mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
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
parent
878813f206
commit
92ffd978b0
282
audio.c
282
audio.c
@ -43,6 +43,8 @@
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#endif
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//#define USE_ALSA ///< enable alsa support
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//#define USE_OSS ///< enable oss support
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#define noSEARCH_HDMI_BUG
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#define noSEARCH_HDMI_BUG2
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#include <stdio.h>
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#include <stdint.h>
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@ -88,6 +90,8 @@
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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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@ -110,6 +114,95 @@ static int64_t AudioPTS; ///< audio pts clock
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static pthread_cond_t AudioStartCond; ///< condition variable
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#endif
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#ifdef SEARCH_HDMI_BUG2
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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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@ -152,6 +245,7 @@ static int AlsaAddToRingbuffer(const void *samples, int 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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@ -207,7 +301,19 @@ static int AlsaPlayRingbuffer(void)
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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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#ifdef SEARCH_HDMI_BUG
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{
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uint16_t buf[8192];
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unsigned u;
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for (u = 0; u < sizeof(buf) / 2; u++) {
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buf[u] = random() & 0xffff;
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}
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n = sizeof(buf);
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p = buf;
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}
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#else
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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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@ -215,6 +321,7 @@ static int AlsaPlayRingbuffer(void)
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}
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return 0;
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}
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#endif
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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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@ -229,7 +336,7 @@ static int AlsaPlayRingbuffer(void)
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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(4, "audio/alsa: wrote %d/%d frames\n", err, frames);
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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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@ -255,6 +362,19 @@ static int AlsaPlayRingbuffer(void)
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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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RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer));
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if ((err = snd_pcm_drop(AlsaPCMHandle))) {
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Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
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}
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}
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#if 0
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//----------------------------------------------------------------------------
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@ -462,16 +582,10 @@ static void AlsaThread(void)
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if (AlsaFlushBuffer) {
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// we can flush too many, but wo cares
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Debug(3, "audio/alsa: flushing buffers\n");
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RingBufferReadAdvance(AlsaRingBuffer,
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RingBufferUsedBytes(AlsaRingBuffer));
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#if 1
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if ((err = snd_pcm_drop(AlsaPCMHandle))) {
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Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
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}
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AlsaFlushBuffers();
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if ((err = snd_pcm_prepare(AlsaPCMHandle))) {
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Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err));
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}
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#endif
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AlsaFlushBuffer = 0;
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break;
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}
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@ -499,8 +613,9 @@ static void AlsaThread(void)
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*/
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static void AlsaEnqueue(const void *samples, int count)
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{
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if (!AlsaRingBuffer || !AlsaPCMHandle) {
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Debug(3, "audio/alsa: not ready\n");
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if (!AlsaRingBuffer || !AlsaPCMHandle || !AudioSampleRate) {
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printf("%p %p %d\n", AlsaRingBuffer, AlsaPCMHandle, AudioSampleRate);
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Debug(3, "audio/alsa: enqueue not ready\n");
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return;
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}
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if (AlsaAddToRingbuffer(samples, count)) {
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@ -518,35 +633,48 @@ static void AlsaEnqueue(const void *samples, int count)
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#endif
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/**
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** Initialize alsa pcm device.
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**
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** @see AudioPCMDevice
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** Open alsa pcm device.
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*/
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static void AlsaInitPCM(void)
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static snd_pcm_t *AlsaOpenPCM(void)
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{
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const char *device;
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snd_pcm_t *handle;
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snd_pcm_hw_params_t *hw_params;
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int err;
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snd_pcm_uframes_t buffer_size;
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if (!(device = AudioPCMDevice)) {
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if (!(device = getenv("ALSA_DEVICE"))) {
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device = "default";
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}
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}
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// FIXME: must set alsa error output to /dev/null
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if ((err =
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snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK,
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SND_PCM_NONBLOCK)) < 0) {
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Fatal(_("audio/alsa: playback open '%s' error: %s\n"), device,
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Error(_("audio/alsa: playback open '%s' error: %s\n"), device,
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snd_strerror(err));
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// FIXME: no fatal error for plugins!
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return NULL;
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}
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if ((err = snd_pcm_nonblock(handle, 0)) < 0) {
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Error(_("audio/alsa: can't set block mode: %s\n"), snd_strerror(err));
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}
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return handle;
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}
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/**
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** Initialize alsa pcm device.
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**
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** @see AudioPCMDevice
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*/
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static void AlsaInitPCM(void)
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{
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snd_pcm_t *handle;
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snd_pcm_hw_params_t *hw_params;
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int err;
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snd_pcm_uframes_t buffer_size;
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if (!(handle = AlsaOpenPCM())) {
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return;
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}
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snd_pcm_hw_params_alloca(&hw_params);
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// choose all parameters
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@ -556,8 +684,7 @@ static void AlsaInitPCM(void)
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snd_strerror(err));
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}
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AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params);
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Info(_("audio/alsa: hw '%s' supports pause: %s\n"), device,
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AlsaCanPause ? "yes" : "no");
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Info(_("audio/alsa: supports pause: %s\n"), AlsaCanPause ? "yes" : "no");
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snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
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Info(_("audio/alsa: max buffer size %lu\n"), buffer_size);
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@ -656,7 +783,7 @@ static uint64_t AlsaGetDelay(void)
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snd_pcm_sframes_t delay;
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uint64_t pts;
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if (!AlsaPCMHandle) {
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if (!AlsaPCMHandle || !AudioSampleRate) {
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return 0UL;
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}
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// FIXME: thread safe? __assert_fail_base in snd_pcm_delay
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@ -702,12 +829,14 @@ static int AlsaSetup(int *freq, int *channels)
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snd_pcm_uframes_t period_size;
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int err;
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int ret;
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snd_pcm_t *handle;
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if (!AlsaPCMHandle) { // alsa not running yet
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return -1;
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}
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#if 1
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// flush any buffered data
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#ifndef SEARCH_HDMI_BUG2
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#ifdef USE_AUDIO_THREAD
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if (AudioRunning) {
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while (AudioRunning) {
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@ -718,11 +847,21 @@ static int AlsaSetup(int *freq, int *channels)
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} else
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#endif
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{
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RingBufferReadAdvance(AlsaRingBuffer,
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RingBufferUsedBytes(AlsaRingBuffer));
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AlsaFlushBuffers();
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}
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#endif
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AudioPTS = INT64_C(0x8000000000000000);
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if (1) { // close+open to fix hdmi no sound bugs
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handle = AlsaPCMHandle;
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AlsaPCMHandle = NULL;
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snd_pcm_close(handle);
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if (!(handle = AlsaOpenPCM())) {
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return -1;
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}
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AlsaPCMHandle = handle;
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}
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ret = 0;
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try_again:
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AudioChannels = *channels;
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@ -844,6 +983,41 @@ static int AlsaSetup(int *freq, int *channels)
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// FIXME: use hw_params for buffer_size period_size
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#endif
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#if 1
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if (0) { // no underruns allowed, play silence
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snd_pcm_sw_params_t *sw_params;
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snd_pcm_uframes_t boundary;
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snd_pcm_sw_params_alloca(&sw_params);
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err = snd_pcm_sw_params_current(AlsaPCMHandle, sw_params);
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if (err < 0) {
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Error(_("audio: snd_pcm_sw_params_current failed: %s\n"),
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snd_strerror(err));
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}
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if ((err = snd_pcm_sw_params_get_boundary(sw_params, &boundary)) < 0) {
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Error(_("audio: snd_pcm_sw_params_get_boundary failed: %s\n"),
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snd_strerror(err));
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}
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Debug(4, "audio/alsa: boundary %lu frames\n", boundary);
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if ((err =
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snd_pcm_sw_params_set_stop_threshold(AlsaPCMHandle, sw_params,
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boundary)) < 0) {
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Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"),
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snd_strerror(err));
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}
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if ((err =
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snd_pcm_sw_params_set_silence_size(AlsaPCMHandle, sw_params,
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boundary)) < 0) {
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Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"),
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snd_strerror(err));
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}
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if ((err = snd_pcm_sw_params(AlsaPCMHandle, sw_params)) < 0) {
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Error(_("audio: snd_pcm_sw_params failed: %s\n"),
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snd_strerror(err));
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}
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}
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#endif
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// update buffer
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snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size);
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@ -954,6 +1128,7 @@ static int OssAddToRingbuffer(const void *samples, int count)
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if (n != count) {
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Error(_("audio/oss: 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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@ -1375,12 +1550,59 @@ static void *AudioPlayHandlerThread(void *dummy)
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Debug(3, "audio: wait on start condition\n");
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pthread_mutex_lock(&AudioMutex);
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AudioRunning = 0;
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#ifndef SEARCH_HDMI_BUG
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do {
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pthread_cond_wait(&AudioStartCond, &AudioMutex);
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// cond_wait can return, without signal!
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} while (!AudioRunning);
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#else
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usleep(1 * 1000);
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AudioRunning = 1;
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#endif
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pthread_mutex_unlock(&AudioMutex);
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#ifdef SEARCH_HDMI_BUG2
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if (atomic_read(&AudioRingFilled) > 1) {
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int sample_rate;
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int channels;
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// skip all sample changes between
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while (atomic_read(&AudioRingFilled) > 1) {
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Debug(3, "audio: skip ring buffer\n");
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AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX;
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atomic_dec(&AudioRingFilled);
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}
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#ifdef USE_ALSA
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// FIXME: flush only if there is something to flush
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AlsaFlushBuffers();
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sample_rate = AudioRing[AudioRingRead].SampleRate;
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channels = AudioRing[AudioRingRead].Channels;
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Debug(3, "audio: thread channels %d sample-rate %d hz\n", channels,
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sample_rate);
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if (AlsaSetup(&sample_rate, &channels)) {
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Error(_("audio: can't set channels %d sample-rate %d hz\n"),
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channels, sample_rate);
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}
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Debug(3, "audio: thread channels %d sample-rate %d hz\n",
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AudioChannels, AudioSampleRate);
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if (1) {
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int16_t buf[6144 / 2];
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buf[0] = htole16(0xF872); // iec 61937 sync word
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buf[1] = htole16(0x4E1F);
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buf[2] = htole16((7 << 5) << 8 | 0x00);
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buf[3] = htole16(0x0000);
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memset(buf + 4, 0, 6144 - 8);
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AlsaEnqueue(buf, 6144);
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}
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#endif
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}
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#endif
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Debug(3, "audio: play start\n");
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#ifdef USE_ALSA
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AlsaThread();
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@ -1406,6 +1628,8 @@ static void AudioInitThread(void)
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pthread_yield();
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} while (!AlsaPCMHandle);
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#endif
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pthread_yield();
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usleep(5 * 1000);
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}
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/**
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@ -1550,6 +1774,10 @@ int AudioSetup(int *freq, int *channels)
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// FIXME: set flag invalid setup
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return -1;
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}
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#if defined(SEARCH_HDMI_BUG) || defined(SEARCH_HDMI_BUG2)
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// FIXME: need to store possible combination and report this
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return AudioRingAdd(*freq, *channels);
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#endif
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#ifdef USE_ALSA
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return AlsaSetup(freq, channels);
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#endif
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@ -1577,6 +1805,9 @@ void AudioInit(void)
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int freq;
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int chan;
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#ifdef SEARCH_HDMI_BUG2
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AudioRingInit();
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#endif
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#ifdef USE_ALSA
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AlsaInit();
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#endif
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@ -1609,6 +1840,9 @@ void AudioExit(void)
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#ifdef USE_OSS
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OssExit();
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#endif
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#ifdef SEARCH_HDMI_BUG2
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AudioRingExit();
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#endif
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}
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#ifdef AUDIO_TEST
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