RFC: time synchronised playback

audio.c

@@ -28,6 +28,9 @@
 #include <string.h>
 #include "audio.h"
 #include "config.h"
+#include "common.h"
+#include "player.h"
+#include "rtp.h"
 
 #ifdef CONFIG_SNDIO
 extern audio_output audio_sndio;
@@ -61,6 +64,7 @@ static audio_output *outputs[] = {
     NULL
 };
 
+long long audio_delay;
 
 audio_output *audio_get_output(char *name) {
     audio_output **out;
@@ -89,3 +93,35 @@ void audio_ls_outputs(void) {
         (*out)->help();
     }
 }
+
+//gets called for generic outputs
+long long audio_get_delay(void) {
+    return audio_delay;
+}
+
+void audio_estimate_delay(audio_output *output) {
+    signed short *silence;
+    int frame_size = 200;
+    long long base_time, cur_time, last_time, frame_time, frame_time_limit;
+
+    silence = malloc(4 * frame_size);
+    memset(silence, 0, 4 * frame_size);
+    frame_time = (frame_size * 1000000) / 44100;
+    frame_time_limit = frame_time * 2 / 3;
+
+    base_time = tstp_us();
+    last_time = base_time;
+    int loop = 0;
+    while (loop < 1000) {
+        output->play(silence, frame_size);
+        cur_time = tstp_us();
+        if ((cur_time - last_time) > frame_time_limit)
+            break;
+        last_time = cur_time;
+        loop++;
+    }
+    debug(3, "totaltime %lld, last loop time %lld, loop %d\n", last_time-base_time,cur_time - last_time, loop);
+    audio_delay = (loop * frame_time) - (last_time - base_time);
+    debug(2,"Generic output delay %lld\n", audio_delay);
+    free(silence);
+}

audio.h

@@ -17,9 +17,14 @@ typedef struct {
 
     // may be NULL, in which case soft volume is applied
     void (*volume)(double vol);
+
+    //time in us it takes before a new sample is output
+    long long (*get_delay)(void);
 } audio_output;
 
 audio_output *audio_get_output(char *name);
 void audio_ls_outputs(void);
+long long audio_get_delay(void);
+void audio_estimate_delay(audio_output *output);
 
 #endif //_AUDIO_H

audio_alsa.c

@@ -40,6 +40,7 @@ static void start(int sample_rate);
 static void play(short buf[], int samples);
 static void stop(void);
 static void volume(double vol);
+static long long get_delay(void);
 
 audio_output audio_alsa = {
     .name = "alsa",
@@ -49,7 +50,8 @@ audio_output audio_alsa = {
     .start = &start,
     .stop = &stop,
     .play = &play,
-    .volume = NULL
+    .volume = NULL,
+    .get_delay = &get_delay
 };
 
 static snd_pcm_t *alsa_handle = NULL;
@@ -65,6 +67,8 @@ static char *alsa_mix_dev = NULL;
 static char *alsa_mix_ctrl = "Master";
 static int alsa_mix_index = 0;
 
+static int device_sample_rate;
+
 static void help(void) {
     printf("    -d output-device    set the output device [default*|...]\n"
            "    -t mixer-type       set the mixer type [software*|hardware]\n"
@@ -153,9 +157,10 @@ static void deinit(void) {
 static void start(int sample_rate) {
     if (sample_rate != 44100)
         die("Unexpected sample rate!");
+    device_sample_rate = sample_rate;
 
     int ret, dir = 0;
-    snd_pcm_uframes_t frames = 64;
+    snd_pcm_uframes_t period_size,  buffer_size;
     ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0);
     if (ret < 0)
         die("Alsa initialization failed: unable to open pcm device: %s\n", snd_strerror(ret));
@@ -166,10 +171,52 @@ static void start(int sample_rate) {
     snd_pcm_hw_params_set_format(alsa_handle, alsa_params, SND_PCM_FORMAT_S16);
     snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2);
     snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_params, (unsigned int *)&sample_rate, &dir);
-    snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params, &frames, &dir);
-    ret = snd_pcm_hw_params(alsa_handle, alsa_params);
-    if (ret < 0)
-        die("unable to set hw parameters: %s\n", snd_strerror(ret));
+
+    // setting period and buffer is a simplified version of what XBMC does
+    snd_pcm_hw_params_get_period_size_max(alsa_params, &period_size, NULL);
+    snd_pcm_hw_params_get_buffer_size_max(alsa_params, &buffer_size);
+    debug(1, "Hardware supports period_size_max: %d, buffer_size_max: %d\n", period_size, buffer_size);
+
+    // we want about 333 ms of buffer, and 50ms period
+    // buffer might still need some tweaking to get reliable operation on RPi + USB DAC...
+    // make sure we do not exceed what HW supports
+    period_size = (period_size < sample_rate / 20 ? period_size : sample_rate / 20);
+    buffer_size = (buffer_size < sample_rate / 3 ? buffer_size : sample_rate / 3);
+
+    // make sure buffer size is at least 4 times period size
+    period_size = (period_size < buffer_size / 4 ? period_size : buffer_size / 4);
+    debug(1, "Trying to set period_size: %d, buffer_size: %d\n", period_size, buffer_size);
+
+    // we keep the originals and try setting period and buffer using copies
+    snd_pcm_uframes_t period_temp, buffer_temp;
+    period_temp = period_size;
+    buffer_temp = buffer_size;
+    snd_pcm_hw_params_t *alsa_params_copy;
+    snd_pcm_hw_params_alloca(&alsa_params_copy);
+    snd_pcm_hw_params_copy(alsa_params_copy, alsa_params);
+
+    // some HW seems to be picky about the order period and buffer are set, so try both ways
+    // first try with buffer_size, period_size
+    if (snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params_copy, &buffer_temp) != 0
+    		|| snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params_copy, &period_temp, NULL) != 0
+    		|| snd_pcm_hw_params(alsa_handle, alsa_params_copy) != 0) {
+        period_temp = period_size;
+        buffer_temp = buffer_size;
+        snd_pcm_hw_params_copy(alsa_params_copy, alsa_params);
+        // retry with period_size, buffer_size
+        if (snd_pcm_hw_params_set_period_size_near(alsa_handle, alsa_params_copy, &period_temp, NULL) != 0
+        		|| snd_pcm_hw_params_set_buffer_size_near(alsa_handle, alsa_params_copy, &buffer_temp) != 0
+        		|| snd_pcm_hw_params(alsa_handle, alsa_params_copy) != 0) {
+        	// set what alsa would have
+        	warn("Setting period and buffer failed, going with the defaults\n");
+            ret = snd_pcm_hw_params(alsa_handle, alsa_params);
+            if (ret < 0)
+                die("unable to set hw parameters: %s\n", snd_strerror(ret));
+            // using alsa defaults, so see what they are
+            snd_pcm_get_params(alsa_handle, &buffer_size, &period_size);
+            debug(1, "Defaults are period_size: %d, buffer_size: %d\n", period_size, buffer_size);
+        }
+    }
 }
 
 static void play(short buf[], int samples) {
@@ -193,3 +240,19 @@ static void volume(double vol) {
     if(snd_mixer_selem_set_playback_volume_all(alsa_mix_elem, alsa_volume) != 0)
         die ("Failed to set playback volume");
 }
+
+static long long get_delay(void) {
+  snd_pcm_sframes_t frames = 0;
+  snd_pcm_delay(alsa_handle, &frames);
+
+  if (frames < 0)
+  {
+#if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
+    snd_pcm_forward(alsa_handle, -frames);
+#endif
+    frames = 0;
+  }
+
+  return (long long)frames * 1000000LL / (long long)device_sample_rate;
+}
+

audio_ao.c

@@ -125,5 +125,6 @@ audio_output audio_ao = {
     .start = &start,
     .stop = &stop,
     .play = &play,
-    .volume = NULL
+    .volume = NULL,
+    .get_delay = NULL
 };

audio_dummy.c

@@ -59,9 +59,9 @@ static void play(short buf[], int samples) {
 
     samples_played += samples;
 
-    long long finishtime = starttime + samples_played * 1e6 / Fs;
-
-    usleep(finishtime - nowtime);
+    long long sleeptime = starttime + samples_played * 1e6 / Fs - nowtime;
+    if (sleeptime > 0)
+        usleep(sleeptime );
 }
 
 static void stop(void) {

audio_pipe.c

@@ -136,5 +136,6 @@ audio_output audio_pipe = {
     .start = &start,
     .stop = &stop,
     .play = &play,
-    .volume = NULL
+    .volume = NULL,
+    .get_delay = NULL
 };

audio_pulse.c

@@ -116,6 +116,18 @@ static void stop(void) {
         fprintf(stderr, __FILE__": pa_simple_drain() failed: %s\n", pa_strerror(pa_error));
 }
 
+static long long get_delay() {
+  pa_usec_t latency;
+  latency = pa_simple_get_latency(pa_dev, &pa_error);
+  if (pa_error < 0 )
+  {
+    latency = (pa_usec_t) 0;
+    fprintf(stderr, __FILE__": get_delay() failed: %s\n", pa_strerror(pa_error));
+  }
+
+  return (long long)latency;
+}
+
 audio_output audio_pulse = {
     .name = "pulse",
     .help = &help,
@@ -124,5 +136,7 @@ audio_output audio_pulse = {
     .start = &start,
     .stop = &stop,
     .play = &play,
-    .volume = NULL
+    .volume = NULL,
+    .get_delay = &get_delay
 };
+

audio_sndio.c

@@ -23,6 +23,14 @@
 static struct sio_hdl *sio;
 static struct sio_par par;
 
+long long writepos; /* frames written */
+
+
+static void position_changed(void *addr, int delta)
+{
+	writepos -= delta;
+}
+
 static int init(int argc, char **argv) {
 	sio = sio_open(SIO_DEVANY, SIO_PLAY, 0);
 	if (!sio)
@@ -41,6 +49,8 @@ static int init(int argc, char **argv) {
 	if (!sio_getpar(sio, &par))
 		die("sndio: failed to get audio parameters");
 
+	sio_onmove(sio, position_changed, NULL);
+
 	return 0;
 }
 
@@ -51,11 +61,14 @@ static void deinit(void) {
 static void start(int sample_rate) {
 	if (sample_rate != par.rate)
 		die("unexpected sample rate!");
+	writepos = 0;
 	sio_start(sio);
 }
 
 static void play(short buf[], int samples) {
-	sio_write(sio, (char *)buf, samples * par.bps * par.pchan);
+	int bytes_per_frame = par.bps * par.pchan;
+	sio_write(sio, (char *)buf, samples * bytes_per_frame);
+	writepos += samples;
 }
 
 static void stop(void) {
@@ -72,6 +85,12 @@ static void volume(double vol) {
 	sio_setvol(sio, v);
 }
 
+static long long get_delay(void) {
+	long long delay = writepos  * 1000000LL / (long long)par.rate;
+
+	return delay;
+}
+
 audio_output audio_sndio = {
 	.name = "sndio",
 	.help = &help,
@@ -80,5 +99,6 @@ audio_output audio_sndio = {
 	.start = &start,
 	.stop = &stop,
 	.play = &play,
-	.volume = &volume
+	.volume = &volume,
+	.get_delay = get_delay
 };

common.c

@@ -27,6 +27,7 @@
 #include <stdio.h>
 #include <stdarg.h>
 #include <stdlib.h>
+#include <math.h>
 #include <memory.h>
 #include <unistd.h>
 #include <openssl/rsa.h>
@@ -202,3 +203,33 @@ void command_stop(void) {
     if (!config.cmd_blocking)
         exit(0);
 }
+
+void biquad_init(biquad_t *bq, double a[], double b[]) {
+    bq->hist[0] = bq->hist[1] = 0.0;
+    memcpy(bq->a, a, 2*sizeof(double));
+    memcpy(bq->b, b, 3*sizeof(double));
+}
+
+void biquad_lpf(biquad_t *bq, double freq, double Q) {
+    double w0 = 2.0 * M_PI * freq;
+    double alpha = sin(w0)/(2.0*Q);
+
+    double a_0 = 1.0 + alpha;
+    double b[3], a[2];
+    b[0] = (1.0-cos(w0))/(2.0*a_0);
+    b[1] = (1.0-cos(w0))/a_0;
+    b[2] = b[0];
+    a[0] = -2.0*cos(w0)/a_0;
+    a[1] = (1-alpha)/a_0;
+
+    biquad_init(bq, a, b);
+}
+
+double biquad_filt(biquad_t *bq, double in) {
+    double w = in - bq->a[0]*bq->hist[0] - bq->a[1]*bq->hist[1];
+    double out = bq->b[1]*bq->hist[0] + bq->b[2]*bq->hist[1] + bq->b[0]*w;
+    bq->hist[1] = bq->hist[0];
+    bq->hist[0] = w;
+
+    return out;
+}

common.h

@@ -26,7 +26,7 @@ typedef struct {
     audio_output *output;
     char *mdns_name;
     mdns_backend *mdns;
-    int buffer_start_fill;
+    int delay;
     int daemonise;
     char *cmd_start, *cmd_stop;
     int cmd_blocking;
@@ -63,4 +63,14 @@ extern shairport_cfg config;
 void shairport_shutdown(int retval);
 void shairport_startup_complete(void);
 
+typedef struct {
+    double hist[2];
+    double a[2];
+    double b[3];
+} biquad_t;
+
+void biquad_init(biquad_t *bq, double a[], double b[]);
+void biquad_lpf(biquad_t *bq, double freq, double Q);
+double biquad_filt(biquad_t *bq, double in);
+
 #endif // _COMMON_H