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vdr-plugin-softhddevice
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Code Issues Releases Wiki Activity

Detect audio stream type only after stream switch.

This commit is contained in:
Johns 2012-02-11 18:22:48 +01:00
parent baa4500a2c
commit a91533f6d1
6 changed files with 92 additions and 87 deletions
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1
ChangeLog
View File

@ -1,6 +1,7 @@
User johns
Date:
Detect audio stream type only after stream switch.
Detect more h264 streams with leading zeros.
VDPAU: support for studio levels added.
Add support for skip chroma deinterlace to software deinterlacer.

5
README.txt
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@ -156,6 +156,11 @@ Setup: /etc/vdr/setup.conf
softhddevice.Suspend.X11 = 0
1 suspend stops X11 server (not working yet)
VideoDisplayFormat = ?
0 pan and scan
1 letter box
2 center cut-out
Setup: /etc/vdr/remote.conf
------

4
Todo
View File

@ -40,6 +40,7 @@ video:
OSD can only be shown after some stream could be shown
yaepghd changed position is lost on channel switch
use x11 screen size without geometry configuration
pause (live tv) has sometime problems with SAT1 HD Pro7 HD
vdpau:
software decoder path not working
@ -83,10 +84,9 @@ audio:
selectable.
software volume support
add pause support for replay pause
some plugin send a private LPCM stream (CODEC_ID_PCM_DVD), which
isn't supported by ffmpeg, write own parser.
Mute should do a real mute and not only set volume to zero.
Starting suspended and muted, didn't register the mute.
switching stero/dolby produces klick sound
audio/alsa:
better downmix of >2 channels on 2 channel hardware

8
audio.c
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@ -2070,13 +2070,7 @@ int64_t AudioGetClock(void)
*/
void AudioSetVolume(int volume)
{
#ifdef USE_ALSA
AlsaSetVolume(volume);
#endif
#ifdef USE_OSS
OssSetVolume(volume);
#endif
(void)volume;
return AudioUsedModule->SetVolume(volume);
}
/**

2
codec.c
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@ -877,7 +877,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
int err;
int isAC3;
// FIXME: use swr_convert from swresample
// FIXME: use swr_convert from swresample (only in ffmpeg!)
// FIXME: tell ac3 decoder to use downmix
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);

159
softhddev.c
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@ -76,6 +76,7 @@ static volatile char NewAudioStream; ///< new audio stream
static volatile char SkipAudio; ///< skip audio stream
static AudioDecoder *MyAudioDecoder; ///< audio decoder
static enum CodecID AudioCodecID; ///< current codec id
static int AudioChannelID; ///< current audio channel id
/**
** mpeg bitrate table.
@ -202,8 +203,7 @@ static int FindAudioSync(const AVPacket * avpkt)
** @param size size of PES packet
** @param id PES packet type
*/
int PlayAudio(const uint8_t * data, int size,
__attribute__ ((unused)) uint8_t id)
int PlayAudio(const uint8_t * data, int size, uint8_t id)
{
int n;
int osize;
@ -263,87 +263,92 @@ int PlayAudio(const uint8_t * data, int size,
// Detect audio code
// MPEG-PS mp2 MPEG1, MPEG2, AC3, LPCM, AAC LATM
// Syncword - 0x0B77
if (data[0] == 0x0B && data[1] == 0x77) {
if (AudioCodecID != CODEC_ID_AC3) {
Debug(3, "[softhddev]%s: AC-3 %d\n", __FUNCTION__, id);
CodecAudioClose(MyAudioDecoder);
CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_AC3);
AudioCodecID = CODEC_ID_AC3;
}
// Syncword - 0xFFFC - 0xFFFF
} else if (data[0] == 0xFF && (data[1] & 0xFC) == 0xFC) {
if (AudioCodecID != CODEC_ID_MP2) {
Debug(3, "[softhddev]%s: MP2 %d\n", __FUNCTION__, id);
CodecAudioClose(MyAudioDecoder);
CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
AudioCodecID = CODEC_ID_MP2;
}
// latm header 0x56E0 11bits: 0x2B7
} else if (data[0] == 0x56 && (data[1] & 0xE0) == 0xE0) {
if (AudioCodecID != CODEC_ID_AAC_LATM) {
Debug(3, "[softhddev]%s: AAC LATM %d\n", __FUNCTION__, id);
CodecAudioClose(MyAudioDecoder);
CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_AAC_LATM);
AudioCodecID = CODEC_ID_AAC_LATM;
}
// Private stream + LPCM ID
} else if (data[-n - 9 + 3] == 0xBD && data[0] == 0xA0) {
if (AudioCodecID != CODEC_ID_PCM_DVD) {
static int samplerates[] = { 48000, 96000, 44100, 32000 };
int samplerate;
int channels;
int bits_per_sample;
Debug(3, "[softhddev]%s: LPCM %d sr:%d bits:%d chan:%d\n",
__FUNCTION__, id, data[5] >> 4,
(((data[5] >> 6) & 0x3) + 4) * 4, (data[5] & 0x7) + 1);
CodecAudioClose(MyAudioDecoder);
bits_per_sample = (((data[5] >> 6) & 0x3) + 4) * 4;
if (bits_per_sample != 16) {
Error(_
("softhddev: LPCM %d bits per sample aren't supported\n"),
bits_per_sample);
// FIXME: handle unsupported formats.
// only if unknown or new channel id
if (AudioCodecID == CODEC_ID_NONE || AudioChannelID != id) {
AudioChannelID = id;
// Syncword - 0x0B77
if (data[0] == 0x0B && data[1] == 0x77) {
if (AudioCodecID != CODEC_ID_AC3) {
Debug(3, "[softhddev]%s: AC-3 %d\n", __FUNCTION__, id);
CodecAudioClose(MyAudioDecoder);
CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_AC3);
AudioCodecID = CODEC_ID_AC3;
}
samplerate = samplerates[data[5] >> 4];
channels = (data[5] & 0x7) + 1;
AudioSetup(&samplerate, &channels, 0);
if (samplerate != samplerates[data[5] >> 4]) {
Error(_("softhddev: LPCM %d sample-rate is unsupported\n"),
samplerates[data[5] >> 4]);
// FIXME: support resample
// Syncword - 0xFFFC - 0xFFFF
} else if (data[0] == 0xFF && (data[1] & 0xFC) == 0xFC) {
if (AudioCodecID != CODEC_ID_MP2) {
Debug(3, "[softhddev]%s: MP2 %d\n", __FUNCTION__, id);
CodecAudioClose(MyAudioDecoder);
CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
AudioCodecID = CODEC_ID_MP2;
}
if (channels != (data[5] & 0x7) + 1) {
Error(_("softhddev: LPCM %d channels are unsupported\n"),
(data[5] & 0x7) + 1);
// FIXME: support resample
// latm header 0x56E0 11bits: 0x2B7
} else if (data[0] == 0x56 && (data[1] & 0xE0) == 0xE0) {
// && (((data[1] & 0x1F) << 8) + (data[2] & 0xFF)) < size
if (AudioCodecID != CODEC_ID_AAC_LATM) {
Debug(3, "[softhddev]%s: AAC LATM %d\n", __FUNCTION__, id);
CodecAudioClose(MyAudioDecoder);
CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_AAC_LATM);
AudioCodecID = CODEC_ID_AAC_LATM;
}
//CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_PCM_DVD);
AudioCodecID = CODEC_ID_PCM_DVD;
}
} else {
// no start package
// FIXME: Nick/Viva sends this shit, need to find sync in packet
// FIXME: otherwise it takes too long until sound appears
// Private stream + LPCM ID
} else if (data[-n - 9 + 3] == 0xBD && data[0] == 0xA0) {
if (AudioCodecID != CODEC_ID_PCM_DVD) {
static int samplerates[] = { 48000, 96000, 44100, 32000 };
int samplerate;
int channels;
int bits_per_sample;
if (AudioCodecID == CODEC_ID_NONE) {
Debug(3, "[softhddev]%s: ??? %d\n", __FUNCTION__, id);
avpkt->data = (void *)data;
avpkt->size = size;
n = FindAudioSync(avpkt);
if (n < 0) {
return osize;
}
Debug(3, "[softhddev]%s: LPCM %d sr:%d bits:%d chan:%d\n",
__FUNCTION__, id, data[5] >> 4,
(((data[5] >> 6) & 0x3) + 4) * 4, (data[5] & 0x7) + 1);
CodecAudioClose(MyAudioDecoder);
avpkt->pts = AV_NOPTS_VALUE;
CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
AudioCodecID = CODEC_ID_MP2;
data += n;
size -= n;
bits_per_sample = (((data[5] >> 6) & 0x3) + 4) * 4;
if (bits_per_sample != 16) {
Error(_
("softhddev: LPCM %d bits per sample aren't supported\n"),
bits_per_sample);
// FIXME: handle unsupported formats.
}
samplerate = samplerates[data[5] >> 4];
channels = (data[5] & 0x7) + 1;
AudioSetup(&samplerate, &channels, 0);
if (samplerate != samplerates[data[5] >> 4]) {
Error(_("softhddev: LPCM %d sample-rate is unsupported\n"),
samplerates[data[5] >> 4]);
// FIXME: support resample
}
if (channels != (data[5] & 0x7) + 1) {
Error(_("softhddev: LPCM %d channels are unsupported\n"),
(data[5] & 0x7) + 1);
// FIXME: support resample
}
//CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_PCM_DVD);
AudioCodecID = CODEC_ID_PCM_DVD;
}
} else {
// no start package
// FIXME: Nick/Viva sends this shit, need to find sync in packet
// FIXME: otherwise it takes too long until sound appears
if (AudioCodecID == CODEC_ID_NONE) {
Debug(3, "[softhddev]%s: ??? %d\n", __FUNCTION__, id);
avpkt->data = (void *)data;
avpkt->size = size;
n = FindAudioSync(avpkt);
if (1 && n < 0) {
return osize;
}
avpkt->pts = AV_NOPTS_VALUE;
CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
AudioCodecID = CODEC_ID_MP2;
data += n;
size -= n;
}
}
// no decoder or codec known
// still no decoder or codec known
if (AudioCodecID == CODEC_ID_NONE) {
return osize;
}