///
/// @file audio.c @brief Audio module
///
/// Copyright (c) 2009 - 2012 by Johns. All Rights Reserved.
///
/// Contributor(s):
///
/// License: AGPLv3
///
/// This program is free software: you can redistribute it and/or modify
/// it under the terms of the GNU Affero General Public License as
/// published by the Free Software Foundation, either version 3 of the
/// License.
///
/// This program is distributed in the hope that it will be useful,
/// but WITHOUT ANY WARRANTY; without even the implied warranty of
/// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
/// GNU Affero General Public License for more details.
///
/// $Id$
//////////////////////////////////////////////////////////////////////////////
///
/// @defgroup Audio The audio module.
///
/// This module contains all audio output functions.
///
/// ALSA PCM/Mixer api is supported.
/// @see http://www.alsa-project.org/alsa-doc/alsa-lib
///
/// @note alsa async playback is broken, don't use it!
///
/// OSS PCM/Mixer api is supported.
/// @see http://manuals.opensound.com/developer/
///
///
/// @todo FIXME: there can be problems with little/big endian.
/// @todo FIXME: can combine oss and alsa ring buffer
///
#ifdef USE_ALSA // only with alsa supported
#define USE_AUDIO_THREAD ///< use thread for audio playback
#endif
//#define USE_ALSA ///< enable alsa support
//#define USE_OSS ///< enable oss support
#define noSEARCH_HDMI_BUG
#define noSEARCH_HDMI_BUG2
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <inttypes.h>
#include <libintl.h>
#define _(str) gettext(str) ///< gettext shortcut
#define _N(str) str ///< gettext_noop shortcut
#ifdef USE_ALSA
#include <alsa/asoundlib.h>
#endif
#ifdef USE_OSS
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
// SNDCTL_DSP_HALT_OUTPUT compatibility
#ifndef SNDCTL_DSP_HALT_OUTPUT
# if defined(SNDCTL_DSP_RESET_OUTPUT)
# define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET_OUTPUT
# elif defined(SNDCTL_DSP_RESET)
# define SNDCTL_DSP_HALT_OUTPUT SNDCTL_DSP_RESET
# else
# error "No valid SNDCTL_DSP_HALT_OUTPUT found."
# endif
#endif
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <string.h>
#endif
#ifdef USE_AUDIO_THREAD
#ifndef __USE_GNU
#define __USE_GNU
#endif
#include <pthread.h>
#ifndef HAVE_PTHREAD_NAME
/// only available with newer glibc
#define pthread_setname_np(thread, name)
#endif
#endif
#include <alsa/iatomic.h> // portable atomic_t
#include "ringbuffer.h"
#include "misc.h"
#include "audio.h"
//----------------------------------------------------------------------------
// Variables
//----------------------------------------------------------------------------
static const char *AudioPCMDevice; ///< alsa/oss PCM device name
static const char *AudioMixerDevice; ///< alsa/oss mixer device name
static const char *AudioMixerChannel; ///< alsa/oss mixer channel name
static volatile char AudioRunning; ///< thread running / stopped
static int AudioPaused; ///< audio paused
static unsigned AudioSampleRate; ///< audio sample rate in hz
static unsigned AudioChannels; ///< number of audio channels
static const int AudioBytesProSample = 2; ///< number of bytes per sample
static int64_t AudioPTS; ///< audio pts clock
#ifdef USE_AUDIO_THREAD
static pthread_cond_t AudioStartCond; ///< condition variable
#endif
#ifdef SEARCH_HDMI_BUG2
//----------------------------------------------------------------------------
// ring buffer
//----------------------------------------------------------------------------
// FIXME: use this code, to combine alsa&oss ring buffers
#define AUDIO_RING_MAX 8 ///< number of audio ring buffers
/**
** Audio ring buffer.
*/
typedef struct _audio_ring_ring_
{
char FlushBuffers; ///< flag: flush buffers
unsigned SampleRate; ///< sample rate in hz
unsigned Channels; ///< number of channels
} AudioRingRing;
/// ring of audio ring buffers
static AudioRingRing AudioRing[AUDIO_RING_MAX];
static int AudioRingWrite; ///< audio ring write pointer
static int AudioRingRead; ///< audio ring read pointer
static atomic_t AudioRingFilled; ///< how many of the ring is used
/**
** Add sample rate, number of channel change to ring.
**
** @param freq sample frequency
** @param channels number of channels
*/
static int AudioRingAdd(int freq, int channels)
{
int filled;
filled = atomic_read(&AudioRingFilled);
if (filled == AUDIO_RING_MAX) { // no free slot
// FIXME: can wait for ring buffer empty
Error(_("audio: out of ring buffers\n"));
return -1;
}
AudioRing[AudioRingWrite].FlushBuffers = 1;
AudioRing[AudioRingWrite].SampleRate = freq;
AudioRing[AudioRingWrite].Channels = channels;
AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
atomic_inc(&AudioRingFilled);
#ifdef USE_AUDIO_THREAD
// tell thread, that something todo
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
#endif
return 0;
}
/**
** Setup audio ring.
*/
static void AudioRingInit(void)
{
int i;
for (i = 0; i < AUDIO_RING_MAX; ++i) {
// FIXME:
//AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit
}
// one slot always reservered
AudioRingWrite = 1;
atomic_set(&AudioRingFilled, 1);
}
/**
** Cleanup audio ring.
*/
static void AudioRingExit(void)
{
int i;
for (i = 0; i < AUDIO_RING_MAX; ++i) {
// FIXME:
//RingBufferDel(AlsaRingBuffer);
}
}
#endif
#ifdef USE_ALSA
//============================================================================
// A L S A
//============================================================================
//----------------------------------------------------------------------------
// Alsa variables
//----------------------------------------------------------------------------
static snd_pcm_t *AlsaPCMHandle; ///< alsa pcm handle
static char AlsaCanPause; ///< hw supports pause
static int AlsaUseMmap; ///< use mmap
static RingBuffer *AlsaRingBuffer; ///< audio ring buffer
static unsigned AlsaStartThreshold; ///< start play, if filled
static volatile char AlsaFlushBuffer; ///< flag empty buffer
static snd_mixer_t *AlsaMixer; ///< alsa mixer handle
static snd_mixer_elem_t *AlsaMixerElem; ///< alsa pcm mixer element
static int AlsaRatio; ///< internal -> mixer ratio * 1000
//----------------------------------------------------------------------------
// alsa 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 AlsaAddToRingbuffer(const void *samples, int count)
{
int n;
n = RingBufferWrite(AlsaRingBuffer, samples, count);
if (n != count) {
Error(_("audio/alsa: 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 (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
// restart play-back
return 1;
}
}
return 0;
}
/**
** Play samples from ringbuffer.
*/
static int AlsaPlayRingbuffer(void)
{
int first;
int avail;
int n;
int err;
int frames;
const void *p;
first = 1;
for (;;) {
// how many bytes can be written?
n = snd_pcm_avail_update(AlsaPCMHandle);
if (n < 0) {
if (n == -EAGAIN) {
continue;
}
Error(_("audio/alsa: underrun error?\n"));
err = snd_pcm_recover(AlsaPCMHandle, n, 0);
if (err >= 0) {
continue;
}
Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"),
snd_strerror(n));
return -1;
}
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n);
if (avail < 256) { // too much overhead
if (first) {
// happens with broken alsa drivers
Error(_("audio/alsa: broken driver %d\n"), avail);
usleep(5 * 1000);
}
Debug(4, "audio/alsa: break state %s\n",
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
break;
}
#ifdef SEARCH_HDMI_BUG
{
uint16_t buf[8192];
unsigned u;
for (u = 0; u < sizeof(buf) / 2; u++) {
buf[u] = random() & 0xffff;
}
n = sizeof(buf);
p = buf;
}
#else
n = RingBufferGetReadPointer(AlsaRingBuffer, &p);
if (!n) { // ring buffer empty
if (first) { // only error on first loop
return 1;
}
return 0;
}
#endif
if (n < avail) { // not enough bytes in ring buffer
avail = n;
}
if (!avail) { // full or buffer empty
break;
}
frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail);
again:
if (AlsaUseMmap) {
err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames);
} else {
err = snd_pcm_writei(AlsaPCMHandle, p, frames);
}
//Debug(3, "audio/alsa: wrote %d/%d frames\n", err, frames);
if (err != frames) {
if (err < 0) {
if (err == -EAGAIN) {
goto again;
}
/*
if (err == -EBADFD) {
goto again;
}
*/
Error(_("audio/alsa: underrun error?\n"));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
goto again;
}
Error(_("audio/alsa: snd_pcm_writei failed: %s\n"),
snd_strerror(err));
return -1;
}
// this could happen, if underrun happened
Error(_("audio/alsa: error not all frames written\n"));
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err);
}
RingBufferReadAdvance(AlsaRingBuffer, avail);
first = 0;
}
return 0;
}
/**
** Flush alsa buffers.
*/
static void AlsaFlushBuffers(void)
{
int err;
snd_pcm_state_t state;
RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer));
state = snd_pcm_state(AlsaPCMHandle);
Debug(3, "audio/alsa: state %d - %s\n", state, snd_pcm_state_name(state));
if (state != SND_PCM_STATE_OPEN) {
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
}
// ****ing alsa crash, when in open state here
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err));
}
}
AudioPTS = INT64_C(0x8000000000000000);
}
#if 0
//----------------------------------------------------------------------------
// async playback
//----------------------------------------------------------------------------
// async playback is broken, don't use it!
/**
** Alsa async pcm callback function.
**
** @param handler alsa async handler
*/
static void AlsaAsyncCallback(snd_async_handler_t * handler)
{
Debug(3, "audio/%s: %p\n", __FUNCTION__, handler);
// how many bytes can be written?
for (;;) {
n = snd_pcm_avail_update(AlsaPCMHandle);
if (n < 0) {
Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"),
snd_strerror(n));
break;
}
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n);
if (avail < 512) { // too much overhead
break;
}
n = RingBufferGetReadPointer(AlsaRingBuffer, &p);
if (!n) { // ring buffer empty
Debug(3, "audio/alsa: ring buffer empty\n");
break;
}
if (n < avail) { // not enough bytes in ring buffer
avail = n;
}
if (!avail) { // full
break;
}
frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail);
again:
if (AlsaUseMmap) {
err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames);
} else {
err = snd_pcm_writei(AlsaPCMHandle, p, frames);
}
Debug(3, "audio/alsa: %d => %d\n", frames, err);
if (err < 0) {
Error(_("audio/alsa: underrun error?\n"));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
goto again;
}
Error(_("audio/alsa: snd_pcm_writei failed: %s\n"),
snd_strerror(err));
}
if (err != frames) {
Error(_("audio/alsa: error not all frames written\n"));
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err);
}
RingBufferReadAdvance(AlsaRingBuffer, avail);
}
}
/**
** 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)
{
snd_pcm_state_t state;
int n;
//int err;
Debug(3, "audio: %6zd + %4d\n", RingBufferUsedBytes(AlsaRingBuffer),
count);
n = RingBufferWrite(AlsaRingBuffer, samples, count);
if (n != count) {
Fatal(_("audio: can't place %d samples in ring buffer\n"), count);
}
// check if running, wait until enough buffered
state = snd_pcm_state(AlsaPCMHandle);
if (state == SND_PCM_STATE_PREPARED) {
Debug(3, "audio/alsa: state %d - %s\n", state,
snd_pcm_state_name(state));
// FIXME: adjust start ratio
if (RingBufferFreeBytes(AlsaRingBuffer)
< RingBufferUsedBytes(AlsaRingBuffer)) {
// restart play-back
#if 0
if (AlsaCanPause) {
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
#if 0
//----------------------------------------------------------------------------
// direct playback
//----------------------------------------------------------------------------
// direct play produces underuns on some hardware
/**
** 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)
{
snd_pcm_state_t state;
int avail;
int n;
int err;
int frames;
const void *p;
Debug(3, "audio/alsa: %6zd + %4d\n", RingBufferUsedBytes(AlsaRingBuffer),
count);
n = RingBufferWrite(AlsaRingBuffer, samples, count);
if (n != count) {
Error(_("audio/alsa: can't place %d samples in ring buffer\n"), count);
}
// check if running, wait until enough buffered
state = snd_pcm_state(AlsaPCMHandle);
Debug(4, "audio/alsa: state %d - %s\n", state, snd_pcm_state_name(state));
if (state == SND_PCM_STATE_PREPARED) {
// FIXME: adjust start ratio
if (RingBufferFreeBytes(AlsaRingBuffer)
> RingBufferUsedBytes(AlsaRingBuffer)) {
return;
}
Debug(3, "audio/alsa: state %d - %s start play\n", state,
snd_pcm_state_name(state));
}
// Update audio clock
AudioPTS +=
(size * 90000) / (AudioSampleRate * AudioChannels *
AudioBytesProSample);
}
#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 the buffers
}
}
}
/**
** 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 (!AlsaRingBuffer || !AlsaPCMHandle || !AudioSampleRate) {
printf("%p %p %d\n", 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);
}
}
#endif
/**
** Open alsa pcm device.
*/
static snd_pcm_t *AlsaOpenPCM(void)
{
const char *device;
snd_pcm_t *handle;
int err;
if (!(device = AudioPCMDevice)) {
if (!(device = getenv("ALSA_DEVICE"))) {
device = "default";
}
}
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())) {
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) == 0) {
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
**
** @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)
{
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
int err;
int ret;
snd_pcm_t *handle;
if (!AlsaPCMHandle) { // alsa not running yet
return -1;
}
#if 1
// flush any buffered data
#ifndef SEARCH_HDMI_BUG2
#ifdef USE_AUDIO_THREAD
if (AudioRunning) {
while (AudioRunning) {
AlsaFlushBuffer = 1;
usleep(1 * 1000);
}
AlsaFlushBuffer = 0;
} else
#endif
{
AlsaFlushBuffers();
}
#endif
AudioPTS = INT64_C(0x8000000000000000);
if (1) { // close+open to fix hdmi no sound bugs
handle = AlsaPCMHandle;
AlsaPCMHandle = NULL;
snd_pcm_close(handle);
if (!(handle = AlsaOpenPCM())) {
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);
// min 333ms
if (AlsaStartThreshold < (*freq * *channels * AudioBytesProSample) / 3U) {
AlsaStartThreshold = (*freq * *channels * AudioBytesProSample) / 3U;
}
// 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;
}
}
#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
//----------------------------------------------------------------------------
// 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)
{
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));
return;
}
AudioPTS = INT64_C(0x8000000000000000);
}
//----------------------------------------------------------------------------
// OSS pcm polled
//----------------------------------------------------------------------------
/**
** 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;
}
}
/**
** Play all samples possible, without blocking.
*/
static void OssPoller(void)
{
if (OssPcmFildes == -1) { // setup failure
return;
}
if (AudioRunning) {
OssPlayRingbuffer();
}
}
//----------------------------------------------------------------------------
/**
** Initialize oss pcm device.
**
** @see AudioPCMDevice
*/
static void OssInitPCM(void)
{
const char *device;
int fildes;
if (!(device = AudioPCMDevice)) {
if (!(device = getenv("OSS_AUDIODEV"))) {
device = "/dev/dsp";
}
}
if ((fildes = open(device, O_WRONLY)) < 0) {
Error(_("audio/oss: can't open dsp device '%s': %s\n"), device,
strerror(errno));
return;
}
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
**
** @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 ret;
int tmp;
if (OssPcmFildes == -1) { // oss not ready
return -1;
}
// flush any buffered data
{
AudioRunning = 0;
OssFlushBuffers();
}
AudioPTS = INT64_C(0x8000000000000000);
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;
// min 333ms
if (OssStartThreshold < (*freq * *channels * AudioBytesProSample) / 3U) {
OssStartThreshold = (*freq * *channels * AudioBytesProSample) / 3U;
}
// 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;
}
}
#endif // USE_OSS
//----------------------------------------------------------------------------
// thread playback
//----------------------------------------------------------------------------
#ifdef USE_AUDIO_THREAD
static pthread_t AudioThread; ///< audio play thread
static pthread_mutex_t AudioMutex; ///< audio condition mutex
/**
** 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;
#ifndef SEARCH_HDMI_BUG
do {
pthread_cond_wait(&AudioStartCond, &AudioMutex);
// cond_wait can return, without signal!
} while (!AudioRunning);
#else
usleep(1 * 1000);
AudioRunning = 1;
#endif
pthread_mutex_unlock(&AudioMutex);
#ifdef SEARCH_HDMI_BUG2
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);
if (1) {
int16_t buf[6144 / 2];
buf[0] = htole16(0xF872); // iec 61937 sync word
buf[1] = htole16(0x4E1F);
buf[2] = htole16((7 << 5) << 8 | 0x00);
buf[3] = htole16(0x0000);
memset(buf + 4, 0, 6144 - 8);
AlsaEnqueue(buf, 6144);
}
#endif
}
#endif
Debug(3, "audio: play start\n");
#ifdef USE_ALSA
AlsaThread();
#endif
}
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_detach(AudioThread);
#ifdef very_old_unused_USE_ALSA
// wait until thread has opened and is ready
do {
pthread_yield();
} while (!AlsaPCMHandle);
#endif
pthread_yield();
usleep(5 * 1000);
}
/**
** 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);
}
}
#endif
//----------------------------------------------------------------------------
//----------------------------------------------------------------------------
/**
** 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)
{
#ifdef USE_ALSA
AlsaEnqueue(samples, count);
#endif
#ifdef USE_OSS
OssEnqueue(samples, count);
#endif
(void)samples;
(void)count;
}
/**
** Flush audio buffers.
*/
void AudioFlushBuffers(void)
{
#ifdef USE_ALSA
#ifdef USE_AUDIO_THREAD
if (AudioRunning) {
while (AudioRunning) {
AlsaFlushBuffer = 1;
usleep(1 * 1000);
}
AlsaFlushBuffer = 0;
} else
#endif
{
AlsaFlushBuffers();
}
#endif
#ifdef USE_OSS
OssFlushBuffers();
#endif
}
/**
** Call back to play audio polled.
*/
void AudioPoller(void)
{
#ifndef USE_AUDIO_THREAD
#ifdef USE_ALSA
Error(_("audio/alsa: poller not implemented\n"));
#endif
#ifdef USE_OSS
OssPoller();
#endif
#endif
}
/**
** Get free bytes in audio output.
*/
int AudioFreeBytes(void)
{
#ifdef USE_ALSA
return RingBufferFreeBytes(AlsaRingBuffer);
#endif
#ifdef USE_OSS
return RingBufferFreeBytes(OssRingBuffer);
#endif
return -1;
}
/**
** 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 audio delay in time stamps.
**
** @returns audio delay in time stamps.
*/
uint64_t AudioGetDelay(void)
{
#ifdef USE_ALSA
return AlsaGetDelay();
#endif
#ifdef USE_OSS
return OssGetDelay();
#endif
return 0UL;
}
/**
** Get current audio clock.
**
** @returns the audio clock in time stamps.
*/
int64_t AudioGetClock(void)
{
int64_t delay;
delay = AudioGetDelay();
if (delay && (uint64_t) AudioPTS != INT64_C(0x8000000000000000)) {
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
**
** @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)
{
Debug(3, "audio: channels %d frequency %d hz\n", *channels, *freq);
// invalid parameter
if (!freq || !channels || !*freq || !*channels) {
Debug(3, "audio: bad channels or frequency parameters\n");
// FIXME: set flag invalid setup
return -1;
}
#if defined(SEARCH_HDMI_BUG) || defined(SEARCH_HDMI_BUG2)
// FIXME: need to store possible combination and report this
return AudioRingAdd(*freq, *channels);
#endif
#ifdef USE_ALSA
return AlsaSetup(freq, channels);
#endif
#ifdef USE_OSS
return OssSetup(freq, channels);
#endif
return -1;
}
/**
** Set pcm audio device.
**
** @param device name of pcm device (fe. "hw:0,9" or "/dev/dsp")
*/
void AudioSetDevice(const char *device)
{
AudioPCMDevice = device;
}
/**
** Initialize audio output module.
*/
void AudioInit(void)
{
int freq;
int chan;
#ifdef SEARCH_HDMI_BUG2
AudioRingInit();
#endif
#ifdef USE_ALSA
AlsaInit();
#endif
#ifdef USE_OSS
OssInit();
#endif
freq = 48000;
chan = 2;
if (AudioSetup(&freq, &chan)) { // set default parameters
Error(_("audio: can't do initial setup\n"));
}
#ifdef USE_AUDIO_THREAD
AudioInitThread();
#endif
AudioPaused = 1;
}
/**
** Cleanup audio output module.
*/
void AudioExit(void)
{
#ifdef USE_AUDIO_THREAD
AudioExitThread();
#endif
#ifdef USE_ALSA
AlsaExit();
#endif
#ifdef USE_OSS
OssExit();
#endif
#ifdef SEARCH_HDMI_BUG2
AudioRingExit();
#endif
}
#ifdef AUDIO_TEST
//----------------------------------------------------------------------------
// Test
//----------------------------------------------------------------------------
void AudioTest(void)
{
for (;;) {
unsigned u;
uint8_t buffer[16 * 1024]; // some random data
int i;
for (u = 0; u < sizeof(buffer); u++) {
buffer[u] = random() & 0xffff;
}
Debug(3, "audio/test: loop\n");
for (i = 0; i < 100; ++i) {
while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) {
AlsaEnqueue(buffer, sizeof(buffer));
}
usleep(20 * 1000);
}
break;
}
}
#include <getopt.h>
int SysLogLevel; ///< show additional debug informations
/**
** Print version.
*/
static void PrintVersion(void)
{
printf("audio_test: audio tester Version " VERSION
#ifdef GIT_REV
"(GIT-" GIT_REV ")"
#endif
",\n\t(c) 2009 - 2011 by Johns\n"
"\tLicense AGPLv3: GNU Affero General Public License version 3\n");
}
/**
** Print usage.
*/
static void PrintUsage(void)
{
printf("Usage: audio_test [-?dhv]\n"
"\t-d\tenable debug, more -d increase the verbosity\n"
"\t-? -h\tdisplay this message\n" "\t-v\tdisplay version information\n"
"Only idiots print usage on stderr!\n");
}
/**
** Main entry point.
**
** @param argc number of arguments
** @param argv arguments vector
**
** @returns -1 on failures, 0 clean exit.
*/
int main(int argc, char *const argv[])
{
SysLogLevel = 0;
//
// Parse command line arguments
//
for (;;) {
switch (getopt(argc, argv, "hv?-c:d")) {
case 'd': // enabled debug
++SysLogLevel;
continue;
case EOF:
break;
case 'v': // print version
PrintVersion();
return 0;
case '?':
case 'h': // help usage
PrintVersion();
PrintUsage();
return 0;
case '-':
PrintVersion();
PrintUsage();
fprintf(stderr, "\nWe need no long options\n");
return -1;
case ':':
PrintVersion();
fprintf(stderr, "Missing argument for option '%c'\n", optopt);
return -1;
default:
PrintVersion();
fprintf(stderr, "Unkown option '%c'\n", optopt);
return -1;
}
break;
}
if (optind < argc) {
PrintVersion();
while (optind < argc) {
fprintf(stderr, "Unhandled argument '%s'\n", argv[optind++]);
}
return -1;
}
//
// main loop
//
AudioInit();
for (;;) {
unsigned u;
uint8_t buffer[16 * 1024]; // some random data
for (u = 0; u < sizeof(buffer); u++) {
buffer[u] = random() & 0xffff;
}
Debug(3, "audio/test: loop\n");
for (;;) {
while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) {
AlsaEnqueue(buffer, sizeof(buffer));
}
}
}
AudioExit();
return 0;
}
#endif