
}
}
return k; // tell how many we actually did
}
void portaudio_source::output_error_msg(const char* msg, int err)
{
fprintf(stderr,
"audio_portaudio_source[%s]: %s: %s\n",
d_device_name.c_str(),
msg,
Pa_GetErrorText(err));
}
void portaudio_source::bail(const char* msg, int err)
{
output_error_msg(msg, err);
throw std::runtime_error("audio_portaudio_source");
}
} /* namespace audio */
} /* namespace gr */