Reduces audio latency, increases audio buffer time.

6 files changed, 19 insertions, 29 deletions

diff --git a/ChangeLog b/ChangeLog
index 391cbc5..140e85a 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,6 +1,7 @@
 User johns
 Date:
 
+    Reduces audio latency, increases audio buffer time.
     Made video_test working again.
     Disabled VA-API Intel vaAssociateSubpicture workaround.
     Fix bug: Must release lock for VideoPollEvent.
diff --git a/Makefile b/Makefile
index deed91f..257dae0 100644
--- a/Makefile
+++ b/Makefile
@@ -18,7 +18,7 @@ GIT_REV = $(shell git describe --always 2>/dev/null)
 
 ### Configuration (edit this for your needs)
 
-CONFIG := -DDEBUG
+CONFIG := #-DDEBUG
 #CONFIG += -DHAVE_PTHREAD_NAME
 CONFIG += $(shell pkg-config --exists vdpau && echo "-DUSE_VDPAU")
 CONFIG += $(shell pkg-config --exists libva && echo "-DUSE_VAAPI")
diff --git a/audio.c b/audio.c
index 2142b5a..6c41503 100644
--- a/audio.c
+++ b/audio.c
@@ -137,8 +137,7 @@ 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 = 300;	///< audio buffer time in ms
-static int AudioMoreBufferTime = 1;	///< increase buffer time
+static const int AudioBufferTime = 450;	///< audio buffer time in ms
 
 #ifdef USE_AUDIO_THREAD
 static pthread_t AudioThread;		///< audio play thread
@@ -1087,13 +1086,11 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
     AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
     // buffer time/delay in ms
     if (AlsaStartThreshold <
-	(*freq * *channels * AudioBytesProSample * AudioMoreBufferTime *
-	    AudioBufferTime) / 1000U) {
+	(*freq * *channels * AudioBytesProSample * AudioBufferTime) / 1000U) {
 	AlsaStartThreshold =
-	    (*freq * *channels * AudioBytesProSample * AudioMoreBufferTime *
-	    AudioBufferTime) / 1000U;
+	    (*freq * *channels * AudioBytesProSample * AudioBufferTime) /
+	    1000U;
     }
-    AudioMoreBufferTime = 1;
     // no bigger, than the buffer
     if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) {
 	AlsaStartThreshold = RingBufferFreeBytes(AlsaRingBuffer);
@@ -1712,13 +1709,12 @@ static int OssSetup(int *freq, int *channels, int use_ac3)
 	OssStartThreshold = bi.bytes + tmp;
 	// buffer time/delay in ms
 	if (OssStartThreshold <
-	    (*freq * *channels * AudioBytesProSample * AudioMoreBufferTime *
-		AudioBufferTime) / 1000U) {
+	    (*freq * *channels * AudioBytesProSample * AudioBufferTime) /
+	    1000U) {
 	    OssStartThreshold =
-		(*freq * *channels * AudioBytesProSample *
-		AudioMoreBufferTime * AudioBufferTime) / 1000U;
+		(*freq * *channels * AudioBytesProSample * AudioBufferTime) /
+		1000U;
 	}
-	AudioMoreBufferTime = 1;
 	// no bigger, than the buffer
 	if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) {
 	    OssStartThreshold = RingBufferFreeBytes(OssRingBuffer);
@@ -2102,16 +2098,6 @@ int AudioSetup(int *freq, int *channels, int use_ac3)
 }
 
 /**
-**	Increase audio buffer time.
-**
-**	Some channels need a bigger audio buffer to buffer video.
-*/
-void AudioIncreaseBufferTime(void)
-{
-    AudioMoreBufferTime = 4;
-}
-
-/**
 **	Set pcm audio device.
 **
 **	@param device	name of pcm device (fe. "hw:0,9" or "/dev/dsp")
diff --git a/audio.h b/audio.h
index 8924159..2a4c45f 100644
--- a/audio.h
+++ b/audio.h
@@ -42,8 +42,6 @@ extern int AudioSetup(int *, int *, int);	///< setup audio output
 //extern void AudioPlay(void);		///< play audio
 //extern void AudioPause(void);		///< pause audio
 
-extern void AudioIncreaseBufferTime(void);	///< use bigger buffer
-
 extern void AudioSetDevice(const char *);	///< set PCM audio device
 extern void AudioSetDeviceAC3(const char *);	///< set Passthrough device
 extern void AudioInit(void);		///< setup audio module
diff --git a/softhddev.c b/softhddev.c
index 5abdd11..60d70c9 100644
--- a/softhddev.c
+++ b/softhddev.c
@@ -292,7 +292,6 @@ int PlayAudio(const uint8_t * data, int size,
 	    }
 
 	    avpkt->pts = AV_NOPTS_VALUE;
-	    AudioIncreaseBufferTime();
 	    CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
 	    AudioCodecID = CODEC_ID_MP2;
 	    data += n;
@@ -794,11 +793,15 @@ int PlayVideo(const uint8_t * data, int size)
 	    Debug(3, "video: not detected\n");
 	    return size;
 	}
-	// FIXME: incomplete packets produce artefacts after channel switch
-	if (0 && VideoCodecID == CODEC_ID_MPEG2VIDEO) {
+	// incomplete packets produce artefacts after channel switch
+	// packet < 65526 is the last split packet, detect it here for
+	// better latency
+	if (size < 65526 && VideoCodecID == CODEC_ID_MPEG2VIDEO) {
 	    // mpeg codec supports incomplete packets
 	    // waiting for a full complete packages, increases needed delays
+	    VideoEnqueue(pts, check, size - 9 - n);
 	    VideoNextPacket(CODEC_ID_MPEG2VIDEO);
+	    return size;
 	}
     }
 
diff --git a/video.c b/video.c
index 6e62d5c..bf946e0 100644
--- a/video.c
+++ b/video.c
@@ -3471,7 +3471,8 @@ static void VaapiAdvanceFrame(void)
 	    }
 	    // debug duplicate frames
 	} else if (filled == 1) {
-	    decoder->FramesDuped++;
+	    ++decoder->FramesDuped;
+	    decoder->DropNextFrame = 0;
 	    Warning(_
 		("video: display buffer empty, duping frame (%d/%d) %d\n"),
 		decoder->FramesDuped, decoder->FrameCounter,
@@ -6423,6 +6424,7 @@ static void VdpauAdvanceFrame(void)
 	    if (filled <= 1 + 2 * decoder->Interlaced) {
 		// keep use of last surface
 		++decoder->FramesDuped;
+		decoder->DropNextFrame = 0;
 		Warning(_
 		    ("video: display buffer empty, duping frame (%d/%d) %d\n"),
 		    decoder->FramesDuped, decoder->FrameCounter,