mirror of
https://github.com/jojo61/vdr-plugin-softhdcuvid.git
synced 2023-10-10 13:37:41 +02:00
- 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
This commit is contained in:
Dirk Nehring
parent
43135c43bc
commit
cf1a661c7b
61
Makefile
61
Makefile
@ -54,7 +54,7 @@ endif
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ifeq ($(CUVID),1)
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ifeq ($(DRM),1)
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$(error Missmatch in Plugin selection)
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$(error Mismatch in Plugin selection)
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exit 1;
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endif
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endif
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@ -62,7 +62,7 @@ endif
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ifeq ($(CUVID),1)
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ifeq ($(VAAPI),1)
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$(error Missmatch in Plugin selection)
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$(error Mismatch in Plugin selection)
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exit 1;
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endif
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endif
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@ -78,26 +78,11 @@ PLUGIN = softhdcuvid
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# support OPENGLOSD always needed
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OPENGLOSD=1
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# support alsa audio output module
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ALSA ?= $(shell pkg-config --exists alsa && echo 1)
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# support OSS audio output module
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OSS ?= 1
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# use DMPS
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SCREENSAVER=1
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OPENGL=1
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# use ffmpeg libswresample
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SWRESAMPLE ?= $(shell pkg-config --exists libswresample && echo 1)
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SWRESAMPLE = 1
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# use libav libavresample
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#ifneq ($(SWRESAMPLE),1)
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#AVRESAMPLE ?= $(shell pkg-config --exists libavresample && echo 1#)
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#AVRESAMPLE = 1
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#endif
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CONFIG += -DHAVE_GL # needed for mpv libs
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#CONFIG += -DSTILL_DEBUG=2 # still picture debug verbose level
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CONFIG += -DAV_INFO -DAV_INFO_TIME=3000 # info/debug a/v sync
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@ -149,16 +134,6 @@ APIVERSION = $(call PKGCFG,apiversion)
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### Parse softhddevice config
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ifeq ($(ALSA),1)
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CONFIG += -DUSE_ALSA
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_CFLAGS += $(shell pkg-config --cflags alsa)
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LIBS += $(shell pkg-config --libs alsa)
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endif
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ifeq ($(OSS),1)
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CONFIG += -DUSE_OSS
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endif
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ifeq ($(OPENGL),1)
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#_CFLAGS += $(shell pkg-config --cflags libva-glx)
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#LIBS += $(shell pkg-config --libs libva-glx)
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@ -226,11 +201,13 @@ SOFILE = libvdr-$(PLUGIN).so
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#
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# Test that libswresample is available
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#
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#ifneq (exists, $(shell pkg-config libswresample && echo exists))
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# $(warning ******************************************************************)
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# $(warning 'libswresample' not found!)
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# $(error ******************************************************************)
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#endif
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ifneq (exists, $(shell pkg-config libswresample && echo exists))
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$(warning ******************************************************************)
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$(warning 'libswresample' not found!)
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$(error ******************************************************************)
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endif
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_CFLAGS += $(shell pkg-config --cflags libswresample)
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LIBS += $(shell pkg-config --libs libswresample)
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#
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# Test and set config for libavutil
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@ -265,22 +242,22 @@ endif
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_CFLAGS += $(shell pkg-config --cflags libavcodec)
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LIBS += $(shell pkg-config --libs libavcodec libavfilter)
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#
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# Test and set config for alsa
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#
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ifneq (exists, $(shell pkg-config alsa && echo exists))
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$(warning ******************************************************************)
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$(warning 'alsa' not found!)
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$(error ******************************************************************)
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endif
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_CFLAGS += $(shell pkg-config --cflags alsa)
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LIBS += $(shell pkg-config --libs alsa)
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ifeq ($(SCREENSAVER),1)
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CONFIG += -DUSE_SCREENSAVER
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_CFLAGS += $(shell pkg-config --cflags xcb-screensaver xcb-dpms)
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LIBS += $(shell pkg-config --libs xcb-screensaver xcb-dpms)
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endif
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ifeq ($(SWRESAMPLE),1)
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CONFIG += -DUSE_SWRESAMPLE
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_CFLAGS += $(shell pkg-config --cflags libswresample)
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LIBS += $(shell pkg-config --libs libswresample)
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endif
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ifeq ($(AVRESAMPLE),1)
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CONFIG += -DUSE_AVRESAMPLE
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_CFLAGS += $(shell pkg-config --cflags libavresample)
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LIBS += $(shell pkg-config --libs libavresample)
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endif
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#_CFLAGS += $(shell pkg-config --cflags libavcodec x11 x11-xcb xcb xcb-icccm)
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#LIBS += -lrt $(shell pkg-config --libs libavcodec x11 x11-xcb xcb xcb-icccm)
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39
README.md
39
README.md
@ -25,14 +25,13 @@ A software and GPU emulated UHD output device plugin for VDR.
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o Video decoder CUVID or VAAPI
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o Video output opengl or DRM
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o Audio FFMpeg / Alsa / Analog
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o Audio FFMpeg / Alsa / Digital
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o Audio FFMpeg / OSS / Analog
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o Audio FFMpeg / ALSA / Analog
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o Audio FFMpeg / ALSA / Digital
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o HDMI/SPDIF pass-through
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o Software volume, compression, normalize and channel resample
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o VDR ScaleVideo API
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o CUDA deinterlacer
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o Suspend / Dettach
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o Suspend / Detach
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o Support for ambilight
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o Support for Screencopy
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o PIP (Picture-in-Picture) (only for CUVID)
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@ -96,9 +95,6 @@ Install:
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make
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make install
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You can edit Makefile to enable/disable Alsa / OSS
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support. The default is to autodetect as much as possible.
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You have to start vdr with -P 'softhdcuvid -d :0.0 ..<more option>.. '
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Beginners Guide for libplacebo:
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@ -148,27 +144,17 @@ Setup: environment
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Following is supported:
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DISPLAY=:0.0
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x11 display name
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X11 display name
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only if alsa is configured
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ALSA configuration:
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ALSA_DEVICE=default
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alsa PCM device name
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ALSA PCM device name
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ALSA_PASSTHROUGH_DEVICE=
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alsa pass-though (AC-3,E-AC-3,DTS,...) device name
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ALSA pass-though (AC-3,E-AC-3,DTS,...) device name
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ALSA_MIXER=default
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alsa control device name
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ALSA control device name
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ALSA_MIXER_CHANNEL=PCM
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alsa control channel name
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only if oss is configured
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OSS_AUDIODEV=/dev/dsp
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oss dsp device name
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OSS_PASSTHROUGHDEV=
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oss pass-though (AC-3,E-AC-3,DTS,...) device name
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OSS_MIXERDEV=/dev/mixer
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oss mixer device name
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OSS_MIXER_CHANNEL=pcm
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oss mixer channel name
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ALSA control channel name
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Setup: /etc/vdr/setup.conf
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------
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@ -197,7 +183,7 @@ Setup: /etc/vdr/setup.conf
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(only 0, 1, 4 supported with VA-API)
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softhddevice.<res>.SkipChromaDeinterlace = 0
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0 = disabled, 1 = enabled (for slower cards, poor qualität)
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0 = disabled, 1 = enabled (for slower cards, poor qualit<EFBFBD>t)
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softhddevice.<res>.InverseTelecine = 0
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0 = disabled, 1 = enabled
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@ -336,10 +322,7 @@ Commandline:
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Selects audio output module and device.
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"" to disable audio output
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/... to use oss audio module (if compiled with oss
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support)
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other to use alsa audio module (if compiled with alsa
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support)
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other to use ALSA audio module
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SVDRP:
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------
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520
audio.c
520
audio.c
@ -30,17 +30,12 @@
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///
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/// @note alsa async playback is broken, don't use it!
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///
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/// OSS PCM/Mixer api is supported.
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/// @see http://manuals.opensound.com/developer/
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///
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///
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/// @todo FIXME: there can be problems with little/big endian.
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///
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#ifdef DEBUG
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#undef DEBUG
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#endif
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//#define USE_ALSA ///< enable alsa support
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//#define USE_OSS ///< enable OSS support
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#define USE_AUDIO_THREAD ///< use thread for audio playback
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#define USE_AUDIO_MIXER ///< use audio module mixer
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@ -57,41 +52,12 @@
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#define _(str) gettext(str) ///< gettext shortcut
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#define _N(str) str ///< gettext_noop shortcut
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#ifdef USE_ALSA
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#include <alsa/asoundlib.h>
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#endif
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#ifdef USE_OSS
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#include <sys/ioctl.h>
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#include <sys/soundcard.h>
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#include <sys/stat.h>
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#include <sys/types.h>
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// SNDCTL_DSP_HALT_OUTPUT compatibility
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#ifndef SNDCTL_DSP_HALT_OUTPUT
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#if defined(SNDCTL_DSP_RESET_OUTPUT)
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#define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET_OUTPUT
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#elif defined(SNDCTL_DSP_RESET)
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#define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET
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#else
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#error "No valid SNDCTL_DSP_HALT_OUTPUT found."
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#endif
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#endif
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#include <errno.h>
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#include <fcntl.h>
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#include <poll.h>
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#include <unistd.h>
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#endif
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#ifdef USE_AUDIO_THREAD
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#ifndef __USE_GNU
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#define __USE_GNU
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#endif
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#include <pthread.h>
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#include <sys/resource.h>
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#include <sys/syscall.h>
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#ifndef HAVE_PTHREAD_NAME
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/// only available with newer glibc
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#define pthread_setname_np(thread, name)
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#endif
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#endif
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#include "iatomic.h" // portable atomic_t
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@ -723,8 +689,6 @@ static void AudioRingExit(void) {
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AudioRingWrite = 0;
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}
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#ifdef USE_ALSA
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//============================================================================
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// A L S A
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//============================================================================
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@ -1358,480 +1322,6 @@ static const AudioModule AlsaModule = {
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.Exit = AlsaExit,
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};
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#endif // USE_ALSA
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#ifdef USE_OSS
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//============================================================================
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// O S S
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//============================================================================
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//----------------------------------------------------------------------------
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// OSS variables
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//----------------------------------------------------------------------------
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static int OssPcmFildes = -1; ///< pcm file descriptor
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static int OssMixerFildes = -1; ///< mixer file descriptor
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static int OssMixerChannel; ///< mixer channel index
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static int OssFragmentTime; ///< fragment time in ms
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//----------------------------------------------------------------------------
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// OSS pcm
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//----------------------------------------------------------------------------
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/**
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** Play samples from ringbuffer.
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**
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** @retval 0 ok
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** @retval 1 ring buffer empty
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** @retval -1 underrun error
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*/
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static int OssPlayRingbuffer(void) {
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int first;
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first = 1;
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for (;;) {
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audio_buf_info bi;
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const void *p;
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int n;
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if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"), strerror(errno));
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return -1;
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}
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Debug(4, "audio/oss: %d bytes free\n", bi.bytes);
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n = RingBufferGetReadPointer(AudioRing[AudioRingRead].RingBuffer, &p);
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if (!n) { // ring buffer empty
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if (first) { // only error on first loop
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return 1;
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}
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return 0;
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}
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if (n < bi.bytes) { // not enough bytes in ring buffer
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bi.bytes = n;
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}
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if (bi.bytes <= 0) { // full or buffer empty
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break; // bi.bytes could become negative!
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}
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if (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough) {
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// FIXME: quick&dirty cast
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AudioSoftAmplifier((int16_t *)p, bi.bytes);
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// FIXME: if not all are written, we double amplify them
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}
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for (;;) {
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n = write(OssPcmFildes, p, bi.bytes);
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if (n != bi.bytes) {
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if (n < 0) {
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if (n == EAGAIN) {
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continue;
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}
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Error(_("audio/oss: write error: %s\n"), strerror(errno));
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return 1;
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}
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Warning(_("audio/oss: error not all bytes written\n"));
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}
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break;
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}
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// advance how many could written
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RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, n);
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first = 0;
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}
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return 0;
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}
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/**
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** Flush OSS buffers.
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*/
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static void OssFlushBuffers(void) {
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if (OssPcmFildes != -1) {
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// flush kernel buffers
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if (ioctl(OssPcmFildes, SNDCTL_DSP_HALT_OUTPUT, NULL) < 0) {
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Error(_("audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"), strerror(errno));
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}
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}
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}
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#ifdef USE_AUDIO_THREAD
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//----------------------------------------------------------------------------
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// thread playback
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//----------------------------------------------------------------------------
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/**
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** OSS thread
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**
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** @retval -1 error
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** @retval 0 underrun
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** @retval 1 running
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*/
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static int OssThread(void) {
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int err;
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if (!OssPcmFildes) {
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usleep(OssFragmentTime * 1000);
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return -1;
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}
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for (;;) {
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struct pollfd fds[1];
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if (AudioPaused) {
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return 1;
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}
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// wait for space in kernel buffers
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fds[0].fd = OssPcmFildes;
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fds[0].events = POLLOUT | POLLERR;
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// wait for space in kernel buffers
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err = poll(fds, 1, OssFragmentTime);
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if (err < 0) {
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if (err == EAGAIN) {
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continue;
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}
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Error(_("audio/oss: error poll %s\n"), strerror(errno));
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usleep(OssFragmentTime * 1000);
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return -1;
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}
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break;
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}
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if (!err || AudioPaused) { // timeout or some commands
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return 1;
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}
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if ((err = OssPlayRingbuffer())) { // empty / error
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if (err < 0) { // underrun error
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return -1;
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}
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sched_yield();
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usleep(OssFragmentTime * 1000); // let fill/empty the buffers
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return 0;
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}
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return 1;
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}
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#endif
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//----------------------------------------------------------------------------
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/**
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** Open OSS pcm device.
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**
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** @param passthrough use pass-through (AC-3, ...) device
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*/
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static int OssOpenPCM(int passthrough) {
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const char *device;
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int fildes;
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// &&|| hell
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if (!(passthrough && ((device = AudioPassthroughDevice) || (device = getenv("OSS_PASSTHROUGHDEV")))) &&
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!(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
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device = "/dev/dsp";
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}
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if (!AudioDoingInit) {
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Info(_("audio/oss: using %sdevice '%s'\n"), passthrough ? "pass-through " : "", device);
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}
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if ((fildes = open(device, O_WRONLY)) < 0) {
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Error(_("audio/oss: can't open dsp device '%s': %s\n"), device, strerror(errno));
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return -1;
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}
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return fildes;
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}
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/**
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** Initialize OSS pcm device.
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**
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** @see AudioPCMDevice
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*/
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static void OssInitPCM(void) {
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int fildes;
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fildes = OssOpenPCM(0);
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OssPcmFildes = fildes;
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}
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//----------------------------------------------------------------------------
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// OSS Mixer
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//----------------------------------------------------------------------------
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/**
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** Set OSS mixer volume (0-1000)
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**
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** @param volume volume (0 .. 1000)
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*/
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static void OssSetVolume(int volume) {
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int v;
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if (OssMixerFildes != -1) {
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v = (volume * 255) / 1000;
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v &= 0xff;
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v = (v << 8) | v;
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if (ioctl(OssMixerFildes, MIXER_WRITE(OssMixerChannel), &v) < 0) {
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Error(_("audio/oss: ioctl(MIXER_WRITE): %s\n"), strerror(errno));
|
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}
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}
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}
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|
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/**
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** Mixer channel name table.
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*/
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static const char *OssMixerChannelNames[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
|
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|
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/**
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** Initialize OSS mixer.
|
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*/
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static void OssInitMixer(void) {
|
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const char *device;
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const char *channel;
|
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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;
|
||||
}
|
||||
|
||||
70
codec.c
70
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
|
||||
|
||||
#ifdef USE_SWRESAMPLE
|
||||
#if LIBSWRESAMPLE_VERSION_INT < AV_VERSION_INT(0, 15, 100)
|
||||
struct SwrContext *Resample; ///< ffmpeg software resample context
|
||||
#else
|
||||
AVFrame *Frame; ///< decoded audio frame buffer
|
||||
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.
|
||||
**
|
||||
|
||||
90
po/de_DE.po
90
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 ""
|
||||
|
||||
|
||||
Loading…
Reference in New Issue
Block a user