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authorJohns <johns98@gmx.net>2012-03-02 18:16:20 +0100
committerJohns <johns98@gmx.net>2012-03-02 18:16:20 +0100
commitb0d9f4102074e85cf4594ad712f0239fa93082c9 (patch)
treef3d664138413bf9bd2459a5c52c69bb6a05665b5
parent4d1a516c808202141af4268199ae09e9c4bec3fc (diff)
downloadvdr-plugin-softhddevice-b0d9f4102074e85cf4594ad712f0239fa93082c9.tar.gz
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Rewrote video/audio start code.
-rw-r--r--ChangeLog1
-rw-r--r--audio.c120
-rw-r--r--audio.h3
-rw-r--r--video.c63
4 files changed, 148 insertions, 39 deletions
diff --git a/ChangeLog b/ChangeLog
index f0bcfb7..81c1334 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,6 +1,7 @@
User johns
Date:
+ Rewrote video/audio start code.
Add support for attach/detach plugin.
OSS needs bigger audio buffers.
Improved audio drift correction support.
diff --git a/audio.c b/audio.c
index 8767f4e..d2e8323 100644
--- a/audio.c
+++ b/audio.c
@@ -107,9 +107,11 @@ typedef struct _audio_module_
void (*Thread) (void); ///< module thread handler
void (*Enqueue) (const void *, int); ///< enqueue samples for output
+ void (*VideoReady) (void); ///< video ready, start audio
void (*FlushBuffers) (void); ///< flush sample buffers
void (*Poller) (void); ///< output poller
int (*FreeBytes) (void); ///< number of bytes free in buffer
+ int (*UsedBytes) (void); ///< number of bytes used in buffer
uint64_t(*GetDelay) (void); ///< get current audio delay
void (*SetVolume) (int); ///< set output volume
int (*Setup) (int *, int *, int); ///< setup channels, samplerate
@@ -137,6 +139,7 @@ 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 volatile char AudioPaused; ///< audio paused
+static volatile char AudioVideoIsReady; ///< video ready start early
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
@@ -152,7 +155,6 @@ static const int AudioThread; ///< dummy audio thread
#endif
extern int VideoAudioDelay; ///< import audio/video delay
-extern int VideoGetBuffers(void); ///< Get number of input buffers.
#ifdef USE_AUDIORING
@@ -292,16 +294,16 @@ static int AlsaAddToRingbuffer(const void *samples, int count)
}
if (!AudioRunning) {
- Debug(3, "audio/alsa: start %4zd ms %d v-buf\n",
+ Debug(4, "audio/alsa: start %4zdms\n",
(RingBufferUsedBytes(AlsaRingBuffer) * 1000)
- / (AudioSampleRate * AudioChannels * AudioBytesProSample),
- VideoGetBuffers());
+ / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+
// forced start
- if (AlsaStartThreshold * 3 < RingBufferUsedBytes(AlsaRingBuffer)) {
+ if (AlsaStartThreshold * 2 < RingBufferUsedBytes(AlsaRingBuffer)) {
return 1;
}
// enough video + audio buffered
- if (VideoGetBuffers() > 1
+ if (AudioVideoIsReady
&& AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
// restart play-back
return 1;
@@ -446,6 +448,7 @@ static void AlsaFlushBuffers(void)
}
}
AudioRunning = 0;
+ AudioVideoIsReady = 0;
AudioPTS = INT64_C(0x8000000000000000);
}
@@ -470,6 +473,14 @@ static int AlsaFreeBytes(void)
return AlsaRingBuffer ? RingBufferFreeBytes(AlsaRingBuffer) : INT32_MAX;
}
+/**
+** Get used bytes in audio output.
+*/
+static int AlsaUsedBytes(void)
+{
+ return AlsaRingBuffer ? RingBufferUsedBytes(AlsaRingBuffer) : 0;
+}
+
#if 0
//----------------------------------------------------------------------------
@@ -707,6 +718,26 @@ static void AlsaThreadEnqueue(const void *samples, int count)
}
/**
+** Video is ready, start audio if possible,
+*/
+static void AlsaVideoReady(void)
+{
+ if (AudioSampleRate && AudioChannels) {
+ Debug(3, "audio/alsa: start %4zdms video start\n",
+ (RingBufferUsedBytes(AlsaRingBuffer) * 1000)
+ / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+ }
+
+ if (!AudioRunning) {
+ // enough video + audio buffered
+ if (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
+ AudioRunning = 1;
+ pthread_cond_signal(&AudioStartCond);
+ }
+ }
+}
+
+/**
** Flush alsa buffers with thread.
*/
static void AlsaThreadFlushBuffers(void)
@@ -1245,13 +1276,16 @@ static const AudioModule AlsaModule = {
#ifdef USE_AUDIO_THREAD
.Thread = AlsaThread,
.Enqueue = AlsaThreadEnqueue,
+ .VideoReady = AlsaVideoReady,
.FlushBuffers = AlsaThreadFlushBuffers,
#else
.Enqueue = AlsaEnqueue,
+ .VideoReady = AlsaVideoReady,
.FlushBuffers = AlsaFlushBuffers,
#endif
.Poller = AlsaPoller,
.FreeBytes = AlsaFreeBytes,
+ .UsedBytes = AlsaUsedBytes,
.GetDelay = AlsaGetDelay,
.SetVolume = AlsaSetVolume,
.Setup = AlsaSetup,
@@ -1308,16 +1342,16 @@ static int OssAddToRingbuffer(const void *samples, int count)
}
if (!AudioRunning) {
- Debug(3, "audio/oss: start %4zd ms %d v-buf\n",
+ Debug(4, "audio/oss: start %4zdms\n",
(RingBufferUsedBytes(OssRingBuffer) * 1000)
- / (AudioSampleRate * AudioChannels * AudioBytesProSample),
- VideoGetBuffers());
+ / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+
// forced start
- if (OssStartThreshold * 3 < RingBufferUsedBytes(OssRingBuffer)) {
+ if (OssStartThreshold * 2 < RingBufferUsedBytes(OssRingBuffer)) {
return 1;
}
// enough video + audio buffered
- if (VideoGetBuffers() > 1
+ if (AudioVideoIsReady
&& OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
// restart play-back
return 1;
@@ -1392,6 +1426,7 @@ static void OssFlushBuffers(void)
}
}
AudioRunning = 0;
+ AudioVideoIsReady = 0;
AudioPTS = INT64_C(0x8000000000000000);
}
@@ -1450,6 +1485,14 @@ static int OssFreeBytes(void)
return OssRingBuffer ? RingBufferFreeBytes(OssRingBuffer) : INT32_MAX;
}
+/**
+** Get used bytes in audio output.
+*/
+static int OssUsedBytes(void)
+{
+ return OssRingBuffer ? RingBufferUsedBytes(OssRingBuffer) : 0;
+}
+
#ifdef USE_AUDIO_THREAD
//----------------------------------------------------------------------------
@@ -1521,6 +1564,26 @@ static void OssThreadEnqueue(const void *samples, int count)
}
/**
+** Video is ready, start audio if possible,
+*/
+static void OssVideoReady(void)
+{
+ if (AudioSampleRate && AudioChannels) {
+ Debug(3, "audio/oss: start %4zdms video start\n",
+ (RingBufferUsedBytes(OssRingBuffer) * 1000)
+ / (AudioSampleRate * AudioChannels * AudioBytesProSample));
+ }
+
+ if (!AudioRunning) {
+ // enough video + audio buffered
+ if (OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
+ AudioRunning = 1;
+ pthread_cond_signal(&AudioStartCond);
+ }
+ }
+}
+
+/**
** Flush OSS buffers with thread.
*/
static void OssThreadFlushBuffers(void)
@@ -1881,13 +1944,16 @@ static const AudioModule OssModule = {
#ifdef USE_AUDIO_THREAD
.Thread = OssThread,
.Enqueue = OssThreadEnqueue,
+ .VideoReady = OssVideoReady,
.FlushBuffers = OssThreadFlushBuffers,
#else
.Enqueue = OssEnqueue,
+ .VideoReady = OssVideoReady,
.FlushBuffers = OssFlushBuffers,
#endif
.Poller = OssPoller,
.FreeBytes = OssFreeBytes,
+ .UsedBytes = OssUsedBytes,
.GetDelay = OssGetDelay,
.SetVolume = OssSetVolume,
.Setup = OssSetup,
@@ -1924,6 +1990,14 @@ static int NoopFreeBytes(void)
}
/**
+** Get used bytes in audio output.
+*/
+static int NoopUsedBytes(void)
+{
+ return 0; // no driver, nothing used
+}
+
+/**
** Get audio delay in time stamps.
**
** @returns audio delay in time stamps.
@@ -1970,9 +2044,11 @@ static void NoopVoid(void)
static const AudioModule NoopModule = {
.Name = "noop",
.Enqueue = NoopEnqueue,
+ .VideoReady = NoopVoid,
.FlushBuffers = NoopVoid,
.Poller = NoopVoid,
.FreeBytes = NoopFreeBytes,
+ .UsedBytes = NoopUsedBytes,
.GetDelay = NoopGetDelay,
.SetVolume = NoopSetVolume,
.Setup = NoopSetup,
@@ -2002,9 +2078,10 @@ 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)
+
+ Debug(3, "audio: ----> %d ms\n", (AudioUsedBytes() * 1000)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
+
pthread_mutex_unlock(&AudioMutex);
#ifdef USE_AUDIORING
@@ -2131,6 +2208,15 @@ void AudioEnqueue(const void *samples, int count)
}
/**
+** Video is ready.
+*/
+void AudioVideoReady(void)
+{
+ AudioVideoIsReady = 1;
+ AudioUsedModule->VideoReady();
+}
+
+/**
** Flush audio buffers.
*/
void AudioFlushBuffers(void)
@@ -2155,6 +2241,14 @@ int AudioFreeBytes(void)
}
/**
+** Get used bytes in audio output.
+*/
+int AudioUsedBytes(void)
+{
+ return AudioUsedModule->UsedBytes();
+}
+
+/**
** Get audio delay in time stamps.
**
** @returns audio delay in time stamps.
diff --git a/audio.h b/audio.h
index 9163f72..94ac55a 100644
--- a/audio.h
+++ b/audio.h
@@ -31,8 +31,7 @@ extern void AudioEnqueue(const void *, int); ///< buffer audio samples
extern void AudioFlushBuffers(void); ///< flush audio buffers
extern void AudioPoller(void); ///< poll audio events/handling
extern int AudioFreeBytes(void); ///< free bytes in audio output
-
-//extern int AudioUsedBytes(void); ///< used bytes in audio output
+extern int AudioUsedBytes(void); ///< used bytes in audio output
extern uint64_t AudioGetDelay(void); ///< get current audio delay
extern void AudioSetClock(int64_t); ///< set audio clock base
extern int64_t AudioGetClock(); ///< get current audio clock
diff --git a/video.c b/video.c
index 92d3b7e..434fc1e 100644
--- a/video.c
+++ b/video.c
@@ -331,7 +331,7 @@ int VideoAudioDelay;
/// Default zoom mode
static VideoZoomModes Video4to3ZoomMode;
-//static char VideoSoftStartSync; ///< soft start sync audio/video
+static char VideoSoftStartSync = 1; ///< soft start sync audio/video
static char Video60HzMode; ///< handle 60hz displays
@@ -340,6 +340,7 @@ static xcb_atom_t NetWmState; ///< wm-state message atom
static xcb_atom_t NetWmStateFullscreen; ///< fullscreen wm-state message atom
extern uint32_t VideoSwitch; ///< ticks for channel switch
+extern void AudioVideoReady(void); ///< tell audio video is ready
#ifdef USE_VIDEO_THREAD
@@ -381,6 +382,9 @@ static const char *VideoTimeStampString(int64_t ts)
int ss;
int uu;
+ if (ts == (int64_t) AV_NOPTS_VALUE) {
+ return "--:--:--.---";
+ }
idx ^= 1; // support two static buffers
ts = ts / 90;
uu = ts % 1000;
@@ -410,6 +414,7 @@ static void VideoSetPts(int64_t * pts_p, int interlaced, const AVFrame * frame)
if (*pts_p != (int64_t) AV_NOPTS_VALUE) {
*pts_p += interlaced ? 40 * 90 : 20 * 90;
}
+ //av_opt_ptr(avcodec_get_frame_class(), frame, "best_effort_timestamp");
//pts = frame->best_effort_timestamp;
pts = frame->pkt_pts;
if (pts == (int64_t) AV_NOPTS_VALUE || !pts) {
@@ -1342,6 +1347,7 @@ struct _vaapi_decoder_
struct timespec FrameTime; ///< time of last display
int64_t PTS; ///< video PTS clock
+ int StartCounter; ///< number of start frames
int FramesDuped; ///< number of frames duplicated
int FramesMissed; ///< number of frames missed
int FramesDropped; ///< number of frames dropped
@@ -1813,10 +1819,13 @@ static void VaapiCleanup(VaapiDecoder * decoder)
if (decoder->DeintImages[0].image_id != VA_INVALID_ID) {
VaapiDestroyDeinterlaceImages(decoder);
}
+ decoder->SurfaceRead = 0;
+ decoder->SurfaceWrite = 0;
decoder->SurfaceField = 1;
//decoder->FrameCounter = 0;
+ decoder->StartCounter = 0;
decoder->PTS = AV_NOPTS_VALUE;
VideoDeltaPTS = 0;
}
@@ -4379,6 +4388,14 @@ static void VaapiSyncDisplayFrame(VaapiDecoder * decoder)
filled = atomic_read(&decoder->SurfacesFilled);
// FIXME: audio not known assume 333ms delay
+ decoder->StartCounter++;
+ if (!VideoSoftStartSync && decoder->StartCounter < 60
+ && (audio_clock == (int64_t) AV_NOPTS_VALUE
+ || video_clock > audio_clock + VideoAudioDelay + 120 * 90)) {
+ Debug(3, "video: initial slow down %d\n", decoder->StartCounter);
+ decoder->DupNextFrame = 2;
+ }
+
if (decoder->DupNextFrame) {
decoder->DupNextFrame--;
++decoder->FramesDuped;
@@ -4388,23 +4405,17 @@ static void VaapiSyncDisplayFrame(VaapiDecoder * decoder)
if (abs(video_clock - audio_clock) > 5000 * 90) {
Debug(3, "video: pts difference too big\n");
- } else if (video_clock > audio_clock + VideoAudioDelay + 80 * 90) {
+ } else if (video_clock > audio_clock + VideoAudioDelay + 100 * 90) {
Debug(3, "video: slow down video\n");
decoder->DupNextFrame += 2;
- } else if (video_clock > audio_clock + VideoAudioDelay + 30 * 90) {
+ } else if (video_clock > audio_clock + VideoAudioDelay + 45 * 90) {
Debug(3, "video: slow down video\n");
decoder->DupNextFrame++;
- } else if (audio_clock + VideoAudioDelay > video_clock + 40 * 90
+ } else if (audio_clock + VideoAudioDelay > video_clock + 15 * 90
&& filled > 1) {
Debug(3, "video: speed up video\n");
decoder->DropNextFrame = 1;
}
- } else if (audio_clock == (int64_t) AV_NOPTS_VALUE
- && video_clock != (int64_t) AV_NOPTS_VALUE) {
- if (VideoGetBuffers() < 4) {
- Debug(3, "video: initial slow down video\n");
- decoder->DupNextFrame++;
- }
}
#if defined(DEBUG) || defined(AV_INFO)
// debug audio/video sync
@@ -4892,6 +4903,7 @@ typedef struct _vdpau_decoder_
struct timespec FrameTime; ///< time of last display
int64_t PTS; ///< video PTS clock
+ int StartCounter; ///< number of start frames
int FramesDuped; ///< number of frames duplicated
int FramesMissed; ///< number of frames missed
int FramesDropped; ///< number of frames dropped
@@ -5568,6 +5580,8 @@ static void VdpauCleanup(VdpauDecoder * decoder)
decoder->SurfaceField = 0;
+ //decoder->FrameCounter = 0;
+ decoder->StartCounter = 0;
decoder->PTS = AV_NOPTS_VALUE;
VideoDeltaPTS = 0;
}
@@ -7431,6 +7445,14 @@ static void VdpauSyncDisplayFrame(VdpauDecoder * decoder)
filled = atomic_read(&decoder->SurfacesFilled);
// FIXME: audio not known assume 333ms delay
+ decoder->StartCounter++;
+ if (!VideoSoftStartSync && decoder->StartCounter < 60
+ && (audio_clock == (int64_t) AV_NOPTS_VALUE
+ || video_clock > audio_clock + VideoAudioDelay + 120 * 90)) {
+ Debug(3, "video: initial slow down %d\n", decoder->StartCounter);
+ decoder->DupNextFrame = 2;
+ }
+
if (decoder->DupNextFrame) {
decoder->DupNextFrame--;
++decoder->FramesDuped;
@@ -7440,23 +7462,17 @@ static void VdpauSyncDisplayFrame(VdpauDecoder * decoder)
if (abs(video_clock - audio_clock) > 5000 * 90) {
Debug(3, "video: pts difference too big\n");
- } else if (video_clock > audio_clock + VideoAudioDelay + 80 * 90) {
+ } else if (video_clock > audio_clock + VideoAudioDelay + 100 * 90) {
Debug(3, "video: slow down video\n");
decoder->DupNextFrame += 2;
- } else if (video_clock > audio_clock + VideoAudioDelay + 30 * 90) {
+ } else if (video_clock > audio_clock + VideoAudioDelay + 45 * 90) {
Debug(3, "video: slow down video\n");
decoder->DupNextFrame++;
- } else if (audio_clock + VideoAudioDelay > video_clock + 40 * 90
+ } else if (audio_clock + VideoAudioDelay > video_clock + 15 * 90
&& filled > 1 + 2 * decoder->Interlaced) {
Debug(3, "video: speed up video\n");
decoder->DropNextFrame = 1;
}
- } else if (audio_clock == (int64_t) AV_NOPTS_VALUE
- && video_clock != (int64_t) AV_NOPTS_VALUE) {
- if (VideoGetBuffers() < 4) {
- Debug(3, "video: initial slow down video\n");
- decoder->DupNextFrame++;
- }
}
#if defined(DEBUG) || defined(AV_INFO)
// debug audio/video sync
@@ -7483,14 +7499,9 @@ static void VdpauSyncRenderFrame(VdpauDecoder * decoder,
{
VideoSetPts(&decoder->PTS, decoder->Interlaced, frame);
- if (VdpauPreemption) { // display preempted
- return;
- }
-#ifdef DEBUG
if (!atomic_read(&decoder->SurfacesFilled)) {
Debug(3, "video: new stream frame %d\n", GetMsTicks() - VideoSwitch);
}
-#endif
if (decoder->DropNextFrame) { // drop frame requested
++decoder->FramesDropped;
@@ -7502,6 +7513,9 @@ static void VdpauSyncRenderFrame(VdpauDecoder * decoder,
decoder->DropNextFrame--;
return;
}
+ if (VdpauPreemption) { // display preempted
+ return;
+ }
// if video output buffer is full, wait and display surface.
// loop for interlace
while (atomic_read(&decoder->SurfacesFilled) >= VIDEO_SURFACES_MAX) {
@@ -8618,6 +8632,7 @@ void VideoReleaseSurface(VideoHwDecoder * decoder, unsigned surface)
enum PixelFormat Video_get_format(VideoHwDecoder * decoder,
AVCodecContext * video_ctx, const enum PixelFormat *fmt)
{
+ AudioVideoReady();
if (VideoUsedModule) {
return VideoUsedModule->get_format(decoder, video_ctx, fmt);
}