mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
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2371 lines
56 KiB
C
2371 lines
56 KiB
C
///
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/// @file audio.c @brief Audio module
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///
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/// Copyright (c) 2009 - 2012 by Johns. All Rights Reserved.
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///
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/// Contributor(s):
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///
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/// License: AGPLv3
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///
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/// This program is free software: you can redistribute it and/or modify
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/// it under the terms of the GNU Affero General Public License as
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/// published by the Free Software Foundation, either version 3 of the
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/// License.
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///
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/// This program is distributed in the hope that it will be useful,
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/// but WITHOUT ANY WARRANTY; without even the implied warranty of
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/// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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/// GNU Affero General Public License for more details.
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///
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/// $Id$
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//////////////////////////////////////////////////////////////////////////////
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///
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/// @defgroup Audio The audio module.
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///
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/// This module contains all audio output functions.
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///
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/// ALSA PCM/Mixer api is supported.
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/// @see http://www.alsa-project.org/alsa-doc/alsa-lib
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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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/// @todo FIXME: can combine OSS and alsa ring buffer
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///
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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 noUSE_AUDIORING ///< new audio ring code (incomplete)
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#include <stdio.h>
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#include <stdint.h>
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#include <stdlib.h>
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#include <inttypes.h>
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#include <string.h>
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#include <libintl.h>
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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/types.h>
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#include <sys/stat.h>
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#include <sys/ioctl.h>
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#include <sys/soundcard.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 <poll.h>
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#include <unistd.h>
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#include <fcntl.h>
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#include <errno.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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#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 <alsa/iatomic.h> // portable atomic_t
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#include "ringbuffer.h"
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#include "misc.h"
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#include "audio.h"
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//----------------------------------------------------------------------------
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// Declarations
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//----------------------------------------------------------------------------
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/**
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** Audio output module structure and typedef.
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*/
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typedef struct _audio_module_
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{
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const char *Name; ///< audio output module name
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void (*Thread) (void); ///< module thread handler
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void (*Enqueue) (const void *, int); ///< enqueue samples for output
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void (*FlushBuffers) (void); ///< flush sample buffers
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void (*Poller) (void); ///< output poller
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int (*FreeBytes) (void); ///< number of bytes free in buffer
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uint64_t(*GetDelay) (void); ///< get current audio delay
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void (*SetVolume) (int); ///< set output volume
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int (*Setup) (int *, int *, int); ///< setup channels, samplerate
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void (*Init) (void); ///< initialize audio output module
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void (*Exit) (void); ///< cleanup audio output module
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} AudioModule;
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static const AudioModule NoopModule; ///< forward definition of noop module
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//----------------------------------------------------------------------------
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// Variables
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//----------------------------------------------------------------------------
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char AudioAlsaDriverBroken; ///< disable broken driver message
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static const char *AudioModuleName; ///< which audio module to use
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/// Selected audio module.
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static const AudioModule *AudioUsedModule = &NoopModule;
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static const char *AudioPCMDevice; ///< alsa/OSS PCM device name
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static const char *AudioAC3Device; ///< alsa/OSS AC3 device name
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static const char *AudioMixerDevice; ///< alsa/OSS mixer device name
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static const char *AudioMixerChannel; ///< alsa/OSS mixer channel name
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static volatile char AudioRunning; ///< thread running / stopped
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static int AudioPaused; ///< audio paused
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static unsigned AudioSampleRate; ///< audio sample rate in hz
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static unsigned AudioChannels; ///< number of audio channels
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static const int AudioBytesProSample = 2; ///< number of bytes per sample
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static int64_t AudioPTS; ///< audio pts clock
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static const int AudioBufferTime = 350; ///< audio buffer time in ms
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#ifdef USE_AUDIO_THREAD
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static pthread_t AudioThread; ///< audio play thread
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static pthread_mutex_t AudioMutex; ///< audio condition mutex
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static pthread_cond_t AudioStartCond; ///< condition variable
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#else
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static const int AudioThread; ///< dummy audio thread
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#endif
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extern int VideoAudioDelay; /// import audio/video delay
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#ifdef USE_AUDIORING
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//----------------------------------------------------------------------------
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// ring buffer
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//----------------------------------------------------------------------------
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// FIXME: use this code, to combine alsa&OSS ring buffers
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#define AUDIO_RING_MAX 8 ///< number of audio ring buffers
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/**
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** Audio ring buffer.
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*/
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typedef struct _audio_ring_ring_
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{
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char FlushBuffers; ///< flag: flush buffers
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unsigned SampleRate; ///< sample rate in hz
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unsigned Channels; ///< number of channels
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} AudioRingRing;
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/// ring of audio ring buffers
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static AudioRingRing AudioRing[AUDIO_RING_MAX];
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static int AudioRingWrite; ///< audio ring write pointer
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static int AudioRingRead; ///< audio ring read pointer
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static atomic_t AudioRingFilled; ///< how many of the ring is used
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/**
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** Add sample rate, number of channel change to ring.
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**
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** @param freq sample frequency
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** @param channels number of channels
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*/
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static int AudioRingAdd(int freq, int channels)
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{
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int filled;
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filled = atomic_read(&AudioRingFilled);
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if (filled == AUDIO_RING_MAX) { // no free slot
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// FIXME: can wait for ring buffer empty
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Error(_("audio: out of ring buffers\n"));
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return -1;
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}
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AudioRing[AudioRingWrite].FlushBuffers = 1;
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AudioRing[AudioRingWrite].SampleRate = freq;
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AudioRing[AudioRingWrite].Channels = channels;
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AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
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atomic_inc(&AudioRingFilled);
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#ifdef USE_AUDIO_THREAD
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// tell thread, that something todo
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AudioRunning = 1;
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pthread_cond_signal(&AudioStartCond);
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#endif
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return 0;
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}
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/**
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** Setup audio ring.
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*/
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static void AudioRingInit(void)
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{
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int i;
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for (i = 0; i < AUDIO_RING_MAX; ++i) {
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// FIXME:
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//AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit
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}
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// one slot always reservered
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AudioRingWrite = 1;
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atomic_set(&AudioRingFilled, 1);
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}
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/**
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** Cleanup audio ring.
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*/
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static void AudioRingExit(void)
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{
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int i;
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for (i = 0; i < AUDIO_RING_MAX; ++i) {
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// FIXME:
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//RingBufferDel(AlsaRingBuffer);
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}
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}
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#endif
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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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//----------------------------------------------------------------------------
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// Alsa variables
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//----------------------------------------------------------------------------
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static snd_pcm_t *AlsaPCMHandle; ///< alsa pcm handle
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static char AlsaCanPause; ///< hw supports pause
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static int AlsaUseMmap; ///< use mmap
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static RingBuffer *AlsaRingBuffer; ///< audio ring buffer
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static unsigned AlsaStartThreshold; ///< start play, if filled
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#ifdef USE_AUDIO_THREAD
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static volatile char AlsaFlushBuffer; ///< flag empty buffer
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#endif
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static snd_mixer_t *AlsaMixer; ///< alsa mixer handle
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static snd_mixer_elem_t *AlsaMixerElem; ///< alsa pcm mixer element
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static int AlsaRatio; ///< internal -> mixer ratio * 1000
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//----------------------------------------------------------------------------
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// alsa pcm
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//----------------------------------------------------------------------------
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/**
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** Place samples in ringbuffer.
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**
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** @param samples sample buffer
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** @param count number of bytes in sample buffer
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**
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** @returns true if play should be started.
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*/
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static int AlsaAddToRingbuffer(const void *samples, int count)
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{
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int n;
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n = RingBufferWrite(AlsaRingBuffer, samples, count);
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if (n != count) {
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Error(_("audio/alsa: can't place %d samples in ring buffer\n"), count);
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// too many bytes are lost
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// FIXME: should skip more, longer skip, but less often?
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}
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// Update audio clock
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AudioPTS +=
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((int64_t) count * 90000) / (AudioSampleRate * AudioChannels *
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AudioBytesProSample);
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if (!AudioRunning) {
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if (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
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// restart play-back
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return 1;
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}
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}
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return 0;
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}
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/**
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** Play samples from ringbuffer.
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*/
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static int AlsaPlayRingbuffer(void)
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{
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int first;
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int avail;
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int n;
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int err;
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int frames;
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const void *p;
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first = 1;
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for (;;) {
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// how many bytes can be written?
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n = snd_pcm_avail_update(AlsaPCMHandle);
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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/alsa: underrun error?\n"));
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err = snd_pcm_recover(AlsaPCMHandle, n, 0);
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if (err >= 0) {
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continue;
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}
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Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"),
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snd_strerror(n));
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return -1;
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}
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avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n);
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if (avail < 256) { // too much overhead
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if (first) {
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// happens with broken alsa drivers
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if (AudioThread) {
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if (!AudioAlsaDriverBroken) {
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Error(_("audio/alsa: broken driver %d\n"), avail);
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}
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usleep(5 * 1000);
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}
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}
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Debug(4, "audio/alsa: break state %s\n",
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snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
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break;
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}
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n = RingBufferGetReadPointer(AlsaRingBuffer, &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 < avail) { // not enough bytes in ring buffer
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avail = n;
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}
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if (!avail) { // full or buffer empty
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break;
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}
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frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail);
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again:
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if (AlsaUseMmap) {
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err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames);
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} else {
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err = snd_pcm_writei(AlsaPCMHandle, p, frames);
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}
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//Debug(3, "audio/alsa: wrote %d/%d frames\n", err, frames);
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if (err != frames) {
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if (err < 0) {
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if (err == -EAGAIN) {
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goto again;
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}
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/*
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if (err == -EBADFD) {
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goto again;
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}
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*/
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Error(_("audio/alsa: underrun error?\n"));
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err = snd_pcm_recover(AlsaPCMHandle, err, 0);
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if (err >= 0) {
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goto again;
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}
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Error(_("audio/alsa: snd_pcm_writei failed: %s\n"),
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snd_strerror(err));
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return -1;
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}
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// this could happen, if underrun happened
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Error(_("audio/alsa: error not all frames written\n"));
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avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err);
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}
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RingBufferReadAdvance(AlsaRingBuffer, avail);
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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 alsa buffers.
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*/
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static void AlsaFlushBuffers(void)
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{
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int err;
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snd_pcm_state_t state;
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if (AlsaRingBuffer && AlsaPCMHandle) {
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RingBufferReadAdvance(AlsaRingBuffer,
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RingBufferUsedBytes(AlsaRingBuffer));
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state = snd_pcm_state(AlsaPCMHandle);
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Debug(3, "audio/alsa: state %d - %s\n", state,
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snd_pcm_state_name(state));
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if (state != SND_PCM_STATE_OPEN) {
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if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
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Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
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}
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// ****ing alsa crash, when in open state here
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if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
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Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err));
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}
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}
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}
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AudioRunning = 0;
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AudioPTS = INT64_C(0x8000000000000000);
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}
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/**
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** Call back to play audio polled.
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*/
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static void AlsaPoller(void)
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{
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if (!AlsaPCMHandle) { // setup failure
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return;
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}
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if (!AudioThread && AudioRunning) {
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AlsaPlayRingbuffer();
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}
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}
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/**
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** Get free bytes in audio output.
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*/
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static int AlsaFreeBytes(void)
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{
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return AlsaRingBuffer ? RingBufferFreeBytes(AlsaRingBuffer) : INT32_MAX;
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}
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#if 0
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//----------------------------------------------------------------------------
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// async playback
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//----------------------------------------------------------------------------
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// async playback is broken, don't use it!
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/**
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** Alsa async pcm callback function.
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**
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** @param handler alsa async handler
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*/
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static void AlsaAsyncCallback(snd_async_handler_t * handler)
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{
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Debug(3, "audio/%s: %p\n", __FUNCTION__, handler);
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// how many bytes can be written?
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for (;;) {
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n = snd_pcm_avail_update(AlsaPCMHandle);
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if (n < 0) {
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Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"),
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snd_strerror(n));
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break;
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}
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avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n);
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if (avail < 512) { // too much overhead
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break;
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}
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n = RingBufferGetReadPointer(AlsaRingBuffer, &p);
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if (!n) { // ring buffer empty
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Debug(3, "audio/alsa: ring buffer empty\n");
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break;
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}
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if (n < avail) { // not enough bytes in ring buffer
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avail = n;
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}
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if (!avail) { // full
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break;
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}
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frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail);
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again:
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if (AlsaUseMmap) {
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err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames);
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} else {
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err = snd_pcm_writei(AlsaPCMHandle, p, frames);
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}
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Debug(3, "audio/alsa: %d => %d\n", frames, err);
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if (err < 0) {
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Error(_("audio/alsa: underrun error?\n"));
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err = snd_pcm_recover(AlsaPCMHandle, err, 0);
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if (err >= 0) {
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goto again;
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}
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Error(_("audio/alsa: snd_pcm_writei failed: %s\n"),
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snd_strerror(err));
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}
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if (err != frames) {
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Error(_("audio/alsa: error not all frames written\n"));
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avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err);
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}
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RingBufferReadAdvance(AlsaRingBuffer, avail);
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}
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}
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|
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/**
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** Place samples in audio output queue.
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**
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** @param samples sample buffer
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** @param count number of bytes in sample buffer
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*/
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static void AlsaEnqueue(const void *samples, int count)
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{
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snd_pcm_state_t state;
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int n;
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//int err;
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Debug(3, "audio: %6zd + %4d\n", RingBufferUsedBytes(AlsaRingBuffer),
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count);
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n = RingBufferWrite(AlsaRingBuffer, samples, count);
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if (n != count) {
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Fatal(_("audio: can't place %d samples in ring buffer\n"), count);
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}
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// check if running, wait until enough buffered
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state = snd_pcm_state(AlsaPCMHandle);
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if (state == SND_PCM_STATE_PREPARED) {
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Debug(3, "audio/alsa: state %d - %s\n", state,
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snd_pcm_state_name(state));
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// FIXME: adjust start ratio
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if (RingBufferFreeBytes(AlsaRingBuffer)
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< RingBufferUsedBytes(AlsaRingBuffer)) {
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// restart play-back
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#if 0
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if (AlsaCanPause) {
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if ((err = snd_pcm_pause(AlsaPCMHandle, 0))) {
|
|
Error(_("audio: snd_pcm_pause(): %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
} else {
|
|
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
|
|
Error(_("audio: snd_pcm_prepare(): %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
}
|
|
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
|
|
Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err));
|
|
}
|
|
|
|
Debug(3, "audio/alsa: unpaused\n");
|
|
if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) {
|
|
Error(_("audio: snd_pcm_start(): %s\n"), snd_strerror(err));
|
|
}
|
|
#endif
|
|
state = snd_pcm_state(AlsaPCMHandle);
|
|
Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(state));
|
|
Debug(3, "audio/alsa: unpaused\n");
|
|
AudioPaused = 0;
|
|
}
|
|
}
|
|
// Update audio clock
|
|
// AudioPTS += (size * 90000) / (AudioSampleRate * AudioChannels * AudioBytesProSample);
|
|
}
|
|
|
|
#endif
|
|
|
|
//----------------------------------------------------------------------------
|
|
// direct playback
|
|
//----------------------------------------------------------------------------
|
|
|
|
// direct play produces underuns on some hardware
|
|
|
|
#ifndef USE_AUDIO_THREAD
|
|
|
|
/**
|
|
** Place samples in audio output queue.
|
|
**
|
|
** @param samples sample buffer
|
|
** @param count number of bytes in sample buffer
|
|
*/
|
|
static void AlsaEnqueue(const void *samples, int count)
|
|
{
|
|
if (AlsaAddToRingbuffer(samples, count)) {
|
|
AudioRunning = 1;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
#ifdef USE_AUDIO_THREAD
|
|
|
|
//----------------------------------------------------------------------------
|
|
// thread playback
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Alsa thread
|
|
*/
|
|
static void AlsaThread(void)
|
|
{
|
|
for (;;) {
|
|
int err;
|
|
|
|
pthread_testcancel();
|
|
if (AlsaFlushBuffer) {
|
|
// we can flush too many, but wo cares
|
|
Debug(3, "audio/alsa: flushing buffers\n");
|
|
AlsaFlushBuffers();
|
|
/*
|
|
if ((err = snd_pcm_prepare(AlsaPCMHandle))) {
|
|
Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err));
|
|
}
|
|
*/
|
|
AlsaFlushBuffer = 0;
|
|
break;
|
|
}
|
|
// wait for space in kernel buffers
|
|
if ((err = snd_pcm_wait(AlsaPCMHandle, 100)) < 0) {
|
|
Error(_("audio/alsa: wait underrun error?\n"));
|
|
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
|
|
if (err >= 0) {
|
|
continue;
|
|
}
|
|
Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err));
|
|
usleep(100 * 1000);
|
|
continue;
|
|
}
|
|
if (AlsaFlushBuffer) {
|
|
continue;
|
|
}
|
|
if ((err = AlsaPlayRingbuffer())) { // empty / error
|
|
snd_pcm_state_t state;
|
|
|
|
if (err < 0) { // underrun error
|
|
break;
|
|
}
|
|
state = snd_pcm_state(AlsaPCMHandle);
|
|
if (state != SND_PCM_STATE_RUNNING) {
|
|
Debug(3, "audio/alsa: stopping play\n");
|
|
break;
|
|
}
|
|
pthread_yield();
|
|
usleep(20 * 1000); // let fill/empty the buffers
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Place samples in audio output queue.
|
|
**
|
|
** @param samples sample buffer
|
|
** @param count number of bytes in sample buffer
|
|
*/
|
|
static void AlsaThreadEnqueue(const void *samples, int count)
|
|
{
|
|
if (!AlsaRingBuffer || !AlsaPCMHandle || !AudioSampleRate) {
|
|
Debug(3, "audio/alsa: enqueue not ready\n");
|
|
return;
|
|
}
|
|
if (AlsaAddToRingbuffer(samples, count)) {
|
|
snd_pcm_state_t state;
|
|
|
|
state = snd_pcm_state(AlsaPCMHandle);
|
|
Debug(3, "audio/alsa: enqueue state %s\n", snd_pcm_state_name(state));
|
|
|
|
// no lock needed, can wakeup next time
|
|
AudioRunning = 1;
|
|
pthread_cond_signal(&AudioStartCond);
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Flush alsa buffers with thread.
|
|
*/
|
|
static void AlsaThreadFlushBuffers(void)
|
|
{
|
|
// signal thread to flush buffers
|
|
if (AudioThread) {
|
|
AlsaFlushBuffer = 1;
|
|
do {
|
|
AudioRunning = 1; // wakeup in case of sleeping
|
|
pthread_cond_signal(&AudioStartCond);
|
|
usleep(1 * 1000);
|
|
} while (AlsaFlushBuffer); // wait until flushed
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Open alsa pcm device.
|
|
**
|
|
** @param use_ac3 use ac3/pass-through device
|
|
*/
|
|
static snd_pcm_t *AlsaOpenPCM(int use_ac3)
|
|
{
|
|
const char *device;
|
|
snd_pcm_t *handle;
|
|
int err;
|
|
|
|
// &&|| hell
|
|
if (!(use_ac3 && ((device = AudioAC3Device)
|
|
|| (device = getenv("ALSA_AC3_DEVICE"))
|
|
|| (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
|
|
&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
|
|
device = "default";
|
|
}
|
|
Debug(3, "audio/alsa: &&|| hell '%s'\n", device);
|
|
|
|
// open none blocking; if device is already used, we don't want wait
|
|
if ((err =
|
|
snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK,
|
|
SND_PCM_NONBLOCK)) < 0) {
|
|
Error(_("audio/alsa: playback open '%s' error: %s\n"), device,
|
|
snd_strerror(err));
|
|
return NULL;
|
|
}
|
|
|
|
if ((err = snd_pcm_nonblock(handle, 0)) < 0) {
|
|
Error(_("audio/alsa: can't set block mode: %s\n"), snd_strerror(err));
|
|
}
|
|
return handle;
|
|
}
|
|
|
|
/**
|
|
** Initialize alsa pcm device.
|
|
**
|
|
** @see AudioPCMDevice
|
|
*/
|
|
static void AlsaInitPCM(void)
|
|
{
|
|
snd_pcm_t *handle;
|
|
snd_pcm_hw_params_t *hw_params;
|
|
int err;
|
|
snd_pcm_uframes_t buffer_size;
|
|
|
|
if (!(handle = AlsaOpenPCM(0))) {
|
|
return;
|
|
}
|
|
|
|
snd_pcm_hw_params_alloca(&hw_params);
|
|
// choose all parameters
|
|
if ((err = snd_pcm_hw_params_any(handle, hw_params)) < 0) {
|
|
Error(_
|
|
("audio: snd_pcm_hw_params_any: no configurations available: %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params);
|
|
Info(_("audio/alsa: supports pause: %s\n"), AlsaCanPause ? "yes" : "no");
|
|
snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
|
|
Info(_("audio/alsa: max buffer size %lu\n"), buffer_size);
|
|
|
|
AlsaPCMHandle = handle;
|
|
}
|
|
|
|
//----------------------------------------------------------------------------
|
|
// Alsa Mixer
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Set alsa mixer volume (0-100)
|
|
**
|
|
** @param volume volume (0 .. 100)
|
|
*/
|
|
static void AlsaSetVolume(int volume)
|
|
{
|
|
int v;
|
|
|
|
if (AlsaMixer && AlsaMixerElem) {
|
|
v = (volume * AlsaRatio) / 1000;
|
|
snd_mixer_selem_set_playback_volume(AlsaMixerElem, 0, v);
|
|
snd_mixer_selem_set_playback_volume(AlsaMixerElem, 1, v);
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Initialize alsa mixer.
|
|
*/
|
|
static void AlsaInitMixer(void)
|
|
{
|
|
const char *device;
|
|
const char *channel;
|
|
snd_mixer_t *alsa_mixer;
|
|
snd_mixer_elem_t *alsa_mixer_elem;
|
|
long alsa_mixer_elem_min;
|
|
long alsa_mixer_elem_max;
|
|
|
|
if (!(device = AudioMixerDevice)) {
|
|
if (!(device = getenv("ALSA_MIXER"))) {
|
|
device = "default";
|
|
}
|
|
}
|
|
if (!(channel = AudioMixerChannel)) {
|
|
if (!(channel = getenv("ALSA_MIXER_CHANNEL"))) {
|
|
channel = "PCM";
|
|
}
|
|
}
|
|
Debug(3, "audio/alsa: mixer %s - %s open\n", device, channel);
|
|
snd_mixer_open(&alsa_mixer, 0);
|
|
if (alsa_mixer && snd_mixer_attach(alsa_mixer, device) >= 0
|
|
&& snd_mixer_selem_register(alsa_mixer, NULL, NULL) >= 0
|
|
&& snd_mixer_load(alsa_mixer) >= 0) {
|
|
|
|
const char *const alsa_mixer_elem_name = channel;
|
|
|
|
alsa_mixer_elem = snd_mixer_first_elem(alsa_mixer);
|
|
while (alsa_mixer_elem) {
|
|
const char *name;
|
|
|
|
name = snd_mixer_selem_get_name(alsa_mixer_elem);
|
|
if (!strcasecmp(name, alsa_mixer_elem_name)) {
|
|
snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem,
|
|
&alsa_mixer_elem_min, &alsa_mixer_elem_max);
|
|
AlsaRatio =
|
|
(1000 * (alsa_mixer_elem_max - alsa_mixer_elem_min)) / 100;
|
|
Debug(3, "audio/alsa: PCM mixer found %ld - %ld ratio %d\n",
|
|
alsa_mixer_elem_min, alsa_mixer_elem_max, AlsaRatio);
|
|
break;
|
|
}
|
|
|
|
alsa_mixer_elem = snd_mixer_elem_next(alsa_mixer_elem);
|
|
}
|
|
|
|
AlsaMixer = alsa_mixer;
|
|
AlsaMixerElem = alsa_mixer_elem;
|
|
} else {
|
|
Error(_("audio/alsa: can't open mixer '%s'\n"), device);
|
|
}
|
|
}
|
|
|
|
//----------------------------------------------------------------------------
|
|
// Alsa API
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Get alsa audio delay in time stamps.
|
|
**
|
|
** @returns audio delay in time stamps.
|
|
**
|
|
** @todo FIXME: handle the case no audio running
|
|
*/
|
|
static uint64_t AlsaGetDelay(void)
|
|
{
|
|
int err;
|
|
snd_pcm_sframes_t delay;
|
|
uint64_t pts;
|
|
|
|
if (!AlsaPCMHandle || !AudioSampleRate) {
|
|
return 0UL;
|
|
}
|
|
// FIXME: thread safe? __assert_fail_base in snd_pcm_delay
|
|
|
|
// delay in frames in alsa + kernel buffers
|
|
if ((err = snd_pcm_delay(AlsaPCMHandle, &delay)) < 0) {
|
|
//Debug(3, "audio/alsa: no hw delay\n");
|
|
delay = 0L;
|
|
} else if (snd_pcm_state(AlsaPCMHandle) != SND_PCM_STATE_RUNNING) {
|
|
//Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay);
|
|
}
|
|
//Debug(3, "audio/alsa: %ld frames hw delay\n", delay);
|
|
|
|
// delay can be negative when underrun occur
|
|
if (delay < 0) {
|
|
delay = 0L;
|
|
}
|
|
|
|
pts = ((uint64_t) delay * 90 * 1000) / AudioSampleRate;
|
|
pts += ((uint64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90 * 1000)
|
|
/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
|
|
Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 " ms\n",
|
|
RingBufferUsedBytes(AlsaRingBuffer), pts / 90);
|
|
|
|
return pts;
|
|
}
|
|
|
|
/**
|
|
** Setup alsa audio for requested format.
|
|
**
|
|
** @param freq sample frequency
|
|
** @param channels number of channels
|
|
** @param use_ac3 use ac3/pass-through device
|
|
**
|
|
** @retval 0 everything ok
|
|
** @retval 1 didn't support frequency/channels combination
|
|
** @retval -1 something gone wrong
|
|
**
|
|
** @todo audio changes must be queued and done when the buffer is empty
|
|
*/
|
|
static int AlsaSetup(int *freq, int *channels, int use_ac3)
|
|
{
|
|
snd_pcm_uframes_t buffer_size;
|
|
snd_pcm_uframes_t period_size;
|
|
int err;
|
|
int ret;
|
|
int delay;
|
|
snd_pcm_t *handle;
|
|
|
|
if (!AlsaPCMHandle) { // alsa not running yet
|
|
return -1;
|
|
}
|
|
#if 1 // easy alsa hw setup way
|
|
// flush any buffered data
|
|
AudioFlushBuffers();
|
|
|
|
if (1) { // close+open to fix hdmi no sound bugs
|
|
handle = AlsaPCMHandle;
|
|
AlsaPCMHandle = NULL;
|
|
snd_pcm_close(handle);
|
|
if (!(handle = AlsaOpenPCM(use_ac3))) {
|
|
return -1;
|
|
}
|
|
AlsaPCMHandle = handle;
|
|
}
|
|
|
|
ret = 0;
|
|
try_again:
|
|
AudioChannels = *channels;
|
|
AudioSampleRate = *freq;
|
|
|
|
if ((err =
|
|
snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16,
|
|
AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED :
|
|
SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1,
|
|
125 * 1000))) {
|
|
Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err));
|
|
|
|
/*
|
|
if ( err == -EBADFD ) {
|
|
snd_pcm_close(AlsaPCMHandle);
|
|
AlsaPCMHandle = NULL;
|
|
goto try_again;
|
|
}
|
|
*/
|
|
|
|
switch (*channels) {
|
|
case 1:
|
|
// FIXME: enable channel upmix
|
|
ret = 1;
|
|
*channels = 2;
|
|
goto try_again;
|
|
case 2:
|
|
return -1;
|
|
case 3:
|
|
case 4:
|
|
case 5:
|
|
case 6:
|
|
case 7:
|
|
case 8:
|
|
// FIXME: enable channel downmix
|
|
// FIXME: try 8 -> 7 -> 6 -> 5 -> 4 -> 3 -> 2
|
|
ret = 1;
|
|
*channels = 2;
|
|
goto try_again;
|
|
default:
|
|
Error(_("audio/alsa: unsupported number of channels\n"));
|
|
// FIXME: must stop sound, AudioChannels ... invalid
|
|
return -1;
|
|
}
|
|
}
|
|
#else
|
|
//
|
|
// complex way to setup parameters
|
|
//
|
|
snd_pcm_hw_params_t *hw_params;
|
|
int dir;
|
|
unsigned buffer_time;
|
|
snd_pcm_uframes_t buffer_size;
|
|
|
|
snd_pcm_hw_params_alloca(&hw_params);
|
|
// choose all parameters
|
|
if ((err = snd_pcm_hw_params_any(AlsaPCMHandle, hw_params)) < 0) {
|
|
Error(_
|
|
("audio: snd_pcm_hw_params_any: no configurations available: %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
|
|
if ((err =
|
|
snd_pcm_hw_params_set_rate_resample(AlsaPCMHandle, hw_params, 1))
|
|
< 0) {
|
|
Error(_("audio: can't set rate resample: %s\n"), snd_strerror(err));
|
|
}
|
|
if ((err =
|
|
snd_pcm_hw_params_set_format(AlsaPCMHandle, hw_params,
|
|
SND_PCM_FORMAT_S16)) < 0) {
|
|
Error(_("audio: can't set 16-bit: %s\n"), snd_strerror(err));
|
|
}
|
|
if ((err =
|
|
snd_pcm_hw_params_set_access(AlsaPCMHandle, hw_params,
|
|
SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
|
|
Error(_("audio: can't set interleaved read/write %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
if ((err =
|
|
snd_pcm_hw_params_set_channels(AlsaPCMHandle, hw_params,
|
|
channels)) < 0) {
|
|
Error(_("audio: can't set channels: %s\n"), snd_strerror(err));
|
|
}
|
|
if ((err =
|
|
snd_pcm_hw_params_set_rate(AlsaPCMHandle, hw_params, freq,
|
|
0)) < 0) {
|
|
Error(_("audio: can't set rate: %s\n"), snd_strerror(err));
|
|
}
|
|
// 500000
|
|
// 170667us
|
|
buffer_time = 1000 * 1000 * 1000;
|
|
dir = 1;
|
|
#if 0
|
|
snd_pcm_hw_params_get_buffer_time_max(hw_params, &buffer_time, &dir);
|
|
Info(_("audio/alsa: %dus max buffer time\n"), buffer_time);
|
|
|
|
buffer_time = 5 * 200 * 1000; // 1s
|
|
if ((err =
|
|
snd_pcm_hw_params_set_buffer_time_near(AlsaPCMHandle, hw_params,
|
|
&buffer_time, &dir)) < 0) {
|
|
Error(_("audio: snd_pcm_hw_params_set_buffer_time_near failed: %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
Info(_("audio/alsa: %dus buffer time\n"), buffer_time);
|
|
#endif
|
|
snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
|
|
Info(_("audio/alsa: buffer size %lu\n"), buffer_size);
|
|
buffer_size = buffer_size < 65536 ? buffer_size : 65536;
|
|
if ((err =
|
|
snd_pcm_hw_params_set_buffer_size_near(AlsaPCMHandle, hw_params,
|
|
&buffer_size))) {
|
|
Error(_("audio: can't set buffer size: %s\n"), snd_strerror(err));
|
|
}
|
|
Info(_("audio/alsa: buffer size %lu\n"), buffer_size);
|
|
|
|
if ((err = snd_pcm_hw_params(AlsaPCMHandle, hw_params)) < 0) {
|
|
Error(_("audio: snd_pcm_hw_params failed: %s\n"), snd_strerror(err));
|
|
}
|
|
// FIXME: use hw_params for buffer_size period_size
|
|
#endif
|
|
|
|
#if 1
|
|
if (0) { // no underruns allowed, play silence
|
|
snd_pcm_sw_params_t *sw_params;
|
|
snd_pcm_uframes_t boundary;
|
|
|
|
snd_pcm_sw_params_alloca(&sw_params);
|
|
err = snd_pcm_sw_params_current(AlsaPCMHandle, sw_params);
|
|
if (err < 0) {
|
|
Error(_("audio: snd_pcm_sw_params_current failed: %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
if ((err = snd_pcm_sw_params_get_boundary(sw_params, &boundary)) < 0) {
|
|
Error(_("audio: snd_pcm_sw_params_get_boundary failed: %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
Debug(4, "audio/alsa: boundary %lu frames\n", boundary);
|
|
if ((err =
|
|
snd_pcm_sw_params_set_stop_threshold(AlsaPCMHandle, sw_params,
|
|
boundary)) < 0) {
|
|
Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
if ((err =
|
|
snd_pcm_sw_params_set_silence_size(AlsaPCMHandle, sw_params,
|
|
boundary)) < 0) {
|
|
Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
if ((err = snd_pcm_sw_params(AlsaPCMHandle, sw_params)) < 0) {
|
|
Error(_("audio: snd_pcm_sw_params failed: %s\n"),
|
|
snd_strerror(err));
|
|
}
|
|
}
|
|
#endif
|
|
|
|
// update buffer
|
|
|
|
snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size);
|
|
Info(_("audio/alsa: buffer size %lu, period size %lu\n"), buffer_size,
|
|
period_size);
|
|
Debug(3, "audio/alsa: state %s\n",
|
|
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
|
|
|
|
AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
|
|
// buffer time/delay in ms
|
|
delay = AudioBufferTime;
|
|
if (VideoAudioDelay > -100) {
|
|
delay += 100 + VideoAudioDelay / 90;
|
|
}
|
|
if (AlsaStartThreshold <
|
|
(*freq * *channels * AudioBytesProSample * delay) / 1000U) {
|
|
AlsaStartThreshold =
|
|
(*freq * *channels * AudioBytesProSample * delay) / 1000U;
|
|
}
|
|
// no bigger, than the buffer
|
|
if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) {
|
|
AlsaStartThreshold = RingBufferFreeBytes(AlsaRingBuffer);
|
|
}
|
|
Info(_("audio/alsa: delay %u ms\n"), (AlsaStartThreshold * 1000)
|
|
/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
** Empty log callback
|
|
*/
|
|
static void AlsaNoopCallback( __attribute__ ((unused))
|
|
const char *file, __attribute__ ((unused))
|
|
int line, __attribute__ ((unused))
|
|
const char *function, __attribute__ ((unused))
|
|
int err, __attribute__ ((unused))
|
|
const char *fmt, ...)
|
|
{
|
|
}
|
|
|
|
/**
|
|
** Initialize alsa audio output module.
|
|
*/
|
|
static void AlsaInit(void)
|
|
{
|
|
#ifndef DEBUG
|
|
// disable display alsa error messages
|
|
snd_lib_error_set_handler(AlsaNoopCallback);
|
|
#else
|
|
(void)AlsaNoopCallback;
|
|
#endif
|
|
AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit
|
|
|
|
AlsaInitPCM();
|
|
AlsaInitMixer();
|
|
}
|
|
|
|
/**
|
|
** Cleanup alsa audio output module.
|
|
*/
|
|
static void AlsaExit(void)
|
|
{
|
|
if (AlsaPCMHandle) {
|
|
snd_pcm_close(AlsaPCMHandle);
|
|
AlsaPCMHandle = NULL;
|
|
}
|
|
if (AlsaMixer) {
|
|
snd_mixer_close(AlsaMixer);
|
|
AlsaMixer = NULL;
|
|
AlsaMixerElem = NULL;
|
|
}
|
|
if (AlsaRingBuffer) {
|
|
RingBufferDel(AlsaRingBuffer);
|
|
AlsaRingBuffer = NULL;
|
|
}
|
|
AlsaFlushBuffer = 0;
|
|
}
|
|
|
|
/**
|
|
** Alsa module.
|
|
*/
|
|
static const AudioModule AlsaModule = {
|
|
.Name = "alsa",
|
|
#ifdef USE_AUDIO_THREAD
|
|
.Thread = AlsaThread,
|
|
.Enqueue = AlsaThreadEnqueue,
|
|
.FlushBuffers = AlsaThreadFlushBuffers,
|
|
#else
|
|
.Enqueue = AlsaEnqueue,
|
|
.FlushBuffers = AlsaFlushBuffers,
|
|
#endif
|
|
.Poller = AlsaPoller,
|
|
.FreeBytes = AlsaFreeBytes,
|
|
.GetDelay = AlsaGetDelay,
|
|
.SetVolume = AlsaSetVolume,
|
|
.Setup = AlsaSetup,
|
|
.Init = AlsaInit,
|
|
.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 RingBuffer *OssRingBuffer; ///< audio ring buffer
|
|
static unsigned OssStartThreshold; ///< start play, if filled
|
|
|
|
#ifdef USE_AUDIO_THREAD
|
|
static volatile char OssFlushBuffer; ///< flag empty buffer
|
|
#endif
|
|
|
|
//----------------------------------------------------------------------------
|
|
// OSS pcm
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Place samples in ringbuffer.
|
|
**
|
|
** @param samples sample buffer
|
|
** @param count number of bytes in sample buffer
|
|
**
|
|
** @returns true if play should be started.
|
|
*/
|
|
static int OssAddToRingbuffer(const void *samples, int count)
|
|
{
|
|
int n;
|
|
|
|
n = RingBufferWrite(OssRingBuffer, samples, count);
|
|
if (n != count) {
|
|
Error(_("audio/oss: can't place %d samples in ring buffer\n"), count);
|
|
// too many bytes are lost
|
|
// FIXME: should skip more, longer skip, but less often?
|
|
}
|
|
// Update audio clock
|
|
AudioPTS +=
|
|
((int64_t) count * 90000) / (AudioSampleRate * AudioChannels *
|
|
AudioBytesProSample);
|
|
|
|
if (!AudioRunning) {
|
|
if (OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
|
|
// restart play-back
|
|
return 1;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
** Play samples from ringbuffer.
|
|
*/
|
|
static int OssPlayRingbuffer(void)
|
|
{
|
|
int first;
|
|
const void *p;
|
|
|
|
first = 1;
|
|
for (;;) {
|
|
audio_buf_info bi;
|
|
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(OssRingBuffer, &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!
|
|
}
|
|
|
|
n = write(OssPcmFildes, p, bi.bytes);
|
|
if (n != bi.bytes) {
|
|
if (n < 0) {
|
|
Error(_("audio/oss: write error: %s\n"), strerror(errno));
|
|
return 1;
|
|
}
|
|
Error(_("audio/oss: error not all bytes written\n"));
|
|
}
|
|
// advance how many could written
|
|
RingBufferReadAdvance(OssRingBuffer, n);
|
|
first = 0;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
** Flush OSS buffers.
|
|
*/
|
|
static void OssFlushBuffers(void)
|
|
{
|
|
if (OssRingBuffer && OssPcmFildes != -1) {
|
|
RingBufferReadAdvance(OssRingBuffer,
|
|
RingBufferUsedBytes(OssRingBuffer));
|
|
// flush kernel buffers
|
|
if (ioctl(OssPcmFildes, SNDCTL_DSP_HALT_OUTPUT, NULL) < 0) {
|
|
Error(_("audio/oss: ioctl(SNDCTL_DSP_HALT_OUTPUT): %s\n"),
|
|
strerror(errno));
|
|
}
|
|
}
|
|
AudioRunning = 0;
|
|
AudioPTS = INT64_C(0x8000000000000000);
|
|
}
|
|
|
|
//----------------------------------------------------------------------------
|
|
// OSS pcm polled
|
|
//----------------------------------------------------------------------------
|
|
|
|
#ifndef USE_AUDIO_THREAD
|
|
|
|
/**
|
|
** Place samples in audio output queue.
|
|
**
|
|
** @param samples sample buffer
|
|
** @param count number of bytes in sample buffer
|
|
*/
|
|
static void OssEnqueue(const void *samples, int count)
|
|
{
|
|
#ifdef DEBUG
|
|
static uint32_t last_tick;
|
|
uint32_t tick;
|
|
|
|
tick = GetMsTicks();
|
|
Debug(4, "audio/oss: %4d %d ms\n", count, tick - last_tick);
|
|
last_tick = tick;
|
|
#endif
|
|
|
|
if (OssPcmFildes == -1) { // setup failure
|
|
Debug(3, "audio/oss: not ready\n");
|
|
return;
|
|
}
|
|
if (OssAddToRingbuffer(samples, count)) {
|
|
AudioRunning = 1;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
/**
|
|
** Play all samples possible, without blocking.
|
|
*/
|
|
static void OssPoller(void)
|
|
{
|
|
if (OssPcmFildes == -1) { // setup failure
|
|
return;
|
|
}
|
|
if (!AudioThread && AudioRunning) {
|
|
OssPlayRingbuffer();
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Get free bytes in audio output.
|
|
*/
|
|
static int OssFreeBytes(void)
|
|
{
|
|
return OssRingBuffer ? RingBufferFreeBytes(OssRingBuffer) : INT32_MAX;
|
|
}
|
|
|
|
#ifdef USE_AUDIO_THREAD
|
|
|
|
//----------------------------------------------------------------------------
|
|
// thread playback
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** OSS thread
|
|
*/
|
|
static void OssThread(void)
|
|
{
|
|
for (;;) {
|
|
struct pollfd fds[1];
|
|
int err;
|
|
|
|
pthread_testcancel();
|
|
if (OssFlushBuffer) {
|
|
// we can flush too many, but wo cares
|
|
Debug(3, "audio/oss: flushing buffers\n");
|
|
OssFlushBuffers();
|
|
OssFlushBuffer = 0;
|
|
break;
|
|
}
|
|
|
|
fds[0].fd = OssPcmFildes;
|
|
fds[0].events = POLLOUT | POLLERR;
|
|
// wait for space in kernel buffers
|
|
err = poll(fds, 1, 100);
|
|
if (err < 0) {
|
|
Error(_("audio/oss: error poll %s\n"), strerror(errno));
|
|
usleep(100 * 1000);
|
|
continue;
|
|
}
|
|
|
|
if (OssFlushBuffer) {
|
|
continue;
|
|
}
|
|
|
|
if ((err = OssPlayRingbuffer())) { // empty / error
|
|
if (err < 0) { // underrun error
|
|
break;
|
|
}
|
|
pthread_yield();
|
|
usleep(20 * 1000); // let fill/empty the buffers
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Place samples in audio output queue.
|
|
**
|
|
** @param samples sample buffer
|
|
** @param count number of bytes in sample buffer
|
|
*/
|
|
static void OssThreadEnqueue(const void *samples, int count)
|
|
{
|
|
if (!OssRingBuffer || OssPcmFildes == -1 || !AudioSampleRate) {
|
|
Debug(3, "audio/oss: enqueue not ready\n");
|
|
return;
|
|
}
|
|
if (OssAddToRingbuffer(samples, count)) {
|
|
// no lock needed, can wakeup next time
|
|
AudioRunning = 1;
|
|
pthread_cond_signal(&AudioStartCond);
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Flush OSS buffers with thread.
|
|
*/
|
|
static void OssThreadFlushBuffers(void)
|
|
{
|
|
// signal thread to flush buffers
|
|
if (AudioThread) {
|
|
OssFlushBuffer = 1;
|
|
do {
|
|
AudioRunning = 1; // wakeup in case of sleeping
|
|
pthread_cond_signal(&AudioStartCond);
|
|
usleep(1 * 1000);
|
|
} while (OssFlushBuffer); // wait until flushed
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Open OSS pcm device.
|
|
**
|
|
** @param use_ac3 use ac3/pass-through device
|
|
*/
|
|
static int OssOpenPCM(int use_ac3)
|
|
{
|
|
const char *device;
|
|
int fildes;
|
|
|
|
// &&|| hell
|
|
if (!(use_ac3 && ((device = AudioAC3Device)
|
|
|| (device = getenv("OSS_AC3_AUDIODEV"))))
|
|
&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
|
|
device = "/dev/dsp";
|
|
}
|
|
Debug(3, "audio/oss: &&|| hell '%s'\n", 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 (0-100)
|
|
**
|
|
** @param volume volume (0 .. 100)
|
|
*/
|
|
static void OssSetVolume(int volume)
|
|
{
|
|
int v;
|
|
|
|
if (OssMixerFildes != -1) {
|
|
v = (volume * 255) / 100;
|
|
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 uint64_t OssGetDelay(void)
|
|
{
|
|
int delay;
|
|
uint64_t pts;
|
|
|
|
if (OssPcmFildes == -1) { // setup failure
|
|
return 0UL;
|
|
}
|
|
|
|
if (!AudioRunning) {
|
|
return 0UL;
|
|
}
|
|
// 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 0UL;
|
|
}
|
|
if (delay == -1) {
|
|
delay = 0UL;
|
|
}
|
|
|
|
pts = ((uint64_t) delay * 90 * 1000)
|
|
/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
|
|
pts += ((uint64_t) RingBufferUsedBytes(OssRingBuffer) * 90 * 1000)
|
|
/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
|
|
if (pts > 600 * 90) {
|
|
Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 " ms\n",
|
|
RingBufferUsedBytes(OssRingBuffer), pts / 90);
|
|
}
|
|
|
|
return pts;
|
|
}
|
|
|
|
/**
|
|
** Setup OSS audio for requested format.
|
|
**
|
|
** @param freq sample frequency
|
|
** @param channels number of channels
|
|
** @param use_ac3 use ac3/pass-through device
|
|
**
|
|
** @retval 0 everything ok
|
|
** @retval 1 didn't support frequency/channels combination
|
|
** @retval -1 something gone wrong
|
|
**
|
|
** @todo audio changes must be queued and done when the buffer is empty
|
|
*/
|
|
static int OssSetup(int *freq, int *channels, int use_ac3)
|
|
{
|
|
int ret;
|
|
int tmp;
|
|
int delay;
|
|
|
|
if (OssPcmFildes == -1) { // OSS not ready
|
|
return -1;
|
|
}
|
|
// flush any buffered data
|
|
AudioFlushBuffers();
|
|
|
|
if (1) { // close+open for pcm / ac3
|
|
int fildes;
|
|
|
|
fildes = OssPcmFildes;
|
|
OssPcmFildes = -1;
|
|
close(fildes);
|
|
if (!(fildes = OssOpenPCM(use_ac3))) {
|
|
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 = *freq;
|
|
if (ioctl(OssPcmFildes, SNDCTL_DSP_SPEED, &tmp) == -1) {
|
|
Error(_("audio/oss: ioctl(SNDCTL_DSP_SPEED): %s\n"), strerror(errno));
|
|
return -1;
|
|
}
|
|
if (tmp != *freq) {
|
|
Warning(_("audio/oss: device doesn't support %d Hz sample rate.\n"),
|
|
*freq);
|
|
*freq = tmp;
|
|
ret = 1;
|
|
}
|
|
|
|
AudioChannels = *channels;
|
|
AudioSampleRate = *freq;
|
|
|
|
// FIXME: setup buffers
|
|
|
|
if (1) {
|
|
audio_buf_info bi;
|
|
|
|
if (ioctl(OssPcmFildes, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
|
|
Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"),
|
|
strerror(errno));
|
|
} else {
|
|
Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes);
|
|
}
|
|
|
|
tmp = -1;
|
|
if (ioctl(OssPcmFildes, SNDCTL_DSP_GETODELAY, &tmp) == -1) {
|
|
Error(_("audio/oss: ioctl(SNDCTL_DSP_GETODELAY): %s\n"),
|
|
strerror(errno));
|
|
// FIXME: stop player, set setup failed flag
|
|
return -1;
|
|
}
|
|
if (tmp == -1) {
|
|
tmp = 0;
|
|
}
|
|
// start when enough bytes for initial write
|
|
OssStartThreshold = bi.bytes + tmp;
|
|
// buffer time/delay in ms
|
|
delay = AudioBufferTime;
|
|
if (VideoAudioDelay > -100) {
|
|
delay += 100 + VideoAudioDelay / 90;
|
|
}
|
|
if (OssStartThreshold <
|
|
(*freq * *channels * AudioBytesProSample * delay) / 1000U) {
|
|
OssStartThreshold =
|
|
(*freq * *channels * AudioBytesProSample * delay) / 1000U;
|
|
}
|
|
// no bigger, than the buffer
|
|
if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) {
|
|
OssStartThreshold = RingBufferFreeBytes(OssRingBuffer);
|
|
}
|
|
|
|
Info(_("audio/oss: delay %u ms\n"), (OssStartThreshold * 1000)
|
|
/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
** Initialize OSS audio output module.
|
|
*/
|
|
static void OssInit(void)
|
|
{
|
|
OssRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit
|
|
|
|
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;
|
|
}
|
|
OssFlushBuffer = 0;
|
|
}
|
|
|
|
/**
|
|
** OSS module.
|
|
*/
|
|
static const AudioModule OssModule = {
|
|
.Name = "oss",
|
|
#ifdef USE_AUDIO_THREAD
|
|
.Thread = OssThread,
|
|
.Enqueue = OssThreadEnqueue,
|
|
.FlushBuffers = OssThreadFlushBuffers,
|
|
#else
|
|
.Enqueue = OssEnqueue,
|
|
.FlushBuffers = OssFlushBuffers,
|
|
#endif
|
|
.Poller = OssPoller,
|
|
.FreeBytes = OssFreeBytes,
|
|
.GetDelay = OssGetDelay,
|
|
.SetVolume = OssSetVolume,
|
|
.Setup = OssSetup,
|
|
.Init = OssInit,
|
|
.Exit = OssExit,
|
|
};
|
|
|
|
#endif // USE_OSS
|
|
|
|
//============================================================================
|
|
// Noop
|
|
//============================================================================
|
|
|
|
/**
|
|
** Noop enqueue samples.
|
|
**
|
|
** @param samples sample buffer
|
|
** @param count number of bytes in sample buffer
|
|
*/
|
|
static void NoopEnqueue( __attribute__ ((unused))
|
|
const void *samples, __attribute__ ((unused))
|
|
int count)
|
|
{
|
|
}
|
|
|
|
/**
|
|
** Get free bytes in audio output.
|
|
*/
|
|
static int NoopFreeBytes(void)
|
|
{
|
|
return INT32_MAX; // no driver, much space
|
|
}
|
|
|
|
/**
|
|
** Get audio delay in time stamps.
|
|
**
|
|
** @returns audio delay in time stamps.
|
|
*/
|
|
static uint64_t NoopGetDelay(void)
|
|
{
|
|
return 0UL;
|
|
}
|
|
|
|
/**
|
|
** Set mixer volume (0-100)
|
|
**
|
|
** @param volume volume (0 .. 100)
|
|
*/
|
|
static void NoopSetVolume( __attribute__ ((unused))
|
|
int volume)
|
|
{
|
|
}
|
|
|
|
/**
|
|
** Noop setup.
|
|
**
|
|
** @param freq sample frequency
|
|
** @param channels number of channels
|
|
*/
|
|
static int NoopSetup( __attribute__ ((unused))
|
|
int *channels, __attribute__ ((unused))
|
|
int *freq, __attribute__ ((unused))
|
|
int use_ac3)
|
|
{
|
|
return -1;
|
|
}
|
|
|
|
/**
|
|
** Noop void
|
|
*/
|
|
static void NoopVoid(void)
|
|
{
|
|
}
|
|
|
|
/**
|
|
** Noop module.
|
|
*/
|
|
static const AudioModule NoopModule = {
|
|
.Name = "noop",
|
|
.Enqueue = NoopEnqueue,
|
|
.FlushBuffers = NoopVoid,
|
|
.Poller = NoopVoid,
|
|
.FreeBytes = NoopFreeBytes,
|
|
.GetDelay = NoopGetDelay,
|
|
.SetVolume = NoopSetVolume,
|
|
.Setup = NoopSetup,
|
|
.Init = NoopVoid,
|
|
.Exit = NoopVoid,
|
|
};
|
|
|
|
//----------------------------------------------------------------------------
|
|
// thread playback
|
|
//----------------------------------------------------------------------------
|
|
|
|
#ifdef USE_AUDIO_THREAD
|
|
|
|
/**
|
|
** Audio play thread.
|
|
*/
|
|
static void *AudioPlayHandlerThread(void *dummy)
|
|
{
|
|
Debug(3, "audio: play thread started\n");
|
|
for (;;) {
|
|
Debug(3, "audio: wait on start condition\n");
|
|
pthread_mutex_lock(&AudioMutex);
|
|
AudioRunning = 0;
|
|
do {
|
|
pthread_cond_wait(&AudioStartCond, &AudioMutex);
|
|
// cond_wait can return, without signal!
|
|
} while (!AudioRunning);
|
|
pthread_mutex_unlock(&AudioMutex);
|
|
|
|
#ifdef USE_AUDIORING
|
|
if (atomic_read(&AudioRingFilled) > 1) {
|
|
int sample_rate;
|
|
int channels;
|
|
|
|
// skip all sample changes between
|
|
while (atomic_read(&AudioRingFilled) > 1) {
|
|
Debug(3, "audio: skip ring buffer\n");
|
|
AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX;
|
|
atomic_dec(&AudioRingFilled);
|
|
}
|
|
|
|
#ifdef USE_ALSA
|
|
// FIXME: flush only if there is something to flush
|
|
AlsaFlushBuffers();
|
|
|
|
sample_rate = AudioRing[AudioRingRead].SampleRate;
|
|
channels = AudioRing[AudioRingRead].Channels;
|
|
Debug(3, "audio: thread channels %d sample-rate %d hz\n", channels,
|
|
sample_rate);
|
|
|
|
if (AlsaSetup(&sample_rate, &channels)) {
|
|
Error(_("audio: can't set channels %d sample-rate %d hz\n"),
|
|
channels, sample_rate);
|
|
}
|
|
Debug(3, "audio: thread channels %d sample-rate %d hz\n",
|
|
AudioChannels, AudioSampleRate);
|
|
#endif
|
|
}
|
|
#endif
|
|
|
|
Debug(3, "audio: play start\n");
|
|
AudioUsedModule->Thread();
|
|
}
|
|
|
|
return dummy;
|
|
}
|
|
|
|
/**
|
|
** Initialize audio thread.
|
|
*/
|
|
static void AudioInitThread(void)
|
|
{
|
|
pthread_mutex_init(&AudioMutex, NULL);
|
|
pthread_cond_init(&AudioStartCond, NULL);
|
|
pthread_create(&AudioThread, NULL, AudioPlayHandlerThread, NULL);
|
|
pthread_setname_np(AudioThread, "softhddev audio");
|
|
|
|
pthread_yield();
|
|
usleep(5 * 1000); // give thread some time to start
|
|
}
|
|
|
|
/**
|
|
** Cleanup audio thread.
|
|
*/
|
|
static void AudioExitThread(void)
|
|
{
|
|
void *retval;
|
|
|
|
if (AudioThread) {
|
|
if (pthread_cancel(AudioThread)) {
|
|
Error(_("audio: can't queue cancel play thread\n"));
|
|
}
|
|
if (pthread_join(AudioThread, &retval) || retval != PTHREAD_CANCELED) {
|
|
Error(_("audio: can't cancel play thread\n"));
|
|
}
|
|
pthread_cond_destroy(&AudioStartCond);
|
|
pthread_mutex_destroy(&AudioMutex);
|
|
AudioThread = 0;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
//----------------------------------------------------------------------------
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Table of all audio modules.
|
|
*/
|
|
static const AudioModule *AudioModules[] = {
|
|
#ifdef USE_ALSA
|
|
&AlsaModule,
|
|
#endif
|
|
#ifdef USE_OSS
|
|
&OssModule,
|
|
#endif
|
|
&NoopModule,
|
|
};
|
|
|
|
/**
|
|
** Place samples in audio output queue.
|
|
**
|
|
** @param samples sample buffer
|
|
** @param count number of bytes in sample buffer
|
|
*/
|
|
void AudioEnqueue(const void *samples, int count)
|
|
{
|
|
AudioUsedModule->Enqueue(samples, count);
|
|
}
|
|
|
|
/**
|
|
** Flush audio buffers.
|
|
*/
|
|
void AudioFlushBuffers(void)
|
|
{
|
|
AudioUsedModule->FlushBuffers();
|
|
}
|
|
|
|
/**
|
|
** Call back to play audio polled.
|
|
*/
|
|
void AudioPoller(void)
|
|
{
|
|
AudioUsedModule->Poller();
|
|
}
|
|
|
|
/**
|
|
** Get free bytes in audio output.
|
|
*/
|
|
int AudioFreeBytes(void)
|
|
{
|
|
return AudioUsedModule->FreeBytes();
|
|
}
|
|
|
|
/**
|
|
** Get audio delay in time stamps.
|
|
**
|
|
** @returns audio delay in time stamps.
|
|
*/
|
|
uint64_t AudioGetDelay(void)
|
|
{
|
|
return AudioUsedModule->GetDelay();
|
|
}
|
|
|
|
/**
|
|
** Set audio clock base.
|
|
**
|
|
** @param pts audio presentation timestamp
|
|
*/
|
|
void AudioSetClock(int64_t pts)
|
|
{
|
|
#ifdef DEBUG
|
|
if (AudioPTS != pts) {
|
|
Debug(4, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n",
|
|
AudioPTS, pts);
|
|
|
|
}
|
|
#endif
|
|
AudioPTS = pts;
|
|
}
|
|
|
|
/**
|
|
** Get current audio clock.
|
|
**
|
|
** @returns the audio clock in time stamps.
|
|
*/
|
|
int64_t AudioGetClock(void)
|
|
{
|
|
if ((uint64_t) AudioPTS != INT64_C(0x8000000000000000)) {
|
|
int64_t delay;
|
|
|
|
if ((delay = AudioGetDelay())) {
|
|
return AudioPTS - delay;
|
|
}
|
|
}
|
|
return INT64_C(0x8000000000000000);
|
|
}
|
|
|
|
/**
|
|
** Set mixer volume (0-100)
|
|
**
|
|
** @param volume volume (0 .. 100)
|
|
*/
|
|
void AudioSetVolume(int volume)
|
|
{
|
|
#ifdef USE_ALSA
|
|
AlsaSetVolume(volume);
|
|
#endif
|
|
#ifdef USE_OSS
|
|
OssSetVolume(volume);
|
|
#endif
|
|
(void)volume;
|
|
}
|
|
|
|
/**
|
|
** Setup audio for requested format.
|
|
**
|
|
** @param freq sample frequency
|
|
** @param channels number of channels
|
|
** @param use_ac3 use ac3/pass-through device
|
|
**
|
|
** @retval 0 everything ok
|
|
** @retval 1 didn't support frequency/channels combination
|
|
** @retval -1 something gone wrong
|
|
**
|
|
** @todo audio changes must be queued and done when the buffer is empty
|
|
*/
|
|
int AudioSetup(int *freq, int *channels, int use_ac3)
|
|
{
|
|
Debug(3, "audio: channels %d frequency %d hz %s\n", *channels, *freq,
|
|
use_ac3 ? "ac3" : "pcm");
|
|
|
|
// invalid parameter
|
|
if (!freq || !channels || !*freq || !*channels) {
|
|
Debug(3, "audio: bad channels or frequency parameters\n");
|
|
// FIXME: set flag invalid setup
|
|
return -1;
|
|
}
|
|
#ifdef USE_AUDIORING
|
|
// FIXME: need to store possible combination and report this
|
|
return AudioRingAdd(*freq, *channels, use_ac3);
|
|
#endif
|
|
return AudioUsedModule->Setup(freq, channels, use_ac3);
|
|
}
|
|
|
|
/**
|
|
** Set pcm audio device.
|
|
**
|
|
** @param device name of pcm device (fe. "hw:0,9" or "/dev/dsp")
|
|
**
|
|
** @note this is currently used to select alsa/OSS output module.
|
|
*/
|
|
void AudioSetDevice(const char *device)
|
|
{
|
|
if (!AudioModuleName) {
|
|
AudioModuleName = "alsa"; // detect alsa/OSS
|
|
if (!device[0]) {
|
|
AudioModuleName = "noop";
|
|
} else if (device[0] == '/') {
|
|
AudioModuleName = "oss";
|
|
}
|
|
}
|
|
AudioPCMDevice = device;
|
|
}
|
|
|
|
/**
|
|
** Set pass-through audio device.
|
|
**
|
|
** @param device name of pass-through device (fe. "hw:0,1")
|
|
**
|
|
** @note this is currently usable with alsa only.
|
|
*/
|
|
void AudioSetDeviceAC3(const char *device)
|
|
{
|
|
if (!AudioModuleName) {
|
|
AudioModuleName = "alsa"; // detect alsa/OSS
|
|
if (!device[0]) {
|
|
AudioModuleName = "noop";
|
|
} else if (device[0] == '/') {
|
|
AudioModuleName = "oss";
|
|
}
|
|
}
|
|
AudioAC3Device = device;
|
|
}
|
|
|
|
/**
|
|
** Set pcm audio mixer channel.
|
|
**
|
|
** @param channel name of the mixer channel (fe. PCM or Master)
|
|
**
|
|
** @note this is currently used to select alsa/OSS output module.
|
|
*/
|
|
void AudioSetChannel(const char *channel)
|
|
{
|
|
AudioMixerChannel = channel;
|
|
}
|
|
|
|
/**
|
|
** Initialize audio output module.
|
|
**
|
|
** @todo FIXME: make audio output module selectable.
|
|
*/
|
|
void AudioInit(void)
|
|
{
|
|
int freq;
|
|
int chan;
|
|
unsigned u;
|
|
const char *name;
|
|
|
|
name = "noop";
|
|
#ifdef USE_OSS
|
|
name = "oss";
|
|
#endif
|
|
#ifdef USE_ALSA
|
|
name = "alsa";
|
|
#endif
|
|
if (AudioModuleName) {
|
|
name = AudioModuleName;
|
|
}
|
|
//
|
|
// search selected audio module.
|
|
//
|
|
for (u = 0; u < sizeof(AudioModules) / sizeof(*AudioModules); ++u) {
|
|
if (!strcasecmp(name, AudioModules[u]->Name)) {
|
|
AudioUsedModule = AudioModules[u];
|
|
Info(_("audio: '%s' output module used\n"), AudioUsedModule->Name);
|
|
goto found;
|
|
}
|
|
}
|
|
Error(_("audio: '%s' output module isn't supported\n"), name);
|
|
AudioUsedModule = &NoopModule;
|
|
return;
|
|
|
|
found:
|
|
#ifdef USE_AUDIORING
|
|
AudioRingInit();
|
|
#endif
|
|
AudioUsedModule->Init();
|
|
freq = 48000;
|
|
chan = 2;
|
|
if (AudioSetup(&freq, &chan, 0)) { // set default parameters
|
|
Error(_("audio: can't do initial setup\n"));
|
|
}
|
|
#ifdef USE_AUDIO_THREAD
|
|
if (AudioUsedModule->Thread) { // supports threads
|
|
AudioInitThread();
|
|
}
|
|
#endif
|
|
|
|
AudioPaused = 1;
|
|
}
|
|
|
|
/**
|
|
** Cleanup audio output module.
|
|
*/
|
|
void AudioExit(void)
|
|
{
|
|
#ifdef USE_AUDIO_THREAD
|
|
AudioExitThread();
|
|
#endif
|
|
AudioUsedModule->Exit();
|
|
AudioUsedModule = &NoopModule;
|
|
#ifdef USE_AUDIORING
|
|
AudioRingExit();
|
|
#endif
|
|
AudioRunning = 0;
|
|
}
|
|
|
|
#ifdef AUDIO_TEST
|
|
|
|
//----------------------------------------------------------------------------
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// Test
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//----------------------------------------------------------------------------
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void AudioTest(void)
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{
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for (;;) {
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unsigned u;
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uint8_t buffer[16 * 1024]; // some random data
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int i;
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for (u = 0; u < sizeof(buffer); u++) {
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buffer[u] = random() & 0xffff;
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}
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Debug(3, "audio/test: loop\n");
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for (i = 0; i < 100; ++i) {
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while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) {
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AlsaEnqueue(buffer, sizeof(buffer));
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}
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usleep(20 * 1000);
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}
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break;
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}
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}
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#include <getopt.h>
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int SysLogLevel; ///< show additional debug informations
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/**
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** Print version.
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*/
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static void PrintVersion(void)
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{
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printf("audio_test: audio tester Version " VERSION
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#ifdef GIT_REV
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"(GIT-" GIT_REV ")"
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#endif
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",\n\t(c) 2009 - 2012 by Johns\n"
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"\tLicense AGPLv3: GNU Affero General Public License version 3\n");
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}
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/**
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** Print usage.
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*/
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static void PrintUsage(void)
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{
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printf("Usage: audio_test [-?dhv]\n"
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"\t-d\tenable debug, more -d increase the verbosity\n"
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"\t-? -h\tdisplay this message\n" "\t-v\tdisplay version information\n"
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"Only idiots print usage on stderr!\n");
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}
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/**
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** Main entry point.
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**
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** @param argc number of arguments
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** @param argv arguments vector
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**
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** @returns -1 on failures, 0 clean exit.
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*/
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int main(int argc, char *const argv[])
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{
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SysLogLevel = 0;
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//
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// Parse command line arguments
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//
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for (;;) {
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switch (getopt(argc, argv, "hv?-c:d")) {
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case 'd': // enabled debug
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++SysLogLevel;
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continue;
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case EOF:
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break;
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case 'v': // print version
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PrintVersion();
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return 0;
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case '?':
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case 'h': // help usage
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PrintVersion();
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PrintUsage();
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return 0;
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case '-':
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PrintVersion();
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PrintUsage();
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fprintf(stderr, "\nWe need no long options\n");
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return -1;
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case ':':
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PrintVersion();
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fprintf(stderr, "Missing argument for option '%c'\n", optopt);
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return -1;
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default:
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PrintVersion();
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fprintf(stderr, "Unkown option '%c'\n", optopt);
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return -1;
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}
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break;
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}
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if (optind < argc) {
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PrintVersion();
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while (optind < argc) {
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fprintf(stderr, "Unhandled argument '%s'\n", argv[optind++]);
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}
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return -1;
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}
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//
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// main loop
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//
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AudioInit();
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for (;;) {
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unsigned u;
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uint8_t buffer[16 * 1024]; // some random data
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for (u = 0; u < sizeof(buffer); u++) {
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buffer[u] = random() & 0xffff;
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}
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Debug(3, "audio/test: loop\n");
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for (;;) {
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while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) {
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AlsaEnqueue(buffer, sizeof(buffer));
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}
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}
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}
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AudioExit();
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return 0;
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}
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#endif
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