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vdr-plugin-softhddevice
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Add workaround for alsa not playing hdmi sound. 

Without open/close pcm, hdmi is quiet after second snd_pcm_set_params.
This commit is contained in:
Johns 2012-01-07 02:35:49 +01:00
parent 878813f206
commit 92ffd978b0
1 changed files with 258 additions and 24 deletions
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282
audio.c
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@ -43,6 +43,8 @@
#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>
@ -88,6 +90,8 @@
#endif
#endif
#include <alsa/iatomic.h> // portable atomic_t
#include "ringbuffer.h"
#include "misc.h"
#include "audio.h"
@ -110,6 +114,95 @@ static int64_t AudioPTS; ///< audio pts clock
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
//============================================================================
@ -152,6 +245,7 @@ static int AlsaAddToRingbuffer(const void *samples, int 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 +=
@ -207,7 +301,19 @@ static int AlsaPlayRingbuffer(void)
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
@ -215,6 +321,7 @@ static int AlsaPlayRingbuffer(void)
}
return 0;
}
#endif
if (n < avail) { // not enough bytes in ring buffer
avail = n;
}
@ -229,7 +336,7 @@ static int AlsaPlayRingbuffer(void)
} else {
err = snd_pcm_writei(AlsaPCMHandle, p, frames);
}
Debug(4, "audio/alsa: wrote %d/%d frames\n", err, frames);
//Debug(3, "audio/alsa: wrote %d/%d frames\n", err, frames);
if (err != frames) {
if (err < 0) {
if (err == -EAGAIN) {
@ -255,6 +362,19 @@ static int AlsaPlayRingbuffer(void)
return 0;
}
/**
** Flush alsa buffers.
*/
static void AlsaFlushBuffers(void)
{
int err;
RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer));
if ((err = snd_pcm_drop(AlsaPCMHandle))) {
Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
}
}
#if 0
//----------------------------------------------------------------------------
@ -462,16 +582,10 @@ static void AlsaThread(void)
if (AlsaFlushBuffer) {
// we can flush too many, but wo cares
Debug(3, "audio/alsa: flushing buffers\n");
RingBufferReadAdvance(AlsaRingBuffer,
RingBufferUsedBytes(AlsaRingBuffer));
#if 1
if ((err = snd_pcm_drop(AlsaPCMHandle))) {
Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
}
AlsaFlushBuffers();
if ((err = snd_pcm_prepare(AlsaPCMHandle))) {
Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err));
}
#endif
AlsaFlushBuffer = 0;
break;
}
@ -499,8 +613,9 @@ static void AlsaThread(void)
*/
static void AlsaEnqueue(const void *samples, int count)
{
if (!AlsaRingBuffer || !AlsaPCMHandle) {
Debug(3, "audio/alsa: not ready\n");
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)) {
@ -518,35 +633,48 @@ static void AlsaEnqueue(const void *samples, int count)
#endif
/**
** Initialize alsa pcm device.
**
** @see AudioPCMDevice
** Open alsa pcm device.
*/
static void AlsaInitPCM(void)
static snd_pcm_t *AlsaOpenPCM(void)
{
const char *device;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
snd_pcm_uframes_t buffer_size;
if (!(device = AudioPCMDevice)) {
if (!(device = getenv("ALSA_DEVICE"))) {
device = "default";
}
}
// FIXME: must set alsa error output to /dev/null
if ((err =
snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK)) < 0) {
Fatal(_("audio/alsa: playback open '%s' error: %s\n"), device,
Error(_("audio/alsa: playback open '%s' error: %s\n"), device,
snd_strerror(err));
// FIXME: no fatal error for plugins!
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
@ -556,8 +684,7 @@ static void AlsaInitPCM(void)
snd_strerror(err));
}
AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params);
Info(_("audio/alsa: hw '%s' supports pause: %s\n"), device,
AlsaCanPause ? "yes" : "no");
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);
@ -656,7 +783,7 @@ static uint64_t AlsaGetDelay(void)
snd_pcm_sframes_t delay;
uint64_t pts;
if (!AlsaPCMHandle) {
if (!AlsaPCMHandle || !AudioSampleRate) {
return 0UL;
}
// FIXME: thread safe? __assert_fail_base in snd_pcm_delay
@ -702,12 +829,14 @@ static int AlsaSetup(int *freq, int *channels)
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) {
@ -718,11 +847,21 @@ static int AlsaSetup(int *freq, int *channels)
} else
#endif
{
RingBufferReadAdvance(AlsaRingBuffer,
RingBufferUsedBytes(AlsaRingBuffer));
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;
@ -844,6 +983,41 @@ static int AlsaSetup(int *freq, int *channels)
// 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);
@ -954,6 +1128,7 @@ static int OssAddToRingbuffer(const void *samples, int 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 +=
@ -1375,12 +1550,59 @@ static void *AudioPlayHandlerThread(void *dummy)
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();
@ -1406,6 +1628,8 @@ static void AudioInitThread(void)
pthread_yield();
} while (!AlsaPCMHandle);
#endif
pthread_yield();
usleep(5 * 1000);
}
/**
@ -1550,6 +1774,10 @@ int AudioSetup(int *freq, int *channels)
// 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
@ -1577,6 +1805,9 @@ void AudioInit(void)
int freq;
int chan;
#ifdef SEARCH_HDMI_BUG2
AudioRingInit();
#endif
#ifdef USE_ALSA
AlsaInit();
#endif
@ -1609,6 +1840,9 @@ void AudioExit(void)
#ifdef USE_OSS
OssExit();
#endif
#ifdef SEARCH_HDMI_BUG2
AudioRingExit();
#endif
}
#ifdef AUDIO_TEST