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-rw-r--r--audio.c101
1 files changed, 79 insertions, 22 deletions
diff --git a/audio.c b/audio.c
index 20b4c80..3922aed 100644
--- a/audio.c
+++ b/audio.c
@@ -141,7 +141,7 @@ 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
-static const int AudioBufferTime = 350; ///< audio buffer time in ms
+static int AudioBufferTime = 216; ///< audio buffer time in ms
#ifdef USE_AUDIO_THREAD
static pthread_t AudioThread; ///< audio play thread
@@ -151,7 +151,8 @@ static pthread_cond_t AudioStartCond; ///< condition variable
static const int AudioThread; ///< dummy audio thread
#endif
-extern int VideoAudioDelay; /// import audio/video delay
+extern int VideoAudioDelay; ///< import audio/video delay
+extern int VideoGetBuffers(void); ///< Get number of input buffers.
#ifdef USE_AUDIORING
@@ -297,7 +298,17 @@ static int AlsaAddToRingbuffer(const void *samples, int count)
}
if (!AudioRunning) {
- if (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
+ Debug(3, "audio/alsa: start %zd ms %d\n",
+ (RingBufferUsedBytes(AlsaRingBuffer) * 1000)
+ / (AudioSampleRate * AudioChannels * AudioBytesProSample),
+ VideoGetBuffers());
+ // forced start
+ if (AlsaStartThreshold * 2 < RingBufferUsedBytes(AlsaRingBuffer)) {
+ return 1;
+ }
+ // enough video + audio buffered
+ if (VideoGetBuffers() > 1
+ && AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
// restart play-back
return 1;
}
@@ -654,7 +665,8 @@ static void AlsaThread(void)
}
state = snd_pcm_state(AlsaPCMHandle);
if (state != SND_PCM_STATE_RUNNING) {
- Debug(3, "audio/alsa: stopping play\n");
+ Debug(3, "audio/alsa: stopping play '%s'\n",
+ snd_pcm_state_name(state));
break;
}
pthread_yield();
@@ -720,12 +732,11 @@ static snd_pcm_t *AlsaOpenPCM(int use_ac3)
// &&|| hell
if (!(use_ac3 && ((device = AudioAC3Device)
- || (device = getenv("ALSA_AC3_DEVICE"))
- || (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
+ || (device = getenv("ALSA_AC3_DEVICE"))))
&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
device = "default";
}
- Debug(3, "audio/alsa: &&|| hell '%s'\n", device);
+ Info(_("audio/alsa: using device '%s'\n"), device);
// open none blocking; if device is already used, we don't want wait
if ((err =
@@ -752,7 +763,8 @@ static void AlsaInitPCM(void)
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
- snd_pcm_uframes_t buffer_size;
+
+ //snd_pcm_uframes_t buffer_size;
if (!(handle = AlsaOpenPCM(0))) {
return;
@@ -767,8 +779,9 @@ static void AlsaInitPCM(void)
}
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);
+ // needs audio setup
+ //snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
+ //Info(_("audio/alsa: max buffer size %lu\n"), buffer_size);
AlsaPCMHandle = handle;
}
@@ -941,7 +954,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
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))) {
+ 96 * 1000))) {
Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err));
/*
@@ -1053,7 +1066,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
// FIXME: use hw_params for buffer_size period_size
#endif
-#if 1
+#if 0
if (0) { // no underruns allowed, play silence
snd_pcm_sw_params_t *sw_params;
snd_pcm_uframes_t boundary;
@@ -1091,16 +1104,21 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
// 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);
+ Info(_("audio/alsa: buffer size %lu %lums, period size %lu %lums\n"),
+ buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle,
+ buffer_size) * 1000 / (AudioSampleRate * AudioChannels *
+ AudioBytesProSample), period_size,
+ snd_pcm_frames_to_bytes(AlsaPCMHandle,
+ period_size) * 1000 / (AudioSampleRate * AudioChannels *
+ AudioBytesProSample));
Debug(3, "audio/alsa: state %s\n",
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
// buffer time/delay in ms
delay = AudioBufferTime;
- if (VideoAudioDelay > -100) {
- delay += 100 + VideoAudioDelay / 90;
+ if (VideoAudioDelay > 0) {
+ delay += VideoAudioDelay / 90;
}
if (AlsaStartThreshold <
(*freq * *channels * AudioBytesProSample * delay) / 1000U) {
@@ -1284,7 +1302,17 @@ static int OssAddToRingbuffer(const void *samples, int count)
}
if (!AudioRunning) {
- if (OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
+ Debug(3, "audio/oss: start %zd ms %d\n",
+ (RingBufferUsedBytes(OssRingBuffer) * 1000)
+ / (AudioSampleRate * AudioChannels * AudioBytesProSample),
+ VideoGetBuffers());
+ // forced start
+ if (OssStartThreshold * 2 < RingBufferUsedBytes(OssRingBuffer)) {
+ return 1;
+ }
+ // enough video + audio buffered
+ if (VideoGetBuffers() > 1
+ && OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
// restart play-back
return 1;
}
@@ -1522,7 +1550,7 @@ static int OssOpenPCM(int use_ac3)
&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
device = "/dev/dsp";
}
- Debug(3, "audio/oss: &&|| hell '%s'\n", device);
+ Info(_("audio/oss: using device '%s'\n"), device);
if ((fildes = open(device, O_WRONLY)) < 0) {
Error(_("audio/oss: can't open dsp device '%s': %s\n"), device,
@@ -1774,8 +1802,8 @@ static int OssSetup(int *freq, int *channels, int use_ac3)
OssStartThreshold = bi.bytes + tmp;
// buffer time/delay in ms
delay = AudioBufferTime;
- if (VideoAudioDelay > -100) {
- delay += 100 + VideoAudioDelay / 90;
+ if (VideoAudioDelay > 0) {
+ delay += VideoAudioDelay / 90;
}
if (OssStartThreshold <
(*freq * *channels * AudioBytesProSample * delay) / 1000U) {
@@ -1964,6 +1992,9 @@ static void *AudioPlayHandlerThread(void *dummy)
pthread_cond_wait(&AudioStartCond, &AudioMutex);
// cond_wait can return, without signal!
} while (!AudioRunning);
+ Debug(3, "audio/alsa: ----> %zd ms\n",
+ (RingBufferUsedBytes(AlsaRingBuffer) * 1000)
+ / (AudioSampleRate * AudioChannels * AudioBytesProSample));
pthread_mutex_unlock(&AudioMutex);
#ifdef USE_AUDIORING
@@ -2064,6 +2095,21 @@ static const AudioModule *AudioModules[] = {
*/
void AudioEnqueue(const void *samples, int count)
{
+ if (0) {
+ static uint32_t last;
+ static uint32_t tick;
+ static uint32_t max = 110;
+ uint64_t delay;
+
+ delay = AudioGetDelay();
+ tick = GetMsTicks();
+ if ((last && tick - last > max) || delay < 80 * 90) {
+
+ //max = tick - last;
+ Debug(3, "audio: packet delta %d %lu\n", tick - last, delay / 90);
+ }
+ last = tick;
+ }
AudioUsedModule->Enqueue(samples, count);
}
@@ -2183,7 +2229,7 @@ int AudioSetup(int *freq, int *channels, int use_ac3)
void AudioPlay(void)
{
if (!AudioPaused) {
- Warning("audio: not paused, check the code\n");
+ Debug(3, "audio: not paused, check the code\n");
return;
}
Debug(3, "audio: resumed\n");
@@ -2197,7 +2243,7 @@ void AudioPlay(void)
void AudioPause(void)
{
if (AudioPaused) {
- Warning("audio: already paused, check the code\n");
+ Debug(3, "audio: already paused, check the code\n");
return;
}
Debug(3, "audio: paused\n");
@@ -2205,6 +2251,17 @@ void AudioPause(void)
}
/**
+** Set audio buffer time.
+*/
+void AudioSetBufferTime(int delay)
+{
+ if (!delay) {
+ delay = 216;
+ }
+ AudioBufferTime = delay;
+}
+
+/**
** Set pcm audio device.
**
** @param device name of pcm device (fe. "hw:0,9" or "/dev/dsp")
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-14 16:47:53 +0000