mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
Adds audio normalize support.
This commit is contained in:
Johns
parent
8039e8ae04
commit
44ca71fedb
31
README.txt
31
README.txt
@ -157,27 +157,28 @@ Setup: /etc/vdr/setup.conf
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0 = none, 1 = downmix
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downmix AC-3 to stero.
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softhddevice.AudioSoftvol
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FIXME:
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softhddevice.AudioSoftvol = 0
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0 = off, use hardware volume control
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1 = on, use software volume control
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softhddevice.AudioNormalize
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FIXME:
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softhddevice.AudioNormalize = 0
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0 = off, 1 = enable audio normalize
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softhddevice.AudioMaxNormalize
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FIXME:
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softhddevice.AudioMaxNormalize = 0
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maximal volume factor/1000 of the normalize filter
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softhddevice.AudioCompression
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FIXME:
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softhddevice.AudioCompression = 0
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0 = off, 1 = enable audio compression
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softhddevice.AudioMaxCompression
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FIXME:
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softhddevice.AudioMaxCompression = 0
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maximal volume factor/1000 of the compression filter
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softhddevice.AudioStereoDescent
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FIXME:
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softhddevice.AudioBufferTime
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FIXME:
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softhddevice.AudioStereoDescent = 0
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reduce volume level (/1000) for stereo sources
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softhddevice.AudioBufferTime = 0
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0 = default (336 ms)
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1 - 1000 = size of the buffer in ms
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softhddevice.AutoCrop.Interval = 0
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0 disables auto-crop
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86
audio.c
86
audio.c
@ -48,6 +48,7 @@
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#include <stdlib.h>
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#include <inttypes.h>
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#include <string.h>
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#include <math.h>
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#include <libintl.h>
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#define _(str) gettext(str) ///< gettext shortcut
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@ -163,6 +164,8 @@ static char AudioNormalize; ///< flag use volume normalize
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static char AudioCompression; ///< flag use compress volume
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static char AudioMute; ///< flag muted
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static int AudioAmplifier; ///< software volume factor
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static int AudioNormalizeFactor; ///< current normalize factor
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static const int AudioMinNormalize = 100; ///< min. normalize factor
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static int AudioMaxNormalize; ///< max. normalize factor
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static int AudioCompressionFactor; ///< current compression factor
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static int AudioMaxCompression; ///< max. compression factor
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@ -205,12 +208,13 @@ static const unsigned AudioRatesTable[AudioRatesMax] = {
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// filter
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//----------------------------------------------------------------------------
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static const int AudioNormSamples = 32768; ///< number of samples
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static const int AudioNormSamples = 4096; ///< number of samples
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#define AudioNormIndexes 128 ///< number of average values
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#define AudioNormMaxIndex 128 ///< number of average values
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/// average of n last sample blocks
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static uint32_t AudioNormAverage[AudioNormIndexes];
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static uint32_t AudioNormAverage[AudioNormMaxIndex];
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static int AudioNormIndex; ///< index into average table
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static int AudioNormReady; ///< index counter
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static int AudioNormCounter; ///< sample counter
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/**
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@ -222,21 +226,77 @@ static int AudioNormCounter; ///< sample counter
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static void AudioNormalizer(int16_t * samples, int count)
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{
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int i;
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int l;
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int n;
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uint32_t avg;
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int factor;
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int16_t *data;
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// average samples
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avg = 0;
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n = count / AudioBytesProSample;
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l = count / AudioBytesProSample;
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data = samples;
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do {
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n = l;
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if (AudioNormCounter + n > AudioNormSamples) {
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n = AudioNormSamples - AudioNormCounter;
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}
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avg = AudioNormAverage[AudioNormIndex];
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for (i = 0; i < n; ++i) {
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int t;
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t = samples[i];
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avg += t * t;
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avg /= 2;
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t = data[i];
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avg += (t * t) / AudioNormSamples;
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}
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AudioNormAverage[AudioNormIndex] = avg;
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AudioNormCounter += n;
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if (AudioNormCounter >= AudioNormSamples) {
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if (AudioNormReady < AudioNormMaxIndex) {
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AudioNormReady++;
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} else {
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avg = 0;
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for (i = 0; i < AudioNormMaxIndex; ++i) {
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avg += AudioNormAverage[i] / AudioNormMaxIndex;
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}
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// FIXME: more todo
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// calculate normalize factor
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if (avg > 0) {
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factor = ((INT16_MAX / 8) * 1000U) / (uint32_t) sqrt(avg);
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// smooth normalize
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AudioNormalizeFactor =
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(AudioNormalizeFactor * 500 + factor * 500) / 1000;
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if (AudioNormalizeFactor < AudioMinNormalize) {
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AudioNormalizeFactor = AudioMinNormalize;
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}
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if (AudioNormalizeFactor > AudioMaxNormalize) {
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AudioNormalizeFactor = AudioMaxNormalize;
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}
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} else {
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factor = 1000;
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}
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Debug(4, "audio/noramlize: avg %8d, fac=%6.3f, norm=%6.3f\n",
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avg, factor / 1000.0, AudioNormalizeFactor / 1000.0);
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}
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AudioNormIndex = (AudioNormIndex + 1) % AudioNormMaxIndex;
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AudioNormCounter = 0;
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AudioNormAverage[AudioNormIndex] = 0U;
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}
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data += n;
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l -= n;
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} while (l > 0);
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// apply normalize factor
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for (i = 0; i < count / AudioBytesProSample; ++i) {
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int t;
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t = (samples[i] * AudioNormalizeFactor) / 1000;
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if (t < INT16_MIN) {
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t = INT16_MIN;
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} else if (t > INT16_MAX) {
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t = INT16_MAX;
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}
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samples[i] = t;
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}
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}
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/**
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@ -244,6 +304,14 @@ static void AudioNormalizer(int16_t * samples, int count)
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*/
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static void AudioResetNormalizer(void)
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{
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int i;
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AudioNormCounter = 0;
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AudioNormReady = 0;
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for (i = 0; i < AudioNormMaxIndex; ++i) {
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AudioNormAverage[i] = 0U;
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}
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AudioNormalizeFactor = 1000;
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}
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/**
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@ -690,9 +690,9 @@ void cMenuSetupSoft::Create(void)
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Add(new cMenuEditBoolItem(tr("Volume control"), &AudioSoftvol,
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tr("Hardware"), tr("Software")));
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Add(new cMenuEditBoolItem(tr("Enable normalize volume"),
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&AudioMaxNormalize, trVDR("no"), trVDR("yes")));
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&AudioNormalize, trVDR("no"), trVDR("yes")));
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Add(new cMenuEditIntItem(tr(" Max normalize factor (/1000)"),
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&AudioMaxNormalize, 0, 5000));
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&AudioMaxNormalize, 0, 10000));
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Add(new cMenuEditBoolItem(tr("Enable volume compression"),
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&AudioCompression, trVDR("no"), trVDR("yes")));
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Add(new cMenuEditIntItem(tr(" Max compression factor (/1000)"),
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2
video.c
2
video.c
@ -1857,6 +1857,7 @@ static void VaapiCleanup(VaapiDecoder * decoder)
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decoder->SurfaceWrite = 0;
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decoder->SurfaceField = 0;
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decoder->SyncCounter = 0;
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decoder->FrameCounter = 0;
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decoder->FramesDisplayed = 0;
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decoder->Closing = 0;
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@ -5736,6 +5737,7 @@ static void VdpauCleanup(VdpauDecoder * decoder)
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decoder->SurfaceWrite = 0;
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decoder->SurfaceField = 0;
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decoder->SyncCounter = 0;
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decoder->FrameCounter = 0;
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decoder->FramesDisplayed = 0;
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decoder->Closing = 0;
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