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authorJohns <johns98@gmx.net>2012-01-07 02:35:49 +0100
committerJohns <johns98@gmx.net>2012-01-07 02:35:49 +0100
commit92ffd978b00e1d7138fc965edbf84032192f3fbd (patch)
tree5636331bc8a051f6c3d6838e3d853c58118c1be2 /audio.c
parent878813f206768172aa13ab8b90c37bdb8e5f31ba (diff)
downloadvdr-plugin-softhddevice-92ffd978b00e1d7138fc965edbf84032192f3fbd.tar.gz
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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.
Diffstat (limited to 'audio.c')
-rw-r--r--audio.c282
1 files changed, 258 insertions, 24 deletions
diff --git a/audio.c b/audio.c
index b618250..4afb154 100644
--- a/audio.c
+++ b/audio.c
@@ -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