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vdr-plugin-softhddevice
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Code Issues Releases Wiki Activity

Cleanups.

This commit is contained in:
Johns 2012-03-12 17:58:19 +01:00
parent 09ba3e2993
commit 7b570c507c
5 changed files with 101 additions and 70 deletions
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4
README.txt
View File

@ -166,7 +166,9 @@ Setup: /etc/vdr/setup.conf
32bit RGBA background color
(Red * 16777216 + Green * 65536 + Blue * 256 + Alpha)
or hex RRGGBBAA
grey = 2155905279
grey 127 * 16777216 + 127 * 65536 + 127 * 256 => 2139062016
in the setup menu this is entered as (24bit RGB and 8bit Alpha)
(Red * 65536 + Green * 256 + Blue)
softhddevice.SkipLines = 0
skip 'n' lines at top and bottom of the video picture.

7
audio.c
View File

@ -435,8 +435,7 @@ static void AlsaFlushBuffers(void)
RingBufferReadAdvance(AlsaRingBuffer,
RingBufferUsedBytes(AlsaRingBuffer));
state = snd_pcm_state(AlsaPCMHandle);
Debug(3, "audio/alsa: flush state %d - %s\n", state,
snd_pcm_state_name(state));
Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state));
if (state != SND_PCM_STATE_OPEN) {
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
@ -1147,7 +1146,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
// update buffer
snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size);
Info(_("audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n"),
Debug(3, "audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n",
buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle,
buffer_size) * 1000 / (AudioSampleRate * AudioChannels *
AudioBytesProSample), period_size,
@ -1865,7 +1864,7 @@ static int OssSetup(int *freq, int *channels, int use_ac3)
OssFragmentTime = (bi.fragsize * 1000)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
Info(_("audio/oss: buffer size %d %dms, fragment size %d %dms\n"),
Debug(3, "audio/oss: buffer size %d %dms, fragment size %d %dms\n",
bi.fragsize * bi.fragstotal, (bi.fragsize * bi.fragstotal * 1000)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample), bi.fragsize,
OssFragmentTime);

23
codec.c
View File

@ -728,8 +728,8 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
if (audio_codec->capabilities & CODEC_CAP_TRUNCATED) {
Debug(3, "codec: audio can use truncated packets\n");
// we do not send complete frames
audio_decoder->AudioCtx->flags |= CODEC_FLAG_TRUNCATED;
// we send only complete frames
// audio_decoder->AudioCtx->flags |= CODEC_FLAG_TRUNCATED;
}
audio_decoder->SampleRate = 0;
audio_decoder->Channels = 0;
@ -800,6 +800,10 @@ void CodecSetAudioDownmix(int onoff)
** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C
** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE
** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr
**
** @param buf[IN,OUT] sample buffer
** @param size size of sample buffer in bytes
** @param channels number of channels interleaved in sample buffer
*/
static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
{
@ -960,10 +964,6 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
int err;
int isAC3;
audio_ctx = audio_decoder->AudioCtx;
audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
// FIXME: use swr_convert from swresample (only in ffmpeg!)
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);
@ -975,9 +975,16 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
audio_decoder->RemainCount = 0;
}
audio_ctx = audio_decoder->AudioCtx;
Debug(3, "codec/audio: format change %dHz %d channels %s\n",
audio_ctx->sample_rate, audio_ctx->channels,
CodecPassthroughAC3 ? "pass-through" : "");
audio_decoder->SampleRate = audio_ctx->sample_rate;
audio_decoder->HwSampleRate = audio_ctx->sample_rate;
audio_decoder->Channels = audio_ctx->channels;
audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
// SPDIF/HDMI passthrough
if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
audio_decoder->HwChannels = 2;
@ -1044,8 +1051,7 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
**
** @param audio_decoder audio decoder data
** @param data samples data
** @param count number of samples
**
** @param count number of bytes in sample data
*/
void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
{
@ -1162,7 +1168,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
// update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// FIXME: must first play remainings bytes, than change and play new.
if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3

59
softhddev.c
View File

@ -315,6 +315,12 @@ static inline int FastAc3Check(const uint8_t * p)
if (p[1] != 0x77) {
return 0;
}
if ((p[4] & 0xC0) == 0xC0) { // invalid sample rate
return 0;
}
if ((p[4] & 0x3F) > 37) { // invalid frame size
return 0;
}
return 1;
}
@ -338,7 +344,7 @@ static int Ac3Check(const uint8_t * data, int size)
// crc1 crc1 fscod|frmsizcod
fscod = data[4] >> 6;
frmsizcod = data[4] & 0x3F;
frmsizcod = data[4] & 0x3F; // invalid is checked by fast
frame_size = Ac3FrameSizeTable[frmsizcod][fscod] * 2;
if (frame_size + 2 > size) {
@ -425,12 +431,25 @@ typedef struct _pes_demux_
int64_t DTS; ///< decode time stamp
} PesDemux;
///
/// Reset packetized elementary stream demuxer.
///
static void PesReset(PesDemux * pesdx)
{
pesdx->State = PES_INIT;
pesdx->Index = 0;
pesdx->Skip = 0;
pesdx->StartCode = -1;
pesdx->PTS = AV_NOPTS_VALUE;
pesdx->DTS = AV_NOPTS_VALUE;
}
///
/// Initialize a packetized elementary stream demuxer.
///
/// @param pesdx packetized elementary stream demuxer
///
void PesInit(PesDemux * pesdx)
static void PesInit(PesDemux * pesdx)
{
memset(pesdx, 0, sizeof(*pesdx));
pesdx->Size = PES_MAX_PAYLOAD;
@ -438,20 +457,7 @@ void PesInit(PesDemux * pesdx)
if (!pesdx->Buffer) {
Fatal(_("pesdemux: out of memory\n"));
}
pesdx->PTS = AV_NOPTS_VALUE; // reset
pesdx->DTS = AV_NOPTS_VALUE;
}
///
/// Reset packetized elementary stream demuxer.
///
void PesReset(PesDemux * pesdx)
{
pesdx->State = PES_INIT;
pesdx->Index = 0;
pesdx->Skip = 0;
pesdx->PTS = AV_NOPTS_VALUE;
pesdx->DTS = AV_NOPTS_VALUE;
PesReset(pesdx);
}
///
@ -462,7 +468,8 @@ void PesReset(PesDemux * pesdx)
/// @param size number of payload data bytes
/// @param is_start flag, start of pes packet
///
void PesParse(PesDemux * pesdx, const uint8_t * data, int size, int is_start)
static void PesParse(PesDemux * pesdx, const uint8_t * data, int size,
int is_start)
{
const uint8_t *p;
const uint8_t *q;
@ -609,6 +616,8 @@ void PesParse(PesDemux * pesdx, const uint8_t * data, int size, int is_start)
Debug(3, "pesdemux: pes start code id %#02x\n", code);
// FIXME: need to save start code id?
pesdx->StartCode = code;
// we could have already detect a valid stream type
// don't switch to codec 'none'
}
pesdx->State = PES_HEADER;
@ -668,11 +677,11 @@ void PesParse(PesDemux * pesdx, const uint8_t * data, int size, int is_start)
// only private stream 1, has sub streams
pesdx->State = PES_START;
}
//pesdx->HeaderIndex = 0;
//pesdx->Index = 0;
}
break;
#if 0
// Played with PlayAudio
case PES_LPCM_HEADER: // lpcm header
n = pesdx->HeaderSize - pesdx->HeaderIndex;
if (n > size) {
@ -749,6 +758,7 @@ void PesParse(PesDemux * pesdx, const uint8_t * data, int size, int is_start)
AudioEnqueue(pesdx->Buffer, pesdx->Index);
pesdx->Index = 0;
break;
#endif
}
} while (size > 0);
}
@ -786,7 +796,7 @@ static PesDemux PesDemuxAudio[1]; ///< audio demuxer
///
/// @returns number of bytes consumed from buffer.
///
int TsDemuxer(TsDemux * tsdx, const uint8_t * data, int size)
static int TsDemuxer(TsDemux * tsdx, const uint8_t * data, int size)
{
const uint8_t *p;
@ -876,7 +886,7 @@ int PlayAudio(const uint8_t * data, int size, uint8_t id)
}
if (NewAudioStream) {
// FIXME: does this clear the audio ringbuffer?
// this clears the audio ringbuffer indirect, open and setup does it
CodecAudioClose(MyAudioDecoder);
AudioSetBufferTime(0);
AudioCodecID = CODEC_ID_NONE;
@ -928,7 +938,7 @@ int PlayAudio(const uint8_t * data, int size, uint8_t id)
AudioAvPkt->stream_index = 0;
}
if (AudioChannelID != id) {
if (AudioChannelID != id) { // id changed audio track changed
AudioChannelID = id;
AudioCodecID = CODEC_ID_NONE;
}
@ -1067,6 +1077,8 @@ int PlayAudio(const uint8_t * data, int size, uint8_t id)
/**
** Play transport stream audio packet.
**
** VDR can have buffered data belonging to previous channel!
**
** @param data data of exactly one complete TS packet
** @param size size of TS packet (always TS_PACKET_SIZE)
**
@ -1088,11 +1100,12 @@ int PlayTsAudio(const uint8_t * data, int size)
}
if (NewAudioStream) {
// FIXME: does this clear the audio ringbuffer?
// this clears the audio ringbuffer indirect, open and setup does it
CodecAudioClose(MyAudioDecoder);
// max time between audio packets 200ms + 24ms hw buffer
AudioSetBufferTime(264);
AudioCodecID = CODEC_ID_NONE;
AudioChannelID = -1;
NewAudioStream = 0;
PesReset(PesDemuxAudio);
}

42
video.c
View File

@ -538,15 +538,6 @@ static void VideoUpdateOutput(AVRational input_aspect_ratio, int input_width,
return;
}
///
/// Output video messages.
///
/// Reduce output.
///
//static void VideoMessage(const char *message)
//{
//}
//----------------------------------------------------------------------------
// GLX
//----------------------------------------------------------------------------
@ -4410,7 +4401,7 @@ static void VaapiSyncDisplayFrame(VaapiDecoder * decoder)
Info("video: %s%+5" PRId64 " %4" PRId64 " %3d/\\ms %3d v-buf\n",
Timestamp2String(video_clock),
abs((video_clock - audio_clock) / 90) <
9999 ? ((video_clock - audio_clock) / 90) : 88888,
8888 ? ((video_clock - audio_clock) / 90) : 8888,
AudioGetDelay() / 90, (int)VideoDeltaPTS / 90, VideoGetBuffers());
}
#endif
@ -4667,8 +4658,8 @@ static void VaapiOsdDrawARGB(int x, int y, int width, int height,
}
end = GetMsTicks();
Debug(3, "video/vaapi: osd upload %dx%d+%d+%d %d ms %d\n", width, height,
x, y, end - start, width * height * 4);
Debug(3, "video/vaapi: osd upload %dx%d+%d+%d %dms %d\n", width, height, x,
y, end - start, width * height * 4);
}
///
@ -5017,6 +5008,24 @@ static void VdpauOsdInit(int, int); ///< forward definition
//----------------------------------------------------------------------------
///
/// Output video messages.
///
/// Reduce output.
///
static void VdpauMessage(int level, const char *format, ...)
{
if (SysLogLevel > level || DebugLevel > level) {
va_list ap;
va_start(ap, format);
vsyslog(LOG_ERR, format, ap);
va_end(ap);
}
}
//----------------------------------------------------------------------------
///
/// Create surfaces for VDPAU decoder.
///
@ -5520,6 +5529,8 @@ static VdpauDecoder *VdpauNewHwDecoder(void)
decoder->Procamp.saturation = 1.0;
decoder->Procamp.hue = 0.0; // default values
decoder->PTS = AV_NOPTS_VALUE;
// FIXME: hack
VdpauDecoderN = 1;
VdpauDecoders[0] = decoder;
@ -7349,6 +7360,7 @@ static void VdpauDisplayFrame(void)
VdpauGetErrorString(status));
}
// check if surface was displayed for more than 1 frame
// FIXME: 21 only correct for 50Hz
if (last_time && first_time > last_time + 21 * 1000 * 1000) {
Debug(3, "video/vdpau: %ld display time %ld\n", first_time / 1000,
(first_time - last_time) / 1000);
@ -7482,7 +7494,7 @@ static void VdpauSyncDisplayFrame(VdpauDecoder * decoder)
Info("video: %s%+5" PRId64 " %4" PRId64 " %3d/\\ms %3d v-buf\n",
Timestamp2String(video_clock),
abs((video_clock - audio_clock) / 90) <
9999 ? ((video_clock - audio_clock) / 90) : 88888,
8888 ? ((video_clock - audio_clock) / 90) : 8888,
AudioGetDelay() / 90, (int)VideoDeltaPTS / 90, VideoGetBuffers());
}
#endif
@ -7876,8 +7888,8 @@ static void VdpauOsdDrawARGB(int x, int y, int width, int height,
#endif
end = GetMsTicks();
Debug(3, "video/vdpau: osd upload %dx%d+%d+%d %d ms %d\n", width, height,
x, y, end - start, width * height * 4);
Debug(3, "video/vdpau: osd upload %dx%d+%d+%d %dms %d\n", width, height, x,
y, end - start, width * height * 4);
}
///