X-Git-Url: https://git.sesse.net/?p=pitch;a=blobdiff_plain;f=linux_audio.cpp;h=c07e6bb289a543419d608063796ada457157c66c;hp=b76909e7588fa2048bfa6b7ba8bf2e0abb257847;hb=a500a1c540da51bdb686bdb9657b354c4238bcb2;hpb=14e6df5b242c72acc44cf8f68a56ee35dd5c43cf

diff --git a/linux_audio.cpp b/linux_audio.cpp
index b76909e..c07e6bb 100644
--- a/linux_audio.cpp
+++ b/linux_audio.cpp
@@ -6,58 +6,102 @@
 #include <fcntl.h>
 #include <sys/ioctl.h>
 #include <linux/soundcard.h>
-
+#include <alsa/asoundlib.h>
 #include "linux_audio.h"
 
-int get_dsp_fd(int sample_rate, int fft_length, int overlap)
+snd_pcm_t *get_dsp_handle(int sample_rate)
 {
-	int fd = open("/dev/dsp", O_RDWR);
-	if (fd == -1) {
-		perror("/dev/dsp");
+	int err;
+	snd_pcm_t *handle;
+	snd_pcm_hw_params_t *hw_params;
+
+	if ((err = snd_pcm_open(&handle, "plughw:0,0", SND_PCM_STREAM_CAPTURE, 0)) < 0) {
+		fprintf(stderr, "cannot open audio device plughw:0,0: %s\n", 
+			snd_strerror(err));
 		exit(1);
 	}
-	
-	ioctl(3, SNDCTL_DSP_RESET, 0);
-	
-	int fmt = AFMT_S16_LE;   // FIXME
-	ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
 
-	int chan = 1;
-	ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &chan);
-	
-	int rate = sample_rate;
-	ioctl(fd, SOUND_PCM_WRITE_RATE, &rate);
+	if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
+		fprintf(stderr, "cannot allocate hardware parameter structure: %s\n",
+			snd_strerror(err));
+		exit(1);
+	}
 
-	int max_fragments = 2;
-	int frag_shift = ffs(fft_length / overlap) - 1;
-	int fragments = (max_fragments << 16) | frag_shift;
-        ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fragments);
-	
-	ioctl(3, SNDCTL_DSP_SYNC, 0);
-	
-	return fd;
+	if ((err = snd_pcm_hw_params_any(handle, hw_params)) < 0) {
+		fprintf(stderr, "cannot initialize hardware parameter structure: %s\n",
+			snd_strerror(err));
+		exit(1);
+	}
+
+	if ((err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
+		fprintf(stderr, "cannot set access type: %s\n",
+			snd_strerror(err));
+		exit(1);
+	}
+
+	if ((err = snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {  // FIXME
+		fprintf(stderr, "cannot set sample format: %s\n",
+			snd_strerror(err));
+		exit(1);
+	}
+
+	if ((err = snd_pcm_hw_params_set_rate(handle, hw_params, sample_rate, 0)) < 0) {
+		fprintf(stderr, "cannot set sample rate: %s\n",
+			snd_strerror(err));
+		exit(1);
+	}
+
+	if ((err = snd_pcm_hw_params_set_channels(handle, hw_params, 1)) < 0) {
+		fprintf(stderr, "cannot set channel count: %s\n",
+			snd_strerror(err));
+		exit(1);
+	}
+
+	if ((err = snd_pcm_hw_params(handle, hw_params)) < 0) {
+		fprintf(stderr, "cannot set parameters: %s\n",
+			snd_strerror(err));
+		exit(1);
+	}
+
+	snd_pcm_hw_params_free(hw_params);
+
+	if ((err = snd_pcm_prepare(handle)) < 0) {
+		fprintf(stderr, "cannot prepare audio interface for use: %s\n",
+			snd_strerror(err));
+		exit(1);
+	}
+
+	return handle;
 }
 
 #if 1
-void read_chunk(int fd, short *in, unsigned num_samples)
+void read_chunk(snd_pcm_t *handle, short *in, unsigned num_samples)
 {
 	int ret;
+	int samples_left = num_samples;
 
-	ret = read(fd, in, num_samples * sizeof(short));
-	if (ret == 0) {
-		printf("EOF\n");
-		exit(0);
-	}
-
-	if (ret != int(num_samples * sizeof(short))) {
-		// blah
-		perror("read");
-		exit(1);
+	while (samples_left > 0) {
+		ret = snd_pcm_readi(handle, in, samples_left);
+		if (ret == 0) {
+			printf("EOF\n");
+			exit(0);
+		}
+		if (ret == -EPIPE) {
+			fprintf(stderr, "ALSA underrun\n");
+			snd_pcm_prepare(handle);
+			continue;
+		}
+		if (ret < 0) {
+			fprintf(stderr, "snd_pcm_readi: %s\n", snd_strerror(ret));
+			exit(1);
+		}
+		in += ret;
+		samples_left -= ret;
 	}
 }
 #else
 // make a pure 440hz sine for testing
-void read_chunk(int fd, short *in, unsigned num_samples)
+void read_chunk(snd_pcm_t *handle, short *in, unsigned num_samples)
 {
 	static double theta = 0.0;
 	for (unsigned i = 0; i < num_samples; ++i) {