mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
Detect audio stream type only after stream switch.
This commit is contained in:
Johns
parent
baa4500a2c
commit
a91533f6d1
@ -1,6 +1,7 @@
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User johns
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Date:
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Detect audio stream type only after stream switch.
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Detect more h264 streams with leading zeros.
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VDPAU: support for studio levels added.
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Add support for skip chroma deinterlace to software deinterlacer.
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@ -156,6 +156,11 @@ Setup: /etc/vdr/setup.conf
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softhddevice.Suspend.X11 = 0
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1 suspend stops X11 server (not working yet)
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VideoDisplayFormat = ?
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0 pan and scan
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1 letter box
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2 center cut-out
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Setup: /etc/vdr/remote.conf
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------
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4
Todo
4
Todo
@ -40,6 +40,7 @@ video:
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OSD can only be shown after some stream could be shown
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yaepghd changed position is lost on channel switch
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use x11 screen size without geometry configuration
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pause (live tv) has sometime problems with SAT1 HD Pro7 HD
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vdpau:
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software decoder path not working
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@ -83,10 +84,9 @@ audio:
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selectable.
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software volume support
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add pause support for replay pause
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some plugin send a private LPCM stream (CODEC_ID_PCM_DVD), which
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isn't supported by ffmpeg, write own parser.
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Mute should do a real mute and not only set volume to zero.
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Starting suspended and muted, didn't register the mute.
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switching stero/dolby produces klick sound
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audio/alsa:
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better downmix of >2 channels on 2 channel hardware
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8
audio.c
8
audio.c
@ -2070,13 +2070,7 @@ int64_t AudioGetClock(void)
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*/
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void AudioSetVolume(int volume)
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{
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#ifdef USE_ALSA
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AlsaSetVolume(volume);
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#endif
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#ifdef USE_OSS
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OssSetVolume(volume);
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#endif
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(void)volume;
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return AudioUsedModule->SetVolume(volume);
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}
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/**
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2
codec.c
2
codec.c
@ -877,7 +877,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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int err;
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int isAC3;
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// FIXME: use swr_convert from swresample
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// FIXME: use swr_convert from swresample (only in ffmpeg!)
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// FIXME: tell ac3 decoder to use downmix
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if (audio_decoder->ReSample) {
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audio_resample_close(audio_decoder->ReSample);
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159
softhddev.c
159
softhddev.c
@ -76,6 +76,7 @@ static volatile char NewAudioStream; ///< new audio stream
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static volatile char SkipAudio; ///< skip audio stream
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static AudioDecoder *MyAudioDecoder; ///< audio decoder
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static enum CodecID AudioCodecID; ///< current codec id
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static int AudioChannelID; ///< current audio channel id
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/**
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** mpeg bitrate table.
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@ -202,8 +203,7 @@ static int FindAudioSync(const AVPacket * avpkt)
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** @param size size of PES packet
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** @param id PES packet type
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*/
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int PlayAudio(const uint8_t * data, int size,
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__attribute__ ((unused)) uint8_t id)
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int PlayAudio(const uint8_t * data, int size, uint8_t id)
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{
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int n;
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int osize;
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@ -263,87 +263,92 @@ int PlayAudio(const uint8_t * data, int size,
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// Detect audio code
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// MPEG-PS mp2 MPEG1, MPEG2, AC3, LPCM, AAC LATM
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// Syncword - 0x0B77
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if (data[0] == 0x0B && data[1] == 0x77) {
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if (AudioCodecID != CODEC_ID_AC3) {
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Debug(3, "[softhddev]%s: AC-3 %d\n", __FUNCTION__, id);
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CodecAudioClose(MyAudioDecoder);
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CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_AC3);
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AudioCodecID = CODEC_ID_AC3;
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}
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// Syncword - 0xFFFC - 0xFFFF
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} else if (data[0] == 0xFF && (data[1] & 0xFC) == 0xFC) {
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if (AudioCodecID != CODEC_ID_MP2) {
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Debug(3, "[softhddev]%s: MP2 %d\n", __FUNCTION__, id);
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CodecAudioClose(MyAudioDecoder);
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CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
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AudioCodecID = CODEC_ID_MP2;
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}
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// latm header 0x56E0 11bits: 0x2B7
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} else if (data[0] == 0x56 && (data[1] & 0xE0) == 0xE0) {
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if (AudioCodecID != CODEC_ID_AAC_LATM) {
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Debug(3, "[softhddev]%s: AAC LATM %d\n", __FUNCTION__, id);
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CodecAudioClose(MyAudioDecoder);
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CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_AAC_LATM);
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AudioCodecID = CODEC_ID_AAC_LATM;
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}
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// Private stream + LPCM ID
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} else if (data[-n - 9 + 3] == 0xBD && data[0] == 0xA0) {
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if (AudioCodecID != CODEC_ID_PCM_DVD) {
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static int samplerates[] = { 48000, 96000, 44100, 32000 };
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int samplerate;
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int channels;
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int bits_per_sample;
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Debug(3, "[softhddev]%s: LPCM %d sr:%d bits:%d chan:%d\n",
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__FUNCTION__, id, data[5] >> 4,
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(((data[5] >> 6) & 0x3) + 4) * 4, (data[5] & 0x7) + 1);
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CodecAudioClose(MyAudioDecoder);
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bits_per_sample = (((data[5] >> 6) & 0x3) + 4) * 4;
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if (bits_per_sample != 16) {
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Error(_
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("softhddev: LPCM %d bits per sample aren't supported\n"),
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bits_per_sample);
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// FIXME: handle unsupported formats.
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// only if unknown or new channel id
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if (AudioCodecID == CODEC_ID_NONE || AudioChannelID != id) {
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AudioChannelID = id;
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// Syncword - 0x0B77
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if (data[0] == 0x0B && data[1] == 0x77) {
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if (AudioCodecID != CODEC_ID_AC3) {
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Debug(3, "[softhddev]%s: AC-3 %d\n", __FUNCTION__, id);
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CodecAudioClose(MyAudioDecoder);
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CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_AC3);
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AudioCodecID = CODEC_ID_AC3;
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}
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samplerate = samplerates[data[5] >> 4];
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channels = (data[5] & 0x7) + 1;
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AudioSetup(&samplerate, &channels, 0);
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if (samplerate != samplerates[data[5] >> 4]) {
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Error(_("softhddev: LPCM %d sample-rate is unsupported\n"),
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samplerates[data[5] >> 4]);
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// FIXME: support resample
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// Syncword - 0xFFFC - 0xFFFF
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} else if (data[0] == 0xFF && (data[1] & 0xFC) == 0xFC) {
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if (AudioCodecID != CODEC_ID_MP2) {
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Debug(3, "[softhddev]%s: MP2 %d\n", __FUNCTION__, id);
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CodecAudioClose(MyAudioDecoder);
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CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
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AudioCodecID = CODEC_ID_MP2;
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}
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if (channels != (data[5] & 0x7) + 1) {
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Error(_("softhddev: LPCM %d channels are unsupported\n"),
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(data[5] & 0x7) + 1);
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// FIXME: support resample
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// latm header 0x56E0 11bits: 0x2B7
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} else if (data[0] == 0x56 && (data[1] & 0xE0) == 0xE0) {
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// && (((data[1] & 0x1F) << 8) + (data[2] & 0xFF)) < size
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if (AudioCodecID != CODEC_ID_AAC_LATM) {
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Debug(3, "[softhddev]%s: AAC LATM %d\n", __FUNCTION__, id);
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CodecAudioClose(MyAudioDecoder);
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CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_AAC_LATM);
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AudioCodecID = CODEC_ID_AAC_LATM;
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}
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//CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_PCM_DVD);
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AudioCodecID = CODEC_ID_PCM_DVD;
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}
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} else {
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// no start package
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// FIXME: Nick/Viva sends this shit, need to find sync in packet
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// FIXME: otherwise it takes too long until sound appears
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// Private stream + LPCM ID
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} else if (data[-n - 9 + 3] == 0xBD && data[0] == 0xA0) {
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if (AudioCodecID != CODEC_ID_PCM_DVD) {
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static int samplerates[] = { 48000, 96000, 44100, 32000 };
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int samplerate;
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int channels;
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int bits_per_sample;
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if (AudioCodecID == CODEC_ID_NONE) {
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Debug(3, "[softhddev]%s: ??? %d\n", __FUNCTION__, id);
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avpkt->data = (void *)data;
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avpkt->size = size;
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n = FindAudioSync(avpkt);
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if (n < 0) {
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return osize;
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}
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Debug(3, "[softhddev]%s: LPCM %d sr:%d bits:%d chan:%d\n",
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__FUNCTION__, id, data[5] >> 4,
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(((data[5] >> 6) & 0x3) + 4) * 4, (data[5] & 0x7) + 1);
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CodecAudioClose(MyAudioDecoder);
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avpkt->pts = AV_NOPTS_VALUE;
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CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
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AudioCodecID = CODEC_ID_MP2;
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data += n;
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size -= n;
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bits_per_sample = (((data[5] >> 6) & 0x3) + 4) * 4;
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if (bits_per_sample != 16) {
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Error(_
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("softhddev: LPCM %d bits per sample aren't supported\n"),
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bits_per_sample);
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// FIXME: handle unsupported formats.
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}
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samplerate = samplerates[data[5] >> 4];
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channels = (data[5] & 0x7) + 1;
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AudioSetup(&samplerate, &channels, 0);
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if (samplerate != samplerates[data[5] >> 4]) {
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Error(_("softhddev: LPCM %d sample-rate is unsupported\n"),
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samplerates[data[5] >> 4]);
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// FIXME: support resample
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}
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if (channels != (data[5] & 0x7) + 1) {
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Error(_("softhddev: LPCM %d channels are unsupported\n"),
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(data[5] & 0x7) + 1);
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// FIXME: support resample
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}
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//CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_PCM_DVD);
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AudioCodecID = CODEC_ID_PCM_DVD;
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}
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} else {
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// no start package
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// FIXME: Nick/Viva sends this shit, need to find sync in packet
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// FIXME: otherwise it takes too long until sound appears
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if (AudioCodecID == CODEC_ID_NONE) {
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Debug(3, "[softhddev]%s: ??? %d\n", __FUNCTION__, id);
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avpkt->data = (void *)data;
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avpkt->size = size;
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n = FindAudioSync(avpkt);
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if (1 && n < 0) {
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return osize;
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}
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avpkt->pts = AV_NOPTS_VALUE;
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CodecAudioOpen(MyAudioDecoder, NULL, CODEC_ID_MP2);
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AudioCodecID = CODEC_ID_MP2;
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data += n;
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size -= n;
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}
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}
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// no decoder or codec known
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// still no decoder or codec known
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if (AudioCodecID == CODEC_ID_NONE) {
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return osize;
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}
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