/* -*- 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 #include #include #include #include #include #include #include #include #include //#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;iname); 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"); }