Add ALSA output.

Cleanups.

AudioOutput.cc

@@ -5,6 +5,8 @@
 #include <fcntl.h>
 #include <algorithm>
 
+#include <alsa/asoundlib.h>
+
 #include "SoftFM.h"
 #include "AudioOutput.h"
 
@@ -124,28 +126,88 @@ private:
 // TODO
 };
 
-
-/** Write audio data to ALSA device. */
-class AlsaAudioOutput
-{
-public:
-
-    /**
-     * Construct ALSA output stream.
-     *
-     * dename       :: ALSA PCM device
-     * samplerate   :: audio sample rate in Hz
-     * stereo       :: true if the output stream contains stereo data
-     */
-    AlsaAudioOutput(const std::string& devname,
-                    unsigned int samplerate,
-                    bool stereo);
-
-    ~AlsaAudioOutput();
-    bool write(const SampleVector& samples);
-    std::string error();
-
-private:
-    // TODO
-};
 #endif
+
+/* ****************  class AlsaAudioOutput  **************** */
+
+// Construct ALSA output stream.
+AlsaAudioOutput::AlsaAudioOutput(const std::string& devname,
+                                 unsigned int samplerate,
+                                 bool stereo)
+{
+    m_pcm = NULL;
+    m_nchannels = stereo ? 2 : 1;
+
+    int r = snd_pcm_open(&m_pcm, devname.c_str(),
+                         SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+
+    if (r < 0) {
+        m_error = "can not open PCM device '" + devname + "' (" +
+                  strerror(-r) + ")";
+        m_zombie = true;
+        return;
+    }
+
+    snd_pcm_nonblock(m_pcm, 0);
+
+    r = snd_pcm_set_params(m_pcm,
+                           SND_PCM_FORMAT_S16_LE,
+                           SND_PCM_ACCESS_RW_INTERLEAVED,
+                           m_nchannels,
+                           samplerate,
+                           1,               // allow soft resampling
+                           500000);         // latency in us
+
+    if (r < 0) {
+        m_error = "can not set PCM parameters (";
+        m_error += strerror(-r);
+        m_error += ")";
+        m_zombie = true;
+    }
+}
+
+
+// Destructor.
+AlsaAudioOutput::~AlsaAudioOutput()
+{
+    // Close device.
+    if (m_pcm != NULL) {
+        snd_pcm_close(m_pcm);
+    }
+}
+
+
+// Write audio data.
+bool AlsaAudioOutput::write(const SampleVector& samples)
+{
+    if (m_zombie)
+        return false;
+
+    // Convert samples to bytes.
+    samplesToInt16(samples, m_bytebuf);
+
+    // Write data.
+    unsigned int p = 0;
+    unsigned int n = samples.size() / m_nchannels;
+    unsigned int framesize = 2 * m_nchannels;
+    while (p < n) {
+
+        int k = snd_pcm_writei(m_pcm,
+                               m_bytebuf.data() + p * framesize, n - p);
+        if (k < 0) {
+            m_error = "write failed (";
+            m_error += strerror(errno);
+            m_error += ")";
+            // After an underrun, ALSA keeps returning error codes until we
+            // explicitly fix the stream.
+            snd_pcm_recover(m_pcm, k, 0);
+            return false;
+        } else {
+            p += k;
+        }
+    }
+
+    return true;
+}
+
+/* end */

AudioOutput.h

@@ -119,7 +119,9 @@ public:
     bool write(const SampleVector& samples);
 
 private:
-    // TODO
+    unsigned int         m_nchannels;
+    struct _snd_pcm *    m_pcm;
+    std::vector<uint8_t> m_bytebuf;
 };
 
 #endif

Makefile

@@ -2,8 +2,6 @@
 
 # ----- Tweak these settings to configure for your system
 
-# TODO : -D_FILE_OFFSET_BITS=64
-
 CROSS         =
 CFLAGS_OPT    = -O2 -ffast-math -ftree-vectorize
 CFLAGS_DEBUG  = -g
@@ -12,6 +10,7 @@ CFLAGS_PATH   = -I/home/joris/test/rtl-sdr/inst/include
 CFLAGS_EXTRA  =
 LDFLAGS_PATH  = -L/home/joris/test/rtl-sdr/inst/lib
 LDFLAGS_EXTRA =
+LIBS_ALSA     = -lasound
 LIBS_RTLSDR   = /home/joris/test/rtl-sdr/inst/lib/librtlsdr.a -lusb-1.0
 LIBS_EXTRA    =
 
@@ -19,10 +18,11 @@ LIBS_EXTRA    =
 
 
 CXX = $(CROSS)g++
-CXXFLAGS = -std=c++11 -Wall $(CFLAGS_OPT) $(CFLAGS_DEBUG) \
+CXXFLAGS = -std=c++11 -Wall -D_FILE_OFFSET_BITS=64 \
+           $(CFLAGS_OPT) $(CFLAGS_DEBUG) \
            $(CFLAGS_ARCH) $(CFLAGS_PATH) $(CFLAGS_EXTRA)
 LDFLAGS = $(LDFLAGS_PATH) $(LDFLAGS_EXTRA)
-LDLIBS  = $(LIBS_RTLSDR) $(LIBS_EXTRA)
+LDLIBS  = $(LIBS_ALSA) $(LIBS_RTLSDR) $(LIBS_EXTRA)
 
 OBJS	= RtlSdrSource.o Filter.o FmDecode.o AudioOutput.o main.o
 

main.cc

@@ -171,6 +171,9 @@ void write_output_data(AudioOutput *output, DataBuffer<Sample> *buf,
         // Get samples from buffer and write to output.
         SampleVector samples = buf->pull();
         output->write(samples);
+        if (!(*output)) {
+            fprintf(stderr, "ERROR: AudioOutput: %s\n", output->error().c_str());
+        }
     }
 }
 
@@ -222,12 +225,11 @@ bool parse_opt(const char *s, double& v)
 
 int main(int argc, char **argv)
 {
-    int c;
     double  freq    = -1;
     int     devidx  = 0;
     double  ifrate  = 1.0e6;
     int     pcmrate = 48000;
-    int     stereo  = 1;
+    bool    stereo  = true;
     enum OutputMode { MODE_RAW, MODE_WAV, MODE_ALSA };
     OutputMode outmode = MODE_ALSA;
     string  filename;
@@ -237,7 +239,22 @@ int main(int argc, char **argv)
     fprintf(stderr,
             "SoftFM - Software decoder for FM broadcast radio with RTL-SDR\n");
 
-    while ((c = getopt(argc, argv, "f:d:s:r:MR:W:P::b:")) >= 0) {
+    const struct option longopts[] = {
+        { "freq",       1, NULL, 'f' },
+        { "dev",        1, NULL, 'd' },
+        { "ifrate",     1, NULL, 's' },
+        { "pcmrate",    1, NULL, 'r' },
+        { "mono",       0, NULL, 'M' },
+        { "raw",        1, NULL, 'R' },
+        { "wav",        1, NULL, 'W' },
+        { "play",       2, NULL, 'P' },
+        { "buffer",     1, NULL, 'b' },
+        { NULL,         0, NULL, 0 } };
+
+    int c, longindex;
+    while ((c = getopt_long(argc, argv,
+                            "f:d:s:r:MR:W:P::b:",
+                            longopts, &longindex)) >= 0) {
         switch (c) {
             case 'f':
                 if (!parse_opt(optarg, freq) || freq <= 0) {
@@ -260,7 +277,7 @@ int main(int argc, char **argv)
                 }
                 break;
             case 'M':
-                stereo = 0;
+                stereo = false;
                 break;
             case 'R':
                 outmode = MODE_RAW;
@@ -377,9 +394,16 @@ int main(int argc, char **argv)
             audio_output.reset(new RawAudioOutput(filename));
             break;
         case MODE_WAV:
-        case MODE_ALSA:
-            // TODO
             abort();
+        case MODE_ALSA:
+            audio_output.reset(new AlsaAudioOutput(devname, pcmrate, stereo));
+            break;
+    }
+
+    if (!(*audio_output)) {
+        fprintf(stderr, "ERROR: AudioOutput: %s\n",
+                        audio_output->error().c_str());
+        exit(1);
     }
 
     // If buffering enabled, start background output thread.
@@ -424,8 +448,8 @@ int main(int argc, char **argv)
 // TODO : investigate I/Q imbalance to fix Radio4 noise
 // TODO : investigate if PLL receiver is better than phase discriminator at weak reception
 
+// TODO : show mono/stereo (switching)
 
-// TODO : show mono/stereo
         fprintf(stderr,
                 "\rblk=%6d  freq=%8.4fMHz  IF=%+5.1fdB  BB=%+5.1fdB  audio=%+5.1fdB ",
                 block,
@@ -461,6 +485,7 @@ int main(int argc, char **argv)
     // Join background threads.
     source_thread.join();
     if (outputbuf_samples > 0) {
+        output_buffer.push_end();
         output_thread.join();
     }