155 lines
5.2 KiB
C++
155 lines
5.2 KiB
C++
/*
|
|
Copyright (C) 2020 dec05eba
|
|
|
|
This program 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 of the License, or
|
|
(at your option) any later version.
|
|
|
|
This program is distributed in the 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 this program. If not, see <https://www.gnu.org/licenses/>.
|
|
*/
|
|
|
|
#include "../include/sound.hpp"
|
|
|
|
#include <stdlib.h>
|
|
#include <stdio.h>
|
|
|
|
#ifdef PULSEAUDIO
|
|
#include <pulse/simple.h>
|
|
#include <pulse/error.h>
|
|
|
|
int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int num_channels, unsigned int period_frame_size) {
|
|
pa_sample_spec ss;
|
|
ss.format = PA_SAMPLE_S16LE;
|
|
ss.rate = 48000;
|
|
ss.channels = num_channels;
|
|
int error;
|
|
|
|
pa_simple *pa_handle = pa_simple_new(nullptr, "gpu-screen-recorder", PA_STREAM_RECORD, name, "record", &ss, nullptr, nullptr, &error);
|
|
if(!pa_handle) {
|
|
fprintf(stderr, "pa_simple_new() failed: %s. Audio input device %s might not be valid\n", pa_strerror(error), name);
|
|
return -1;
|
|
}
|
|
|
|
int buffer_size = period_frame_size * 2 * num_channels; // 2 bytes/sample, @num_channels channels
|
|
void *buffer = malloc(buffer_size);
|
|
if(!buffer) {
|
|
fprintf(stderr, "failed to allocate buffer for audio\n");
|
|
pa_simple_free(pa_handle);
|
|
return -1;
|
|
}
|
|
|
|
fprintf(stderr, "Using pulseaudio\n");
|
|
|
|
device->handle = pa_handle;
|
|
device->buffer = buffer;
|
|
device->buffer_size = buffer_size;
|
|
device->frames = period_frame_size;
|
|
return 0;
|
|
}
|
|
|
|
void sound_device_close(SoundDevice *device) {
|
|
pa_simple_free((pa_simple*)device->handle);
|
|
free(device->buffer);
|
|
}
|
|
|
|
int sound_device_read_next_chunk(SoundDevice *device, void **buffer) {
|
|
int error;
|
|
if(pa_simple_read((pa_simple*)device->handle, device->buffer, device->buffer_size, &error) < 0) {
|
|
fprintf(stderr, "pa_simple_read() failed: %s\n", pa_strerror(error));
|
|
return -1;
|
|
}
|
|
*buffer = device->buffer;
|
|
return device->frames;
|
|
}
|
|
#else
|
|
#define ALSA_PCM_NEW_HW_PARAMS_API
|
|
#include <alsa/asoundlib.h>
|
|
|
|
int sound_device_get_by_name(SoundDevice *device, const char *name, unsigned int num_channels, unsigned int period_frame_size) {
|
|
int rc;
|
|
snd_pcm_t *handle;
|
|
|
|
rc = snd_pcm_open(&handle, name, SND_PCM_STREAM_CAPTURE, 0);
|
|
if(rc < 0) {
|
|
fprintf(stderr, "unable to open pcm device 'default', reason: %s\n", snd_strerror(rc));
|
|
return rc;
|
|
}
|
|
|
|
snd_pcm_hw_params_t *params;
|
|
snd_pcm_hw_params_alloca(¶ms);
|
|
// Fill the params with default values
|
|
snd_pcm_hw_params_any(handle, params);
|
|
// Interleaved mode
|
|
snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
|
|
// Signed 16--bit little-endian format
|
|
snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
|
|
snd_pcm_hw_params_set_channels(handle, params, num_channels);
|
|
|
|
// 48000 bits/second samling rate (DVD quality)
|
|
unsigned int val = 48000;
|
|
int dir;
|
|
snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
|
|
|
|
snd_pcm_uframes_t frames = period_frame_size;
|
|
snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir);
|
|
|
|
// Write the parmeters to the driver
|
|
rc = snd_pcm_hw_params(handle, params);
|
|
if(rc < 0) {
|
|
fprintf(stderr, "unable to set hw parameters, reason: %s\n", snd_strerror(rc));
|
|
snd_pcm_close(handle);
|
|
return rc;
|
|
}
|
|
|
|
// Use a buffer large enough to hold one period
|
|
snd_pcm_hw_params_get_period_size(params, &frames, &dir);
|
|
int buffer_size = frames * 2 * num_channels; // 2 bytes/sample, @num_channels channels
|
|
void *buffer = malloc(buffer_size);
|
|
if(!buffer) {
|
|
fprintf(stderr, "failed to allocate buffer for audio\n");
|
|
snd_pcm_close(handle);
|
|
return -1;
|
|
}
|
|
|
|
fprintf(stderr, "Using alsa\n");
|
|
|
|
device->handle = handle;
|
|
device->buffer = buffer;
|
|
device->buffer_size = buffer_size;
|
|
device->frames = frames;
|
|
return 0;
|
|
}
|
|
|
|
void sound_device_close(SoundDevice *device) {
|
|
/* TODO: Is this also needed in @sound_device_get_by_name on failure? */
|
|
// TODO: This has been commented out since it causes the thread to block forever. Why?
|
|
//snd_pcm_drain((snd_pcm_t*)device->handle);
|
|
snd_pcm_close((snd_pcm_t*)device->handle);
|
|
free(device->buffer);
|
|
}
|
|
|
|
int sound_device_read_next_chunk(SoundDevice *device, void **buffer) {
|
|
int rc = snd_pcm_readi((snd_pcm_t*)device->handle, device->buffer, device->frames);
|
|
if (rc == -EPIPE) {
|
|
/* overrun */
|
|
fprintf(stderr, "overrun occured\n");
|
|
snd_pcm_prepare((snd_pcm_t*)device->handle);
|
|
return rc;
|
|
} else if(rc < 0) {
|
|
fprintf(stderr, "failed to read from sound device, reason: %s\n", snd_strerror(rc));
|
|
return rc;
|
|
} else if (rc != (int)device->frames) {
|
|
fprintf(stderr, "short read, read %d frames\n", rc);
|
|
}
|
|
*buffer = device->buffer;
|
|
return rc;
|
|
}
|
|
#endif
|