path: root/tvtime-1.0.2-alsa-r1.patch

Add support for ALSA in combination with USB audio devices, bug #323797
Patch from Ubuntu, rebased to our upstream

diff -Naur tvtime-111b28cca42d.orig/src/audiolib.c tvtime-111b28cca42d/src/audiolib.c
--- tvtime-111b28cca42d.orig/src/audiolib.c	1970-01-01 01:00:00.000000000 +0100
+++ tvtime-111b28cca42d/src/audiolib.c	2011-08-14 11:53:47.726915264 +0200
@@ -0,0 +1,342 @@
+/*
+   Copyright 2008 Empia Technology Inc.
+   Copyright 2008 Markus Rechberger <mrechberger@empiatech.com>
+
+   alsa_set_params is derived from aplay.c
+
+   TODO:
+	more sysfs hacking for determining the empia audio device
+
+*/
+
+#include <stdio.h>
+#include "audiolib.h"
+
+static int alsa_set_params(snd_pcm_t *handle, struct alsa_stream_format *fmt, int start_delay);
+
+char *get_empia_device() {
+        int err;
+        int card = -1;
+        char dev[64];
+        int pcm_device = -1;
+        snd_ctl_t *ctl;
+        snd_pcm_info_t *pcm_info;
+        char *card_name;
+        do {
+                err = snd_card_next(&card);
+                if (card > -1) {
+                        sprintf(dev, "hw:%i", card);
+                        snd_ctl_open(&ctl, dev, 0);
+                        snd_card_get_name(card, &card_name);
+                        snd_pcm_info_alloca(&pcm_info);
+
+                        for (;;)
+                        {
+                                char device[500], descr[1024];
+
+                                if ((err = snd_ctl_pcm_next_device(ctl, &pcm_device)) < 0)
+                                        pcm_device = -1;
+
+                                if (pcm_device < 0)
+                                        break;
+
+                                snd_pcm_info_set_subdevice(pcm_info, 0);
+                                snd_pcm_info_set_device(pcm_info, pcm_device);
+                                snd_pcm_info_set_stream(pcm_info, SND_PCM_STREAM_CAPTURE);
+                                if ((err = snd_ctl_pcm_info(ctl, pcm_info))==0) {
+                                        sprintf(device,"hw:%d,%d", card, pcm_device);
+                                        sprintf(descr,"%s : %s (%s)", card_name, snd_pcm_info_get_name(pcm_info),device);
+                                        if(strcmp(snd_pcm_info_get_name(pcm_info), "USB Audio") == 0) {
+                                                printf("Found \"%s\"\n", descr);
+                                                snd_ctl_close(ctl);
+                                                return strdup(device);
+
+                                        }
+                                }
+                        }
+                        snd_ctl_close(ctl);
+                }
+
+        } while(card > -1);
+        return NULL;
+}
+
+int alsa_get_state(struct alsa_device *dev)
+{
+	return dev->state;
+}
+
+void alsa_set_state(struct alsa_device *dev, enum alsa_state state)
+{
+	dev->state = state;
+}
+
+struct alsa_device *alsa_open(char *input, char *output, unsigned long format, int channels, unsigned int rate)
+{
+	int result;
+	struct alsa_device *dev = malloc(sizeof(struct alsa_device));
+
+	dev->fmt.format = format;
+	dev->fmt.channels = channels;
+	dev->fmt.rate = rate;
+	dev->state = ALSA_STATE_STOPPED;
+
+	if (dev == NULL)
+		return NULL;
+
+
+	if ((result = snd_pcm_open( &dev->playback, output, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))<0)
+	{
+		free(dev);
+		return NULL;
+	}
+
+//	snd_pcm_nonblock( dev->playback, 1);
+
+	alsa_set_params(dev->playback, &dev->fmt, 0);
+
+	if ((result = snd_pcm_open( &dev->capture, input, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK))<0)
+	{
+		free(dev);
+		return NULL;
+	}
+
+//	snd_pcm_nonblock(dev->playback, 1);
+
+	alsa_set_params(dev->capture, &dev->fmt, 1);
+
+	return dev;
+}
+
+void alsa_close(struct alsa_device **dev)
+{
+	struct alsa_device *cdev = *dev;
+	snd_pcm_drain(cdev->playback);
+	snd_pcm_close(cdev->playback);
+	snd_pcm_drain(cdev->capture);
+	snd_pcm_close(cdev->capture);
+	free(*dev);
+	snd_config_update_free_global();
+	*dev = NULL;
+}
+
+#define TRANSFER_BUFSIZE 256
+
+void *alsa_start_thread(void *data)
+{
+	struct alsa_device *dev = data;
+	alsa_loop(dev);
+	return NULL;
+}
+
+int alsa_start_threaded_loop(struct alsa_device *dev)
+{
+	int pret = 0;
+	if (dev->state == ALSA_STATE_STOPPED)
+	{
+		dev->state = ALSA_STATE_RUNNING;
+		pthread_create(&dev->athread, NULL, alsa_start_thread, dev);
+	}
+	return pret;
+}
+
+int alsa_join_threaded_loop(struct alsa_device *dev)
+{
+	if (dev->state == ALSA_STATE_RUNNING)
+	{
+		dev->state = ALSA_STATE_STOPPED;
+		pthread_join(dev->athread, NULL);
+	}
+	return 0;
+}
+
+int alsa_loop(struct alsa_device *dev)
+{
+	int length, ret;
+	unsigned char buf[TRANSFER_BUFSIZE];
+	snd_pcm_uframes_t frames = TRANSFER_BUFSIZE/dev->fmt.bits_per_sample;
+	dev->state = ALSA_STATE_RUNNING;
+	while (dev->state == ALSA_STATE_RUNNING) {
+		length = snd_pcm_readi(dev->capture, buf, frames);
+		switch(length) {
+		case -EAGAIN:
+			snd_pcm_wait(dev->capture, 5);
+			continue;
+		case -EPIPE:
+			snd_pcm_drain(dev->capture);
+			snd_pcm_prepare(dev->capture);
+			continue;
+		default:
+			break;
+		}
+		if (length>0) {
+			ret = snd_pcm_writei(dev->playback, buf, length);
+			switch (ret) {
+			case -EPIPE:
+				snd_pcm_drain(dev->playback);
+				snd_pcm_prepare(dev->playback);
+				continue;
+			default:
+				break;
+			}
+		}
+	}
+	snd_pcm_drain(dev->capture);
+	snd_pcm_drain(dev->playback);
+	return 0;
+}
+
+/* set start_delay to 1 for capture */
+/* taken from aplay.c */
+static int alsa_set_params(snd_pcm_t *handle, struct alsa_stream_format *fmt, int start_delay)
+{
+	snd_pcm_uframes_t chunk_size = 0;
+	snd_pcm_hw_params_t *params;
+	snd_pcm_sw_params_t *swparams;
+	snd_pcm_uframes_t buffer_size;
+
+	unsigned buffer_time = 0;
+	unsigned period_time = 0;
+	snd_pcm_uframes_t buffer_frames = 0;
+	snd_pcm_uframes_t period_frames = 0;
+
+	int err;
+	size_t n;
+	snd_pcm_uframes_t xfer_align;
+	unsigned int rate;
+	int avail_min = -1;
+	int stop_delay = 0;
+
+	snd_pcm_uframes_t start_threshold, stop_threshold;
+	snd_pcm_hw_params_alloca(&params);
+	snd_pcm_sw_params_alloca(&swparams);
+	err = snd_pcm_hw_params_any(handle, params);
+	if (err < 0) {
+		printf("Broken configuration for this PCM: no configurations available");
+		exit(EXIT_FAILURE);
+	}
+
+
+	snd_pcm_access_mask_t *mask = alloca(snd_pcm_access_mask_sizeof());
+	snd_pcm_access_mask_none(mask);
+	snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_RW_INTERLEAVED);
+	err = snd_pcm_hw_params_set_access_mask(handle, params, mask);
+
+	if (err < 0) {
+		printf("Access type not available");
+		exit(EXIT_FAILURE);
+	}
+	err = snd_pcm_hw_params_set_format(handle, params, fmt->format);
+	if (err < 0) {
+		printf("Sample format non available");
+		exit(EXIT_FAILURE);
+	}
+	err = snd_pcm_hw_params_set_channels(handle, params, fmt->channels);
+	if (err < 0) {
+		printf("Channels count non available");
+		exit(EXIT_FAILURE);
+	}
+
+	rate = fmt->rate;
+	err = snd_pcm_hw_params_set_rate_near(handle, params, &rate, 0);
+
+	assert(err >= 0);
+
+	rate = fmt->rate;
+
+	if (buffer_time == 0 && buffer_frames == 0) {
+		err = snd_pcm_hw_params_get_buffer_time_max(params,
+				&buffer_time, 0);
+		assert(err >= 0);
+		if (buffer_time > 500000)
+			buffer_time = 500000;
+	}
+
+	if (period_time == 0 && period_frames == 0) {
+		if (buffer_time > 0)
+			period_time = buffer_time / 4;
+		else
+			period_frames = buffer_frames / 4;
+	}
+
+	if (period_time > 0)
+		err = snd_pcm_hw_params_set_period_time_near(handle, params,
+				&period_time, 0);
+	else
+		err = snd_pcm_hw_params_set_period_size_near(handle, params,
+				&period_frames, 0);
+
+	assert(err >= 0);
+	if (buffer_time > 0)
+		err = snd_pcm_hw_params_set_buffer_time_near(handle, params,
+				&buffer_time, 0);
+	else
+		err = snd_pcm_hw_params_set_buffer_size_near(handle, params,
+				&buffer_frames);
+
+	assert(err >= 0);
+	err = snd_pcm_hw_params(handle, params);
+	if (err < 0) {
+		printf("Unable to install hw params");
+		return -EINVAL;
+	}
+	snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
+	snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
+
+	if (chunk_size == buffer_size) {
+		printf("Can't use period equal to buffer size (%lu == %lu)", chunk_size, buffer_size);
+		return -EINVAL;
+	}
+
+	snd_pcm_sw_params_current(handle, swparams);
+	err = snd_pcm_sw_params_get_xfer_align(swparams, &xfer_align);
+	if (err < 0) {
+		printf("Unable to obtain xfer align\n");
+		exit(EXIT_FAILURE);
+	}
+
+	err = snd_pcm_sw_params_set_sleep_min(handle, swparams, 0);
+
+	assert(err >= 0);
+	if (avail_min < 0)
+		n = chunk_size;
+	else
+		n = (double) rate * avail_min / 1000000;
+
+	err = snd_pcm_sw_params_set_avail_min(handle, swparams, n);
+
+	/* round up to closest transfer boundary */
+	n = (buffer_size / xfer_align) * xfer_align;
+	if (start_delay <= 0) {
+		start_threshold = n + (double) rate * start_delay / 1000000;
+	} else
+		start_threshold = (double) rate * start_delay / 1000000;
+	if (start_threshold < 1)
+		start_threshold = 1;
+	if (start_threshold > n)
+		start_threshold = n;
+
+	err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
+	assert(err >= 0);
+
+	if (stop_delay <= 0)
+		stop_threshold = buffer_size + (double) rate * stop_delay / 1000000;
+	else
+		stop_threshold = (double) rate * stop_delay / 1000000;
+
+	err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
+	assert(err >= 0);
+
+	err = snd_pcm_sw_params_set_xfer_align(handle, swparams, xfer_align);
+	assert(err >= 0);
+
+	if (snd_pcm_sw_params(handle, swparams) < 0) {
+		printf("unable to install sw params");
+		return -EINVAL;
+	}
+
+	fmt->bits_per_sample = snd_pcm_format_physical_width(fmt->format);
+	fmt->bits_per_frame = fmt->bits_per_sample * fmt->channels;
+	fmt->chunk_bytes = chunk_size * fmt->bits_per_frame / 8;
+	return 0;
+}
diff -Naur tvtime-111b28cca42d.orig/src/audiolib.h tvtime-111b28cca42d/src/audiolib.h
--- tvtime-111b28cca42d.orig/src/audiolib.h	1970-01-01 01:00:00.000000000 +0100
+++ tvtime-111b28cca42d/src/audiolib.h	2011-08-14 11:53:47.771915261 +0200
@@ -0,0 +1,38 @@
+#ifndef _AUDIO_LIB_H
+#define _AUDIO_LIB_H
+#include <alsa/asoundlib.h>
+#include <pthread.h>
+
+struct alsa_stream_format {
+	unsigned long format;
+	int channels;
+	unsigned int rate;
+	size_t bits_per_sample;
+	size_t bits_per_frame;
+	size_t chunk_bytes;
+};
+
+enum alsa_state {
+	ALSA_STATE_STOPPED,
+	ALSA_STATE_RUNNING
+};
+
+struct alsa_device {
+	snd_pcm_t *capture;
+	snd_pcm_t *playback;
+	enum alsa_state state;
+	struct alsa_stream_format fmt;
+	pthread_t athread;
+};
+
+struct alsa_device *alsa_open(char *input, char *output, unsigned long format, int channels, unsigned int rate);
+char *get_empia_device();
+int alsa_loop(struct alsa_device *dev);
+int alsa_get_state(struct alsa_device *dev);
+void alsa_close(struct alsa_device **dev);
+int alsa_get_state(struct alsa_device *dev);
+void alsa_set_state(struct alsa_device *dev, enum alsa_state state);
+int alsa_start_threaded_loop(struct alsa_device *dev);
+int alsa_join_threaded_loop(struct alsa_device *dev);
+
+#endif // _AUDIO_LIB_H
diff -Naur tvtime-111b28cca42d.orig/src/Makefile.am tvtime-111b28cca42d/src/Makefile.am
--- tvtime-111b28cca42d.orig/src/Makefile.am	2011-08-14 11:52:43.404917552 +0200
+++ tvtime-111b28cca42d/src/Makefile.am	2011-08-14 11:56:09.243910228 +0200
@@ -37,7 +37,7 @@
 	utils.h utils.c pulldown.h pulldown.c hashtable.h hashtable.c \
 	cpuinfo.h cpuinfo.c videodev2.h menu.c menu.h \
 	outputfilter.h outputfilter.c xmltv.h xmltv.c gettext.h tvtimeglyphs.h \
-	copyfunctions.h copyfunctions.c alsa_stream.c
+	copyfunctions.h copyfunctions.c alsa_stream.c audiolib.h audiolib.c
 
 if ARCH_X86
 DSCALER_SRCS = $(top_srcdir)/plugins/dscalerapi.h \
@@ -74,7 +74,7 @@
 	$(PLUGIN_CFLAGS) $(X11_CFLAGS) $(XML2_FLAG) \
 	$(FONT_CFLAGS) $(AM_CFLAGS)
 tvtime_LDADD  = $(TTF_LIBS) $(ZLIB_LIBS) $(PNG_LIBS) \
-	$(X11_LIBS) $(XML2_LIBS) $(ASOUND_LIBS) -lm -lsupc++ -lpthread
+	$(X11_LIBS) $(XML2_LIBS) $(ASOUND_LIBS) -lm -lsupc++ -lpthread -lasound
 
 tvtime_command_SOURCES = utils.h utils.c tvtimeconf.h tvtimeconf.c \
 	tvtime-command.c
diff -Naur tvtime-111b28cca42d.orig/src/tvtime.c tvtime-111b28cca42d/src/tvtime.c
--- tvtime-111b28cca42d.orig/src/tvtime.c	2011-02-01 02:35:26.000000000 +0100
+++ tvtime-111b28cca42d/src/tvtime.c	2011-08-14 11:56:35.443909297 +0200
@@ -78,6 +78,7 @@
 #include "menu.h"
 #include "tvtimeglyphs.h"
 #include "alsa_stream.h"
+#include "audiolib.h"
 
 /**
  * This is how many frames to wait until deciding if the pulldown phase
@@ -1210,6 +1211,7 @@
     int last_current_id = -1;
     int quiet_screenshots = 0;
     char prevloc[ 256 ];
+    char *empia_device;
     int i;
 
     ct = config_new();
@@ -1581,6 +1583,8 @@
     build_fspos_menu( commands_get_menu( commands, "fspos" ),
                       config_get_fullscreen_position( ct ) );
 
+    empia_device = get_empia_device();
+
     /* Initialize our timestamps. */
     for(;;) {
         const char *fifo_args = 0;
@@ -2056,9 +2060,30 @@
 
         if( tuner_state == TUNER_STATE_HAS_SIGNAL ) {
             has_signal = 1;
+            /* reopen the device for now, there are several issues which don't allow pausing the
+               audio data transfer.
+
+               TODO:
+               The driver(?) seems to crash when setting up the alsa parameters again during the same
+               session.
+            */
+
+            if (empia_device && videoinput_get_audio(vidin) == NULL) {
+                    videoinput_set_audio(vidin, alsa_open(empia_device, "default", SND_PCM_FORMAT_S16_LE, 2 /* 2 channels */, 48000 /* rate */));
+                    if (videoinput_get_audio(vidin)) {
+                            alsa_start_threaded_loop(videoinput_get_audio(vidin));
+                    }
+            }
+
             if( osd ) tvtime_osd_signal_present( osd, 1 );
         } else if( tuner_state == TUNER_STATE_NO_SIGNAL ) {
             if( osd ) tvtime_osd_signal_present( osd, 0 );
+            if (videoinput_get_audio(vidin)) {
+                    alsa_join_threaded_loop(videoinput_get_audio(vidin));
+                    alsa_close(videoinput_get_audio_p(vidin));
+                    videoinput_set_audio(vidin, NULL);
+            }
+
             if( fadepos < 256 ) {
                 crossfade_frame( fadeframe, saveframe, blueframe, width,
                                  height, width*2, width*2, width*2, fadepos );
@@ -2499,6 +2524,11 @@
     /* Return to normal scheduling. */
     set_default_priority();
 
+    if (videoinput_get_audio(vidin)) {
+            alsa_join_threaded_loop(videoinput_get_audio(vidin));
+            alsa_close(videoinput_get_audio_p(vidin));
+    }
+
     /* Remember to save our settings if we were scanning. */
     if( scanning ) {
         station_writeconfig( stationmgr );
diff -Naur tvtime-111b28cca42d.orig/src/videoinput.c tvtime-111b28cca42d/src/videoinput.c
--- tvtime-111b28cca42d.orig/src/videoinput.c	2011-02-01 02:35:26.000000000 +0100
+++ tvtime-111b28cca42d/src/videoinput.c	2011-08-14 12:03:39.783894207 +0200
@@ -38,6 +38,7 @@
 #include <linux/videodev2.h>
 #include "videoinput.h"
 #include "mixer.h"
+#include "audiolib.h"
 
 /**
  * How long to wait when we lose a signal, or acquire a signal.
@@ -198,8 +199,25 @@
     /* V4L2 capture state. */
     capture_buffer_t capbuffers[ MAX_CAPTURE_BUFFERS ];
     int is_streaming;
+
+    struct alsa_device *adev;
 };
 
+struct alsa_device *videoinput_get_audio( videoinput_t *vidin )
+{
+    return vidin->adev;
+}
+
+struct alsa_device **videoinput_get_audio_p( videoinput_t *vidin )
+{
+    return &vidin->adev;
+}
+
+void videoinput_set_audio( videoinput_t *vidin, struct alsa_device *dev )
+{
+    vidin->adev = dev;
+}
+
 static void videoinput_start_capture_v4l2( videoinput_t *vidin )
 {
     if( !vidin->is_streaming ) {
@@ -347,6 +365,7 @@
     vidin->norm = norm;
     vidin->volume = volume;
     vidin->amode = 0;
+    vidin->adev = NULL;
     vidin->height = videoinput_get_norm_height( norm );
     vidin->cur_tuner_state = TUNER_STATE_NO_SIGNAL;
 
diff -Naur tvtime-111b28cca42d.orig/src/videoinput.h tvtime-111b28cca42d/src/videoinput.h
--- tvtime-111b28cca42d.orig/src/videoinput.h	2011-02-01 02:35:26.000000000 +0100
+++ tvtime-111b28cca42d/src/videoinput.h	2011-08-14 11:53:47.775915262 +0200
@@ -278,6 +278,20 @@
 const char *videoinput_get_driver_name( videoinput_t *vidin );
 
 /**
+ * Returns the audio handle of the device struct.
+ */
+struct alsa_device *videoinput_get_audio( videoinput_t *vidin );
+
+/**
+ * Returns a pointer to the alsa_device pointer, used for cleaning up
+ */
+struct alsa_device **videoinput_get_audio_p (videoinput_t *vidin );
+
+/**
+ * Set adev for vidin
+ */
+void videoinput_set_audio( videoinput_t *vidin, struct alsa_device *dev);
+/**
  * Sets the capture card volume to use as a percentage from 0-100.
  * If the value is negative, the capture card volume will remain unset.
  */