- Remove OSS support from source, it is deprecated since kernel 2.6 (released December 2003)

- remove libavresample from source, deprecated since 2017
- make ALSA and libswresample mandatory for now

Makefile

@@ -54,7 +54,7 @@ endif
 
 ifeq ($(CUVID),1)
 ifeq ($(DRM),1)
-$(error Missmatch in Plugin selection)
+$(error Mismatch in Plugin selection)
 exit 1;
 endif
 endif
@@ -62,7 +62,7 @@ endif
 
 ifeq ($(CUVID),1)
 ifeq ($(VAAPI),1)
-$(error Missmatch in Plugin selection)
+$(error Mismatch in Plugin selection)
 exit 1;
 endif
 endif
@@ -78,26 +78,11 @@ PLUGIN = softhdcuvid
 # support OPENGLOSD always needed
 OPENGLOSD=1
 
-# support alsa audio output module
-ALSA ?= $(shell pkg-config --exists alsa && echo 1)
-    # support OSS audio output module
-OSS ?= 1
-
 # use DMPS
 SCREENSAVER=1
 
 OPENGL=1
 
-# use ffmpeg libswresample
-SWRESAMPLE ?= $(shell pkg-config --exists libswresample && echo 1)
-SWRESAMPLE = 1
-
-# use libav libavresample
-#ifneq ($(SWRESAMPLE),1)
-#AVRESAMPLE ?= $(shell pkg-config --exists libavresample && echo 1#)
-#AVRESAMPLE = 1
-#endif
-
 CONFIG += -DHAVE_GL			# needed for mpv libs
 #CONFIG += -DSTILL_DEBUG=2		# still picture debug verbose level
 CONFIG += -DAV_INFO -DAV_INFO_TIME=3000	# info/debug a/v sync
@@ -149,16 +134,6 @@ APIVERSION = $(call PKGCFG,apiversion)
 
 ### Parse softhddevice config
 
-ifeq ($(ALSA),1)
-CONFIG += -DUSE_ALSA
-_CFLAGS += $(shell pkg-config --cflags alsa)
-LIBS += $(shell pkg-config --libs alsa)
-endif
-
-ifeq ($(OSS),1)
-CONFIG += -DUSE_OSS
-endif
-
 ifeq ($(OPENGL),1)
 #_CFLAGS += $(shell pkg-config --cflags libva-glx)
 #LIBS += $(shell pkg-config --libs libva-glx)
@@ -226,11 +201,13 @@ SOFILE = libvdr-$(PLUGIN).so
 #
 # Test that libswresample is available
 #
-#ifneq (exists, $(shell pkg-config libswresample && echo exists))
+ifneq (exists, $(shell pkg-config libswresample && echo exists))
-#  $(warning ******************************************************************)
+  $(warning ******************************************************************)
-#  $(warning 'libswresample' not found!)
+  $(warning 'libswresample' not found!)
-#  $(error ******************************************************************)
+  $(error ******************************************************************)
-#endif
+endif
+_CFLAGS += $(shell pkg-config --cflags libswresample)
+LIBS += $(shell pkg-config --libs libswresample)
 
 #
 # Test and set config for libavutil
@@ -265,22 +242,22 @@ endif
 _CFLAGS += $(shell pkg-config --cflags libavcodec)
 LIBS += $(shell pkg-config --libs libavcodec libavfilter)
 
+#
+# Test and set config for alsa
+#
+ifneq (exists, $(shell pkg-config alsa && echo exists))
+  $(warning ******************************************************************)
+  $(warning 'alsa' not found!)
+  $(error ******************************************************************)
+endif
+_CFLAGS += $(shell pkg-config --cflags alsa)
+LIBS += $(shell pkg-config --libs alsa)
 
 ifeq ($(SCREENSAVER),1)
 CONFIG += -DUSE_SCREENSAVER
 _CFLAGS += $(shell pkg-config --cflags xcb-screensaver xcb-dpms)
 LIBS += $(shell pkg-config --libs xcb-screensaver xcb-dpms)
 endif
-ifeq ($(SWRESAMPLE),1)
-CONFIG += -DUSE_SWRESAMPLE
-_CFLAGS += $(shell pkg-config --cflags libswresample)
-LIBS += $(shell pkg-config --libs libswresample)
-endif
-ifeq ($(AVRESAMPLE),1)
-CONFIG += -DUSE_AVRESAMPLE
-_CFLAGS += $(shell pkg-config --cflags libavresample)
-LIBS += $(shell pkg-config --libs libavresample)
-endif
 
 #_CFLAGS += $(shell pkg-config --cflags libavcodec x11 x11-xcb xcb xcb-icccm)
 #LIBS += -lrt $(shell pkg-config --libs libavcodec x11 x11-xcb xcb xcb-icccm)

README.md

@@ -25,14 +25,13 @@ A software and GPU emulated UHD output device plugin for VDR.
 
     o Video decoder CUVID or VAAPI
     o Video output opengl or DRM
-    o Audio FFMpeg / Alsa / Analog
+    o Audio FFMpeg / ALSA / Analog
-    o Audio FFMpeg / Alsa / Digital
+    o Audio FFMpeg / ALSA / Digital
-    o Audio FFMpeg / OSS / Analog
     o HDMI/SPDIF pass-through
     o Software volume, compression, normalize and channel resample
     o VDR ScaleVideo API
     o CUDA deinterlacer
-    o Suspend / Dettach
+    o Suspend / Detach
     o Support for ambilight
     o Support for Screencopy
     o PIP (Picture-in-Picture) (only for CUVID)
@@ -96,9 +95,6 @@ Install:
 	make
 	make install
 
-	You can edit Makefile to enable/disable  Alsa / OSS
-	support.  The default is to autodetect as much as possible.
-
 	You have to start vdr with -P 'softhdcuvid -d :0.0  ..<more option>.. '
 
 Beginners Guide for libplacebo:
@@ -148,27 +144,17 @@ Setup:	environment
 	Following is supported:
 
 	DISPLAY=:0.0
-		x11 display name
+		X11 display name
 
-    only if alsa is configured
+    ALSA configuration:
 	ALSA_DEVICE=default
-		alsa PCM device name
+		ALSA PCM device name
 	ALSA_PASSTHROUGH_DEVICE=
-		alsa pass-though (AC-3,E-AC-3,DTS,...) device name
+		ALSA pass-though (AC-3,E-AC-3,DTS,...) device name
 	ALSA_MIXER=default
-		alsa control device name
+		ALSA control device name
 	ALSA_MIXER_CHANNEL=PCM
-		alsa control channel name
+		ALSA control channel name
-
-    only if oss is configured
-	OSS_AUDIODEV=/dev/dsp
-		oss dsp device name
-	OSS_PASSTHROUGHDEV=
-		oss pass-though (AC-3,E-AC-3,DTS,...) device name
-	OSS_MIXERDEV=/dev/mixer
-		oss mixer device name
-	OSS_MIXER_CHANNEL=pcm
-		oss mixer channel name
 
 Setup: /etc/vdr/setup.conf
 ------
@@ -197,7 +183,7 @@ Setup: /etc/vdr/setup.conf
 	(only 0, 1, 4 supported with VA-API)
 
 	softhddevice.<res>.SkipChromaDeinterlace = 0
-	0 = disabled, 1 = enabled (for slower cards, poor qualität)
+	0 = disabled, 1 = enabled (for slower cards, poor qualit<EFBFBD>t)
 
 	softhddevice.<res>.InverseTelecine = 0
 	0 = disabled, 1 = enabled
@@ -336,10 +322,7 @@ Commandline:
 
 	Selects audio output module and device.
 	""		to disable audio output
-	/...		to use oss audio module (if compiled with oss
+	other		to use ALSA audio module
-			support)
-	other		to use alsa audio module (if compiled with alsa
-			support)
 
 SVDRP:
 ------

audio.c

@@ -30,17 +30,12 @@
 ///
 /// @note alsa async playback is broken, don't use it!
 ///
-/// OSS PCM/Mixer api is supported.
-/// @see http://manuals.opensound.com/developer/
-///
 ///
 /// @todo FIXME: there can be problems with little/big endian.
 ///
 #ifdef DEBUG
 #undef DEBUG
 #endif
-//#define USE_ALSA	///< enable alsa support
-//#define USE_OSS	///< enable OSS support
 #define USE_AUDIO_THREAD ///< use thread for audio playback
 #define USE_AUDIO_MIXER  ///< use audio module mixer
 
@@ -57,41 +52,12 @@
 #define _(str) gettext(str) ///< gettext shortcut
 #define _N(str) str         ///< gettext_noop shortcut
 
-#ifdef USE_ALSA
 #include <alsa/asoundlib.h>
-#endif
-#ifdef USE_OSS
-#include <sys/ioctl.h>
-#include <sys/soundcard.h>
-#include <sys/stat.h>
-#include <sys/types.h>
-// SNDCTL_DSP_HALT_OUTPUT compatibility
-#ifndef SNDCTL_DSP_HALT_OUTPUT
-#if defined(SNDCTL_DSP_RESET_OUTPUT)
-#define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET_OUTPUT
-#elif defined(SNDCTL_DSP_RESET)
-#define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET
-#else
-#error "No valid SNDCTL_DSP_HALT_OUTPUT found."
-#endif
-#endif
-#include <errno.h>
-#include <fcntl.h>
-#include <poll.h>
-#include <unistd.h>
-#endif
 
 #ifdef USE_AUDIO_THREAD
-#ifndef __USE_GNU
-#define __USE_GNU
-#endif
 #include <pthread.h>
 #include <sys/resource.h>
 #include <sys/syscall.h>
-#ifndef HAVE_PTHREAD_NAME
-/// only available with newer glibc
-#define pthread_setname_np(thread, name)
-#endif
 #endif
 
 #include "iatomic.h" // portable atomic_t
@@ -723,8 +689,6 @@ static void AudioRingExit(void) {
     AudioRingWrite = 0;
 }
 
-#ifdef USE_ALSA
-
 //============================================================================
 //  A L S A
 //============================================================================
@@ -1358,480 +1322,6 @@ static const AudioModule AlsaModule = {
     .Exit = AlsaExit,
 };
 
-#endif // USE_ALSA
-
-#ifdef USE_OSS
-
-//============================================================================
-//  O S S
-//============================================================================
-
-//----------------------------------------------------------------------------
-//  OSS variables
-//----------------------------------------------------------------------------
-
-static int OssPcmFildes = -1;   ///< pcm file descriptor
-static int OssMixerFildes = -1; ///< mixer file descriptor
-static int OssMixerChannel;     ///< mixer channel index
-static int OssFragmentTime;     ///< fragment time in ms
-
-//----------------------------------------------------------------------------
-//  OSS pcm
-//----------------------------------------------------------------------------
-
-/**
-**	Play samples from ringbuffer.
-**
-**	@retval	0	ok
-**	@retval 1	ring buffer empty
-**	@retval -1	underrun error
-*/
-static int OssPlayRingbuffer(void) {
-    int first;
-
-    first = 1;
-    for (;;) {
-        audio_buf_info bi;
-        const void *p;
-        int n;
-
-        if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
-            Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"), strerror(errno));
-            return -1;
-        }
-        Debug(4, "audio/oss: %d bytes free\n", bi.bytes);
-
-        n = RingBufferGetReadPointer(AudioRing[AudioRingRead].RingBuffer, &p);
-        if (!n) {        // ring buffer empty
-            if (first) { // only error on first loop
-                return 1;
-            }
-            return 0;
-        }
-        if (n < bi.bytes) { // not enough bytes in ring buffer
-            bi.bytes = n;
-        }
-        if (bi.bytes <= 0) { // full or buffer empty
-            break;           // bi.bytes could become negative!
-        }
-
-        if (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough) {
-            // FIXME: quick&dirty cast
-            AudioSoftAmplifier((int16_t *)p, bi.bytes);
-            // FIXME: if not all are written, we double amplify them
-        }
-        for (;;) {
-            n = write(OssPcmFildes, p, bi.bytes);
-            if (n != bi.bytes) {
-                if (n < 0) {
-                    if (n == EAGAIN) {
-                        continue;
-                    }
-                    Error(_("audio/oss: write error: %s\n"), strerror(errno));
-                    return 1;
-                }
-                Warning(_("audio/oss: error not all bytes written\n"));
-            }
-            break;
-        }
-        // advance how many could written
-        RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, n);
-        first = 0;
-    }
-
-    return 0;
-}
-
-/**
-**	Flush OSS buffers.
-*/
-static void OssFlushBuffers(void) {
-    if (OssPcmFildes != -1) {
-        // flush kernel buffers
-        if (ioctl(OssPcmFildes, SNDCTL_DSP_HALT_OUTPUT, NULL) < 0) {
-            Error(_("audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"), strerror(errno));
-        }
-    }
-}
-
-#ifdef USE_AUDIO_THREAD
-
-//----------------------------------------------------------------------------
-//  thread playback
-//----------------------------------------------------------------------------
-
-/**
-**	OSS thread
-**
-**	@retval -1	error
-**	@retval 0	underrun
-**	@retval 1	running
-*/
-static int OssThread(void) {
-    int err;
-
-    if (!OssPcmFildes) {
-        usleep(OssFragmentTime * 1000);
-        return -1;
-    }
-    for (;;) {
-        struct pollfd fds[1];
-
-        if (AudioPaused) {
-            return 1;
-        }
-        // wait for space in kernel buffers
-        fds[0].fd = OssPcmFildes;
-        fds[0].events = POLLOUT | POLLERR;
-        // wait for space in kernel buffers
-        err = poll(fds, 1, OssFragmentTime);
-        if (err < 0) {
-            if (err == EAGAIN) {
-                continue;
-            }
-            Error(_("audio/oss: error poll %s\n"), strerror(errno));
-            usleep(OssFragmentTime * 1000);
-            return -1;
-        }
-        break;
-    }
-    if (!err || AudioPaused) { // timeout or some commands
-        return 1;
-    }
-
-    if ((err = OssPlayRingbuffer())) { // empty / error
-        if (err < 0) {                 // underrun error
-            return -1;
-        }
-        sched_yield();
-        usleep(OssFragmentTime * 1000); // let fill/empty the buffers
-        return 0;
-    }
-
-    return 1;
-}
-
-#endif
-
-//----------------------------------------------------------------------------
-
-/**
-**	Open OSS pcm device.
-**
-**	@param passthrough	use pass-through (AC-3, ...) device
-*/
-static int OssOpenPCM(int passthrough) {
-    const char *device;
-    int fildes;
-
-    // &&|| hell
-    if (!(passthrough && ((device = AudioPassthroughDevice) || (device = getenv("OSS_PASSTHROUGHDEV")))) &&
-        !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
-        device = "/dev/dsp";
-    }
-    if (!AudioDoingInit) {
-        Info(_("audio/oss: using %sdevice '%s'\n"), passthrough ? "pass-through " : "", device);
-    }
-
-    if ((fildes = open(device, O_WRONLY)) < 0) {
-        Error(_("audio/oss: can't open dsp device '%s': %s\n"), device, strerror(errno));
-        return -1;
-    }
-    return fildes;
-}
-
-/**
-**	Initialize OSS pcm device.
-**
-**	@see AudioPCMDevice
-*/
-static void OssInitPCM(void) {
-    int fildes;
-
-    fildes = OssOpenPCM(0);
-
-    OssPcmFildes = fildes;
-}
-
-//----------------------------------------------------------------------------
-//  OSS Mixer
-//----------------------------------------------------------------------------
-
-/**
-**	Set OSS mixer volume (0-1000)
-**
-**	@param volume	volume (0 .. 1000)
-*/
-static void OssSetVolume(int volume) {
-    int v;
-
-    if (OssMixerFildes != -1) {
-        v = (volume * 255) / 1000;
-        v &= 0xff;
-        v = (v << 8) | v;
-        if (ioctl(OssMixerFildes, MIXER_WRITE(OssMixerChannel), &v) < 0) {
-            Error(_("audio/oss: ioctl(MIXER_WRITE): %s\n"), strerror(errno));
-        }
-    }
-}
-
-/**
-**	Mixer channel name table.
-*/
-static const char *OssMixerChannelNames[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
-
-/**
-**	Initialize OSS mixer.
-*/
-static void OssInitMixer(void) {
-    const char *device;
-    const char *channel;
-    int fildes;
-    int devmask;
-    int i;
-
-    if (!(device = AudioMixerDevice)) {
-        if (!(device = getenv("OSS_MIXERDEV"))) {
-            device = "/dev/mixer";
-        }
-    }
-    if (!(channel = AudioMixerChannel)) {
-        if (!(channel = getenv("OSS_MIXER_CHANNEL"))) {
-            channel = "pcm";
-        }
-    }
-    Debug(3, "audio/oss: mixer %s - %s open\n", device, channel);
-
-    if ((fildes = open(device, O_RDWR)) < 0) {
-        Error(_("audio/oss: can't open mixer device '%s': %s\n"), device, strerror(errno));
-        return;
-    }
-    // search channel name
-    if (ioctl(fildes, SOUND_MIXER_READ_DEVMASK, &devmask) < 0) {
-        Error(_("audio/oss: ioctl(SOUND_MIXER_READ_DEVMASK): %s\n"), strerror(errno));
-        close(fildes);
-        return;
-    }
-    for (i = 0; i < SOUND_MIXER_NRDEVICES; ++i) {
-        if (!strcasecmp(OssMixerChannelNames[i], channel)) {
-            if (devmask & (1 << i)) {
-                OssMixerFildes = fildes;
-                OssMixerChannel = i;
-                return;
-            }
-            Error(_("audio/oss: channel '%s' not supported\n"), channel);
-            break;
-        }
-    }
-    Error(_("audio/oss: channel '%s' not found\n"), channel);
-    close(fildes);
-}
-
-//----------------------------------------------------------------------------
-//  OSS API
-//----------------------------------------------------------------------------
-
-/**
-**	Get OSS audio delay in time stamps.
-**
-**	@returns audio delay in time stamps.
-*/
-static int64_t OssGetDelay(void) {
-    int delay;
-    int64_t pts;
-
-    // setup failure
-    if (OssPcmFildes == -1 || !AudioRing[AudioRingRead].HwSampleRate) {
-        return 0L;
-    }
-    if (!AudioRunning) { // audio not running
-        Error(_("audio/oss: should not happen\n"));
-        return 0L;
-    }
-    // delay in bytes in kernel buffers
-    delay = -1;
-    if (ioctl(OssPcmFildes, SNDCTL_DSP_GETODELAY, &delay) == -1) {
-        Error(_("audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"), strerror(errno));
-        return 0L;
-    }
-    if (delay < 0) {
-        delay = 0;
-    }
-
-    pts = ((int64_t)delay * 90 * 1000) /
-          (AudioRing[AudioRingRead].HwSampleRate * AudioRing[AudioRingRead].HwChannels * AudioBytesProSample);
-
-    return pts;
-}
-
-/**
-**	Setup OSS audio for requested format.
-**
-**	@param sample_rate	sample rate/frequency
-**	@param channels		number of channels
-**	@param passthrough	use pass-through (AC-3, ...) device
-**
-**	@retval 0	everything ok
-**	@retval 1	didn't support frequency/channels combination
-**	@retval -1	something gone wrong
-*/
-static int OssSetup(int *sample_rate, int *channels, int passthrough) {
-    int ret;
-    int tmp;
-    int delay;
-    audio_buf_info bi;
-
-    if (OssPcmFildes == -1) { // OSS not ready
-        // FIXME: if open fails for fe. pass-through, we never recover
-        return -1;
-    }
-
-    if (1) { // close+open for pcm / AC-3
-        int fildes;
-
-        fildes = OssPcmFildes;
-        OssPcmFildes = -1;
-        close(fildes);
-        if (!(fildes = OssOpenPCM(passthrough))) {
-            return -1;
-        }
-        OssPcmFildes = fildes;
-    }
-
-    ret = 0;
-
-    tmp = AFMT_S16_NE; // native 16 bits
-    if (ioctl(OssPcmFildes, SNDCTL_DSP_SETFMT, &tmp) == -1) {
-        Error(_("audio/oss: ioctl(SNDCTL_DSP_SETFMT): %s\n"), strerror(errno));
-        // FIXME: stop player, set setup failed flag
-        return -1;
-    }
-    if (tmp != AFMT_S16_NE) {
-        Error(_("audio/oss: device doesn't support 16 bit sample format.\n"));
-        // FIXME: stop player, set setup failed flag
-        return -1;
-    }
-
-    tmp = *channels;
-    if (ioctl(OssPcmFildes, SNDCTL_DSP_CHANNELS, &tmp) == -1) {
-        Error(_("audio/oss: ioctl(SNDCTL_DSP_CHANNELS): %s\n"), strerror(errno));
-        return -1;
-    }
-    if (tmp != *channels) {
-        Warning(_("audio/oss: device doesn't support %d channels.\n"), *channels);
-        *channels = tmp;
-        ret = 1;
-    }
-
-    tmp = *sample_rate;
-    if (ioctl(OssPcmFildes, SNDCTL_DSP_SPEED, &tmp) == -1) {
-        Error(_("audio/oss: ioctl(SNDCTL_DSP_SPEED): %s\n"), strerror(errno));
-        return -1;
-    }
-    if (tmp != *sample_rate) {
-        Warning(_("audio/oss: device doesn't support %dHz sample rate.\n"), *sample_rate);
-        *sample_rate = tmp;
-        ret = 1;
-    }
-#ifdef SNDCTL_DSP_POLICY
-    tmp = 3;
-    if (ioctl(OssPcmFildes, SNDCTL_DSP_POLICY, &tmp) == -1) {
-        Error(_("audio/oss: ioctl(SNDCTL_DSP_POLICY): %s\n"), strerror(errno));
-    } else {
-        Info("audio/oss: set policy to %d\n", tmp);
-    }
-#endif
-
-    if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
-        Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"), strerror(errno));
-        bi.fragsize = 4096;
-        bi.fragstotal = 16;
-    } else {
-        Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes);
-    }
-
-    OssFragmentTime = (bi.fragsize * 1000) / (*sample_rate * *channels * AudioBytesProSample);
-
-    Debug(3, "audio/oss: buffer size %d %dms, fragment size %d %dms\n", bi.fragsize * bi.fragstotal,
-          (bi.fragsize * bi.fragstotal * 1000) / (*sample_rate * *channels * AudioBytesProSample), bi.fragsize,
-          OssFragmentTime);
-
-    // start when enough bytes for initial write
-    AudioStartThreshold = (bi.fragsize - 1) * bi.fragstotal;
-
-    // buffer time/delay in ms
-    delay = AudioBufferTime + 300;
-    if (VideoAudioDelay > 0) {
-        delay += VideoAudioDelay / 90;
-    }
-    if (AudioStartThreshold < (*sample_rate * *channels * AudioBytesProSample * delay) / 1000U) {
-        AudioStartThreshold = (*sample_rate * *channels * AudioBytesProSample * delay) / 1000U;
-    }
-    // no bigger, than 1/3 the buffer
-    if (AudioStartThreshold > AudioRingBufferSize / 3) {
-        AudioStartThreshold = AudioRingBufferSize / 3;
-    }
-
-    if (!AudioDoingInit) {
-        Info(_("audio/oss: delay %ums\n"),
-             (AudioStartThreshold * 1000) / (*sample_rate * *channels * AudioBytesProSample));
-    }
-
-    return ret;
-}
-
-/**
-**	Play audio.
-*/
-static void OssPlay(void) {}
-
-/**
-**	Pause audio.
-*/
-void OssPause(void) {}
-
-/**
-**	Initialize OSS audio output module.
-*/
-static void OssInit(void) {
-    OssInitPCM();
-    OssInitMixer();
-}
-
-/**
-**	Cleanup OSS audio output module.
-*/
-static void OssExit(void) {
-    if (OssPcmFildes != -1) {
-        close(OssPcmFildes);
-        OssPcmFildes = -1;
-    }
-    if (OssMixerFildes != -1) {
-        close(OssMixerFildes);
-        OssMixerFildes = -1;
-    }
-}
-
-/**
-**	OSS module.
-*/
-static const AudioModule OssModule = {
-    .Name = "oss",
-#ifdef USE_AUDIO_THREAD
-    .Thread = OssThread,
-#endif
-    .FlushBuffers = OssFlushBuffers,
-    .GetDelay = OssGetDelay,
-    .SetVolume = OssSetVolume,
-    .Setup = OssSetup,
-    .Play = OssPlay,
-    .Pause = OssPause,
-    .Init = OssInit,
-    .Exit = OssExit,
-};
-
-#endif // USE_OSS
-
 //============================================================================
 //  Noop
 //============================================================================
@@ -2089,12 +1579,7 @@ static void AudioExitThread(void) {
 **	Table of all audio modules.
 */
 static const AudioModule *AudioModules[] = {
-#ifdef USE_ALSA
     &AlsaModule,
-#endif
-#ifdef USE_OSS
-    &OssModule,
-#endif
     &NoopModule,
 };
 
@@ -2666,12 +2151,7 @@ void AudioInit(void) {
     int chan;
 
     name = "noop";
-#ifdef USE_OSS
-    name = "oss";
-#endif
-#ifdef USE_ALSA
     name = "alsa";
-#endif
     if (AudioModuleName) {
         name = AudioModuleName;
     }

codec.c

@@ -36,10 +36,6 @@
 #define USE_AUDIO_DRIFT_CORRECTION
 /// compile AC-3 audio drift correction support (very experimental)
 #define USE_AC3_DRIFT_CORRECTION
-/// use ffmpeg libswresample API (autodected, Makefile)
-#define noUSE_SWRESAMPLE
-/// use libav libavresample API (autodected, Makefile)
-#define noUSE_AVRESAMPLE
 
 #include <stdio.h>
 #include <stdlib.h>
@@ -62,17 +58,8 @@
 #include <libavutil/mem.h>
 #include <libavutil/opt.h>
 
-#ifdef USE_SWRESAMPLE
 #include <libswresample/swresample.h>
-#endif
-#ifdef USE_AVRESAMPLE
-#include <libavresample/avresample.h>
-#include <libavutil/opt.h>
-#endif
 
-#ifndef __USE_GNU
-#define __USE_GNU
-#endif
 #include <pthread.h>
 
 // clang-format off
@@ -672,18 +659,8 @@ struct _audio_decoder_ {
     int HwSampleRate; ///< hw sample rate
     int HwChannels;   ///< hw channels
 
-    AVFrame *Frame; ///< decoded audio frame buffer
+    AVFrame *Frame;       ///< decoded audio frame buffer
-
-#ifdef USE_SWRESAMPLE
-#if LIBSWRESAMPLE_VERSION_INT < AV_VERSION_INT(0, 15, 100)
-    struct SwrContext *Resample; ///< ffmpeg software resample context
-#else
     SwrContext *Resample; ///< ffmpeg software resample context
-#endif
-#endif
-#ifdef USE_AVRESAMPLE
-    AVAudioResampleContext *Resample; ///< libav software resample context
-#endif
 
     uint16_t Spdif[24576 / 2]; ///< SPDIF output buffer
     int SpdifIndex;            ///< index into SPDIF output buffer
@@ -810,16 +787,9 @@ void CodecAudioOpen(AudioDecoder *audio_decoder, int codec_id) {
 void CodecAudioClose(AudioDecoder *audio_decoder) {
     // FIXME: output any buffered data
 
-#ifdef USE_SWRESAMPLE
     if (audio_decoder->Resample) {
         swr_free(&audio_decoder->Resample);
     }
-#endif
-#ifdef USE_AVRESAMPLE
-    if (audio_decoder->Resample) {
-        avresample_free(&audio_decoder->Resample);
-    }
-#endif
     if (audio_decoder->AudioCtx) {
         pthread_mutex_lock(&CodecLockMutex);
         avcodec_close(audio_decoder->AudioCtx);
@@ -1094,8 +1064,6 @@ static int CodecAudioPassthroughHelper(AudioDecoder *audio_decoder, const AVPack
     return 0;
 }
 
-#if defined(USE_SWRESAMPLE) || defined(USE_AVRESAMPLE)
-
 /**
 **  Set/update audio pts clock.
 **
@@ -1175,7 +1143,6 @@ static void CodecAudioSetClock(AudioDecoder *audio_decoder, int64_t pts) {
         }
     }
 
-#ifdef USE_SWRESAMPLE
     if (audio_decoder->Resample && audio_decoder->DriftCorr) {
         int distance;
 
@@ -1189,17 +1156,6 @@ static void CodecAudioSetClock(AudioDecoder *audio_decoder, int64_t pts) {
             Debug(3, "codec/audio: swr_set_compensation failed\n");
         }
     }
-#endif
-#ifdef USE_AVRESAMPLE
-    if (audio_decoder->Resample && audio_decoder->DriftCorr) {
-        int distance;
-
-        distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
-        if (avresample_set_compensation(audio_decoder->Resample, audio_decoder->DriftCorr / 10, distance)) {
-            Debug(3, "codec/audio: swr_set_compensation failed\n");
-        }
-    }
-#endif
     if (1) {
         static int c;
 
@@ -1239,7 +1195,6 @@ static void CodecAudioUpdateFormat(AudioDecoder *audio_decoder) {
     }
 #endif
 
-#ifdef USE_SWRESAMPLE
     audio_decoder->Resample = swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout, AV_SAMPLE_FMT_S16,
                                                  audio_decoder->HwSampleRate, audio_ctx->channel_layout,
                                                  audio_ctx->sample_fmt, audio_ctx->sample_rate, 0, NULL);
@@ -1248,27 +1203,6 @@ static void CodecAudioUpdateFormat(AudioDecoder *audio_decoder) {
     } else {
         Error(_("codec/audio: can't setup resample\n"));
     }
-#endif
-#ifdef USE_AVRESAMPLE
-    if (!(audio_decoder->Resample = avresample_alloc_context())) {
-        Error(_("codec/audio: can't setup resample\n"));
-        return;
-    }
-
-    av_opt_set_int(audio_decoder->Resample, "in_channel_layout", audio_ctx->channel_layout, 0);
-    av_opt_set_int(audio_decoder->Resample, "in_sample_fmt", audio_ctx->sample_fmt, 0);
-    av_opt_set_int(audio_decoder->Resample, "in_sample_rate", audio_ctx->sample_rate, 0);
-    av_opt_set_int(audio_decoder->Resample, "out_channel_layout", audio_ctx->channel_layout, 0);
-    av_opt_set_int(audio_decoder->Resample, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
-    av_opt_set_int(audio_decoder->Resample, "out_sample_rate", audio_decoder->HwSampleRate, 0);
-
-    if (avresample_open(audio_decoder->Resample)) {
-        avresample_free(&audio_decoder->Resample);
-        audio_decoder->Resample = NULL;
-        Error(_("codec/audio: can't open resample\n"));
-        return;
-    }
-#endif
 }
 
 /**
@@ -1340,8 +1274,6 @@ void CodecAudioDecode(AudioDecoder *audio_decoder, const AVPacket *avpkt) {
     }
 }
 
-#endif
-
 /**
 **  Flush the audio decoder.
 **

po/de_DE.po

@@ -7,7 +7,7 @@ msgid ""
 msgstr ""
 "Project-Id-Version: VDR \n"
 "Report-Msgid-Bugs-To: <see README>\n"
-"POT-Creation-Date: 2021-12-27 13:13+0100\n"
+"POT-Creation-Date: 2021-12-30 10:23+0100\n"
 "PO-Revision-Date: blabla\n"
 "Last-Translator: blabla\n"
 "Language-Team: blabla\n"
@@ -140,91 +140,6 @@ msgstr ""
 msgid "snd_pcm_drop(): %s\n"
 msgstr ""
 
-#, c-format
-msgid "audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: write error: %s\n"
-msgstr ""
-
-msgid "audio/oss: error not all bytes written\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: error poll %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: using %sdevice '%s'\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: can't open dsp device '%s': %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: ioctl(MIXER_WRITE): %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: can't open mixer device '%s': %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: ioctl(SOUND_MIXER_READ_DEVMASK): %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: channel '%s' not supported\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: channel '%s' not found\n"
-msgstr ""
-
-msgid "audio/oss: should not happen\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: ioctl(SNDCTL_DSP_SETFMT): %s\n"
-msgstr ""
-
-msgid "audio/oss: device doesn't support 16 bit sample format.\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: ioctl(SNDCTL_DSP_CHANNELS): %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: device doesn't support %d channels.\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: ioctl(SNDCTL_DSP_SPEED): %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: device doesn't support %dHz sample rate.\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: ioctl(SNDCTL_DSP_POLICY): %s\n"
-msgstr ""
-
-#, c-format
-msgid "audio/oss: delay %ums\n"
-msgstr ""
-
 #, c-format
 msgid "audio: can't set channels %d sample-rate %dHz\n"
 msgstr ""
@@ -306,9 +221,6 @@ msgstr ""
 msgid "codec/audio: can't setup resample\n"
 msgstr ""
 
-msgid "codec/audio: can't open resample\n"
-msgstr ""
-
 msgid "A software and GPU emulated UHD device"
 msgstr ""