summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
Diffstat
-rw-r--r--ChangeLog1
-rw-r--r--README.txt19
-rw-r--r--audio.c109
-rw-r--r--audio.h6
-rw-r--r--codec.c422
-rw-r--r--codec.h12
-rw-r--r--softhddev.c73
-rw-r--r--softhddev.h2
-rw-r--r--softhddevice.cpp72
9 files changed, 449 insertions, 267 deletions
diff --git a/ChangeLog b/ChangeLog
index 20f0582..891fb78 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,6 +1,7 @@
User johns
Date:
+ Improved pass-through (PCM+EAC3) support.
Support VDR 1.7.36 new build system.
Improves VDPAU display preemption handling.
Add modifiers to X11 remote key names (from Sibbi).
diff --git a/README.txt b/README.txt
index ac9562b..7b3f5fc 100644
--- a/README.txt
+++ b/README.txt
@@ -1,6 +1,6 @@
@file README.txt @brief A software HD output device for VDR
-Copyright (c) 2011, 2012 by Johns. All Rights Reserved.
+Copyright (c) 2011 - 2013 by Johns. All Rights Reserved.
Contributor(s):
@@ -89,8 +89,8 @@ Setup: environment
only if alsa is configured
ALSA_DEVICE=default
alsa PCM device name
- ALSA_AC3_DEVICE=
- alsa AC3/pass-though device name
+ ALSA_PASSTHROUGH_DEVICE=
+ alsa pass-though (AC3,EAC3,DTS,...) device name
ALSA_MIXER=default
alsa control device name
ALSA_MIXER_CHANNEL=PCM
@@ -99,8 +99,8 @@ Setup: environment
only if oss is configured
OSS_AUDIODEV=/dev/dsp
oss dsp device name
- OSS_AC3_AUDIODEV=
- oss AC3/pass-though device name
+ OSS_PASSTHROUGHDEV=
+ oss pass-though (AC3,EAC3,DTS,...) device name
OSS_MIXERDEV=/dev/mixer
oss mixer device name
OSS_MIXER_CHANNEL=pcm
@@ -156,13 +156,16 @@ Setup: /etc/vdr/setup.conf
delay audio or delay video
softhddevice.AudioPassthrough = 0
- 0 = none, 1 = AC-3
+ 0 = none, 1 = PCM, 2 = MPA, 4 = AC-3, 8 = EAC-3
- for AC-3 the pass-through device is used.
+ for PCM/AC-3/EAC-3 the pass-through device is used and the audio
+ stream is passed undecoded to the output device.
+ z.b. 12 = AC-3+EAC-3, 13 = PCM+AC-3+EAC-3
+ note: MPA/DTS/TrueHD/... aren't supported yet
softhddevice.AudioDownmix = 0
0 = none, 1 = downmix
- Use ffmpeg/libav downmix AC-3 to stereo.
+ Use ffmpeg/libav downmix of AC-3/EAC-3 audio to stereo.
softhddevice.AudioSoftvol = 0
0 = off, use hardware volume control
diff --git a/audio.c b/audio.c
index ed6198e..369eec9 100644
--- a/audio.c
+++ b/audio.c
@@ -129,7 +129,7 @@ static const char *AudioModuleName; ///< which audio module to use
/// Selected audio module.
static const AudioModule *AudioUsedModule = &NoopModule;
static const char *AudioPCMDevice; ///< PCM device name
-static const char *AudioAC3Device; ///< AC3 device name
+static const char *AudioPassthroughDevice; ///< Passthrough device name
static const char *AudioMixerDevice; ///< mixer device name
static const char *AudioMixerChannel; ///< mixer channel name
static char AudioDoingInit; ///> flag in init, reduce error
@@ -620,7 +620,7 @@ static void AudioResample(const int16_t * in, int in_chan, int frames,
typedef struct _audio_ring_ring_
{
char FlushBuffers; ///< flag: flush buffers
- char UseAc3; ///< flag: use ac3 pass-through
+ char Passthrough; ///< flag: use pass-through (AC3, ...)
int16_t PacketSize; ///< packet size
unsigned HwSampleRate; ///< hardware sample rate in Hz
unsigned HwChannels; ///< hardware number of channels
@@ -642,14 +642,14 @@ static unsigned AudioStartThreshold; ///< start play, if filled
**
** @param sample_rate sample-rate frequency
** @param channels number of channels
-** @param use_ac3 use ac3/pass-through device
+** @param passthrough use /pass-through (AC3, ...) device
**
** @retval -1 error
** @retval 0 okay
**
** @note this function shouldn't fail. Checks are done during AudoInit.
*/
-static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)
+static int AudioRingAdd(unsigned sample_rate, int channels, int passthrough)
{
unsigned u;
@@ -680,7 +680,7 @@ static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)
// FIXME: don't flush buffers here
AudioRing[AudioRingWrite].FlushBuffers = 1;
- AudioRing[AudioRingWrite].UseAc3 = use_ac3;
+ AudioRing[AudioRingWrite].Passthrough = passthrough;
AudioRing[AudioRingWrite].PacketSize = 0;
AudioRing[AudioRingWrite].InSampleRate = sample_rate;
AudioRing[AudioRingWrite].InChannels = channels;
@@ -836,8 +836,9 @@ static int AlsaPlayRingbuffer(void)
if (!avail) { // full or buffer empty
break;
}
- // muting ac3, can produce disturbance
- if (AudioMute || (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3)) {
+ // muting pass-through ac3, can produce disturbance
+ if (AudioMute || (AudioSoftVolume
+ && !AudioRing[AudioRingRead].Passthrough)) {
// FIXME: quick&dirty cast
AudioSoftAmplifier((int16_t *) p, avail);
// FIXME: if not all are written, we double amplify them
@@ -984,23 +985,23 @@ static int AlsaThread(void)
/**
** Open alsa pcm device.
**
-** @param use_ac3 use ac3/pass-through device
+** @param passthrough use pass-through (AC3, ...) device
*/
-static snd_pcm_t *AlsaOpenPCM(int use_ac3)
+static snd_pcm_t *AlsaOpenPCM(int passthrough)
{
const char *device;
snd_pcm_t *handle;
int err;
// &&|| hell
- if (!(use_ac3 && ((device = AudioAC3Device)
- || (device = getenv("ALSA_AC3_DEVICE"))))
+ if (!(passthrough && ((device = AudioPassthroughDevice)
+ || (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
device = "default";
}
if (!AudioDoingInit) { // reduce blabla during init
- Info(_("audio/alsa: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "",
- device);
+ Info(_("audio/alsa: using %sdevice '%s'\n"),
+ passthrough ? "pass-through " : "", device);
}
// open none blocking; if device is already used, we don't want wait
if ((err =
@@ -1167,7 +1168,7 @@ static int64_t AlsaGetDelay(void)
**
** @param freq sample frequency
** @param channels number of channels
-** @param use_ac3 use ac3/pass-through device
+** @param passthrough use pass-through (AC3, ...) device
**
** @retval 0 everything ok
** @retval 1 didn't support frequency/channels combination
@@ -1175,7 +1176,7 @@ static int64_t AlsaGetDelay(void)
**
** @todo FIXME: remove pointer for freq + channels
*/
-static int AlsaSetup(int *freq, int *channels, int use_ac3)
+static int AlsaSetup(int *freq, int *channels, int passthrough)
{
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
@@ -1183,7 +1184,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
int delay;
if (!AlsaPCMHandle) { // alsa not running yet
- // FIXME: if open fails for ac3, we never recover
+ // FIXME: if open fails for fe. pass-through, we never recover
return -1;
}
if (1) { // close+open to fix HDMI no sound bug
@@ -1193,7 +1194,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
// FIXME: need lock
AlsaPCMHandle = NULL; // other threads should check handle
snd_pcm_close(handle);
- if (!(handle = AlsaOpenPCM(use_ac3))) {
+ if (!(handle = AlsaOpenPCM(passthrough))) {
return -1;
}
AlsaPCMHandle = handle;
@@ -1453,7 +1454,7 @@ static int OssPlayRingbuffer(void)
break; // bi.bytes could become negative!
}
- if (AudioSoftVolume && !AudioRing[AudioRingRead].UseAc3) {
+ if (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough) {
// FIXME: quick&dirty cast
AudioSoftAmplifier((int16_t *) p, bi.bytes);
// FIXME: if not all are written, we double amplify them
@@ -1560,22 +1561,22 @@ static int OssThread(void)
/**
** Open OSS pcm device.
**
-** @param use_ac3 use ac3/pass-through device
+** @param passthrough use pass-through (AC3, ...) device
*/
-static int OssOpenPCM(int use_ac3)
+static int OssOpenPCM(int passthrough)
{
const char *device;
int fildes;
// &&|| hell
- if (!(use_ac3 && ((device = AudioAC3Device)
- || (device = getenv("OSS_AC3_AUDIODEV"))))
+ if (!(passthrough && ((device = AudioPassthroughDevice)
+ || (device = getenv("OSS_PASSTHROUGHDEV"))))
&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
device = "/dev/dsp";
}
if (!AudioDoingInit) {
- Info(_("audio/oss: using %sdevice '%s'\n"), use_ac3 ? "ac3 " : "",
- device);
+ Info(_("audio/oss: using %sdevice '%s'\n"),
+ passthrough ? "pass-through " : "", device);
}
if ((fildes = open(device, O_WRONLY)) < 0) {
@@ -1724,13 +1725,13 @@ static int64_t OssGetDelay(void)
**
** @param sample_rate sample rate/frequency
** @param channels number of channels
-** @param use_ac3 use ac3/pass-through device
+** @param passthrough use pass-through (AC3, ...) device
**
** @retval 0 everything ok
** @retval 1 didn't support frequency/channels combination
** @retval -1 something gone wrong
*/
-static int OssSetup(int *sample_rate, int *channels, int use_ac3)
+static int OssSetup(int *sample_rate, int *channels, int passthrough)
{
int ret;
int tmp;
@@ -1738,7 +1739,7 @@ static int OssSetup(int *sample_rate, int *channels, int use_ac3)
audio_buf_info bi;
if (OssPcmFildes == -1) { // OSS not ready
- // FIXME: if open fails for ac3, we never recover
+ // FIXME: if open fails for fe. pass-through, we never recover
return -1;
}
@@ -1748,7 +1749,7 @@ static int OssSetup(int *sample_rate, int *channels, int use_ac3)
fildes = OssPcmFildes;
OssPcmFildes = -1;
close(fildes);
- if (!(fildes = OssOpenPCM(use_ac3))) {
+ if (!(fildes = OssOpenPCM(passthrough))) {
return -1;
}
OssPcmFildes = fildes;
@@ -1929,13 +1930,14 @@ static void NoopSetVolume( __attribute__ ((unused))
/**
** Noop setup.
**
-** @param freq sample frequency
-** @param channels number of channels
+** @param freq sample frequency
+** @param channels number of channels
+** @param passthrough use pass-through (AC3, ...) device
*/
static int NoopSetup( __attribute__ ((unused))
int *channels, __attribute__ ((unused))
int *freq, __attribute__ ((unused))
- int use_ac3)
+ int passthrough)
{
return -1;
}
@@ -1973,17 +1975,17 @@ static const AudioModule NoopModule = {
*/
static int AudioNextRing(void)
{
- int use_ac3;
+ int passthrough;
int sample_rate;
int channels;
size_t used;
// update audio format
// not always needed, but check if needed is too complex
- use_ac3 = AudioRing[AudioRingRead].UseAc3;
+ passthrough = AudioRing[AudioRingRead].Passthrough;
sample_rate = AudioRing[AudioRingRead].HwSampleRate;
channels = AudioRing[AudioRingRead].HwChannels;
- if (AudioUsedModule->Setup(&sample_rate, &channels, use_ac3)) {
+ if (AudioUsedModule->Setup(&sample_rate, &channels, passthrough)) {
Error(_("audio: can't set channels %d sample-rate %dHz\n"), channels,
sample_rate);
// FIXME: handle error
@@ -2068,10 +2070,10 @@ static void *AudioPlayHandlerThread(void *dummy)
err = AudioUsedModule->Thread();
// underrun, check if new ring buffer is available
if (!err) {
- int use_ac3;
+ int passthrough;
int sample_rate;
int channels;
- int old_use_ac3;
+ int old_passthrough;
int old_sample_rate;
int old_channels;
@@ -2081,20 +2083,21 @@ static void *AudioPlayHandlerThread(void *dummy)
}
Debug(3, "audio: next ring buffer\n");
- old_use_ac3 = AudioRing[AudioRingRead].UseAc3;
+ old_passthrough = AudioRing[AudioRingRead].Passthrough;
old_sample_rate = AudioRing[AudioRingRead].HwSampleRate;
old_channels = AudioRing[AudioRingRead].HwChannels;
atomic_dec(&AudioRingFilled);
AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX;
- use_ac3 = AudioRing[AudioRingRead].UseAc3;
+ passthrough = AudioRing[AudioRingRead].Passthrough;
sample_rate = AudioRing[AudioRingRead].HwSampleRate;
channels = AudioRing[AudioRingRead].HwChannels;
Debug(3, "audio: thread channels %d frequency %dHz %s\n",
- channels, sample_rate, use_ac3 ? "ac3" : "pcm");
+ channels, sample_rate, passthrough ? "pass-through" : "");
// audio config changed?
- if (old_use_ac3 != use_ac3 || old_sample_rate != sample_rate
+ if (old_passthrough != passthrough
+ || old_sample_rate != sample_rate
|| old_channels != channels) {
// FIXME: wait for buffer drain
if (AudioNextRing()) {
@@ -2196,7 +2199,7 @@ void AudioEnqueue(const void *samples, int count)
}
// audio sample modification allowed and needed?
buffer = (void *)samples;
- if (!AudioRing[AudioRingWrite].UseAc3 && (AudioCompression
+ if (!AudioRing[AudioRingWrite].Passthrough && (AudioCompression
|| AudioNormalize
|| AudioRing[AudioRingWrite].InChannels !=
AudioRing[AudioRingWrite].HwChannels)) {
@@ -2416,7 +2419,7 @@ void AudioFlushBuffers(void)
old = AudioRingWrite;
AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
AudioRing[AudioRingWrite].FlushBuffers = 1;
- AudioRing[AudioRingWrite].UseAc3 = AudioRing[old].UseAc3;
+ AudioRing[AudioRingWrite].Passthrough = AudioRing[old].Passthrough;
AudioRing[AudioRingWrite].HwSampleRate = AudioRing[old].HwSampleRate;
AudioRing[AudioRingWrite].HwChannels = AudioRing[old].HwChannels;
AudioRing[AudioRingWrite].InSampleRate = AudioRing[old].InSampleRate;
@@ -2528,7 +2531,7 @@ int64_t AudioGetClock(void)
// delay zero, if no valid time stamp
if ((delay = AudioGetDelay())) {
- if (AudioRing[AudioRingRead].UseAc3) {
+ if (AudioRing[AudioRingRead].Passthrough) {
return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
}
return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
@@ -2548,7 +2551,7 @@ void AudioSetVolume(int volume)
AudioMute = !volume;
// reduce loudness for stereo output
if (AudioStereoDescent && AudioRing[AudioRingRead].InChannels == 2
- && !AudioRing[AudioRingRead].UseAc3) {
+ && !AudioRing[AudioRingRead].Passthrough) {
volume -= AudioStereoDescent;
if (volume < 0) {
volume = 0;
@@ -2565,9 +2568,9 @@ void AudioSetVolume(int volume)
/**
** Setup audio for requested format.
**
-** @param freq sample frequency
-** @param channels number of channels
-** @param use_ac3 use ac3/pass-through device
+** @param freq sample frequency
+** @param channels number of channels
+** @param passthrough use pass-through (AC3, ...) device
**
** @retval 0 everything ok
** @retval 1 didn't support frequency/channels combination
@@ -2575,10 +2578,10 @@ void AudioSetVolume(int volume)
**
** @todo add support to report best fitting format.
*/
-int AudioSetup(int *freq, int *channels, int use_ac3)
+int AudioSetup(int *freq, int *channels, int passthrough)
{
Debug(3, "audio: setup channels %d frequency %dHz %s\n", *channels, *freq,
- use_ac3 ? "ac3" : "pcm");
+ passthrough ? "pass-through" : "");
// invalid parameter
if (!freq || !channels || !*freq || !*channels) {
@@ -2586,7 +2589,7 @@ int AudioSetup(int *freq, int *channels, int use_ac3)
// FIXME: set flag invalid setup
return -1;
}
- return AudioRingAdd(*freq, *channels, use_ac3);
+ return AudioRingAdd(*freq, *channels, passthrough);
}
/**
@@ -2722,7 +2725,7 @@ void AudioSetDevice(const char *device)
**
** @note this is currently usable with alsa only.
*/
-void AudioSetDeviceAC3(const char *device)
+void AudioSetPassthroughDevice(const char *device)
{
if (!AudioModuleName) {
AudioModuleName = "alsa"; // detect alsa/OSS
@@ -2732,7 +2735,7 @@ void AudioSetDeviceAC3(const char *device)
AudioModuleName = "oss";
}
}
- AudioAC3Device = device;
+ AudioPassthroughDevice = device;
}
/**
@@ -2961,7 +2964,7 @@ static void PrintVersion(void)
#ifdef GIT_REV
"(GIT-" GIT_REV ")"
#endif
- ",\n\t(c) 2009 - 2012 by Johns\n"
+ ",\n\t(c) 2009 - 2013 by Johns\n"
"\tLicense AGPLv3: GNU Affero General Public License version 3\n");
}
diff --git a/audio.h b/audio.h
index 808d385..4a0ac51 100644
--- a/audio.h
+++ b/audio.h
@@ -1,7 +1,7 @@
///
/// @file audio.h @brief Audio module headerfile
///
-/// Copyright (c) 2009 - 2012 by Johns. All Rights Reserved.
+/// Copyright (c) 2009 - 2013 by Johns. All Rights Reserved.
///
/// Contributor(s):
///
@@ -48,7 +48,9 @@ extern void AudioSetCompression(int, int); ///< set compression parameters
extern void AudioSetStereoDescent(int); ///< set stereo loudness descent
extern void AudioSetDevice(const char *); ///< set PCM audio device
-extern void AudioSetDeviceAC3(const char *); ///< set pass-through device
+
+ /// set pass-through device
+extern void AudioSetPassthroughDevice(const char *);
extern void AudioSetChannel(const char *); ///< set mixer channel
extern void AudioInit(void); ///< setup audio module
extern void AudioExit(void); ///< cleanup and exit audio module
diff --git a/codec.c b/codec.c
index e3860f7..13801dc 100644
--- a/codec.c
+++ b/codec.c
@@ -1,7 +1,7 @@
///
/// @file codec.c @brief Codec functions
///
-/// Copyright (c) 2009 - 2012 by Johns. All Rights Reserved.
+/// Copyright (c) 2009 - 2013 by Johns. All Rights Reserved.
///
/// Contributor(s):
///
@@ -30,13 +30,13 @@
/// many bugs and incompatiblity in it. Don't use this shit.
///
- /// compile with passthrough support (stable, ac3 only)
+ /// compile with pass-through support (stable, AC-3, E-AC-3 only)
#define USE_PASSTHROUGH
- /// compile audio drift correction support (experimental)
+ /// compile audio drift correction support (very experimental)
#define USE_AUDIO_DRIFT_CORRECTION
- /// compile AC3 audio drift correction support (experimental)
+ /// compile AC-3 audio drift correction support (very experimental)
#define USE_AC3_DRIFT_CORRECTION
- /// use ffmpeg libswresample API
+ /// use ffmpeg libswresample API (autodected, Makefile)
#define noUSE_SWRESAMPLE
#include <stdio.h>
@@ -633,7 +633,7 @@ struct _audio_decoder_
AVCodec *AudioCodec; ///< audio codec
AVCodecContext *AudioCtx; ///< audio codec context
- int PassthroughAC3; ///< current ac-3 pass-through
+ char Passthrough; ///< current pass-through flags
int SampleRate; ///< current stream sample rate
int Channels; ///< current stream channels
@@ -651,6 +651,10 @@ struct _audio_decoder_
#endif
#endif
+ uint16_t Spdif[24576 / 2]; ///< SPDIF output buffer
+ int SpdifIndex; ///< index into SPDIF output buffer
+ int SpdifCount; ///< SPDIF repeat counter
+
int64_t LastDelay; ///< last delay
struct timespec LastTime; ///< last time
int64_t LastPTS; ///< last PTS
@@ -670,24 +674,32 @@ struct _audio_decoder_
#endif
};
+///
+/// IEC Data type enumeration.
+///
+enum IEC61937
+{
+ IEC61937_AC3 = 0x01, ///< AC-3 data
+ // FIXME: more data types
+ IEC61937_EAC3 = 0x15, ///< E-AC-3 data
+};
+
#ifdef USE_AUDIO_DRIFT_CORRECTION
#define CORRECT_PCM 1 ///< do PCM audio-drift correction
-#define CORRECT_AC3 2 ///< do AC3§ audio-drift correction
+#define CORRECT_AC3 2 ///< do AC3 audio-drift correction
static char CodecAudioDrift; ///< flag: enable audio-drift correction
#else
static const int CodecAudioDrift = 0;
#endif
#ifdef USE_PASSTHROUGH
-//static char CodecPassthroughPCM; ///< pass pcm through (unsupported)
-static char CodecPassthroughAC3; ///< pass ac3 through
-
-//static char CodecPassthroughDTS; ///< pass dts through (unsupported)
-//static char CodecPassthroughMPA; ///< pass mpa through (unsupported)
+ ///
+ /// Pass-through flags: CodecPCM, CodecAC3, CodecEAC3, ...
+ ///
+static char CodecPassthrough;
#else
-
-static const int CodecPassthroughAC3 = 0;
+static const int CodecPassthrough = 0;
#endif
-static char CodecDownmix; ///< enable ac-3 downmix
+static char CodecDownmix; ///< enable AC-3 decoder downmix
/**
** Allocate a new audio decoder context.
@@ -840,7 +852,7 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
void CodecSetAudioDrift(int mask)
{
#ifdef USE_AUDIO_DRIFT_CORRECTION
- CodecAudioDrift = mask & 3;
+ CodecAudioDrift = mask & (CORRECT_PCM | CORRECT_AC3);
#endif
(void)mask;
}
@@ -848,12 +860,12 @@ void CodecSetAudioDrift(int mask)
/**
** Set audio pass-through.
**
-** @param mask enable mask (PCM, AC3)
+** @param mask enable mask (PCM, AC3, EAC3)
*/
void CodecSetAudioPassthrough(int mask)
{
#ifdef USE_PASSTHROUGH
- CodecPassthroughAC3 = mask & 1 ? 1 : 0;
+ CodecPassthrough = mask & (CodecPCM | CodecAC3 | CodecEAC3);
#endif
(void)mask;
}
@@ -932,6 +944,178 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
}
}
+/**
+** Handle audio format changes helper.
+**
+** @param audio_decoder audio decoder data
+** @param[out] passthrough pass-through output
+*/
+static int CodecAudioUpdateHelper(AudioDecoder * audio_decoder,
+ int *passthrough)
+{
+ const AVCodecContext *audio_ctx;
+ int err;
+
+ audio_ctx = audio_decoder->AudioCtx;
+ Debug(3, "codec/audio: format change %s %dHz *%d channels%s%s%s%s%s\n",
+ av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
+ audio_ctx->channels, CodecPassthrough & CodecPCM ? " PCM" : "",
+ CodecPassthrough & CodecMPA ? " MPA" : "",
+ CodecPassthrough & CodecAC3 ? " AC3" : "",
+ CodecPassthrough & CodecEAC3 ? " EAC3" : "",
+ CodecPassthrough ? " pass-through" : "");
+
+ *passthrough = 0;
+ audio_decoder->SampleRate = audio_ctx->sample_rate;
+ audio_decoder->HwSampleRate = audio_ctx->sample_rate;
+ audio_decoder->Channels = audio_ctx->channels;
+ audio_decoder->HwChannels = audio_ctx->channels;
+ audio_decoder->Passthrough = CodecPassthrough;
+
+ // SPDIF/HDMI pass-through
+ if ((CodecPassthrough & CodecAC3 && audio_ctx->codec_id == CODEC_ID_AC3)
+ || (CodecPassthrough & CodecEAC3
+ && audio_ctx->codec_id == CODEC_ID_EAC3)) {
+ audio_decoder->HwChannels = 2;
+ audio_decoder->SpdifIndex = 0; // reset buffer
+ audio_decoder->SpdifCount = 0;
+ *passthrough = 1;
+ }
+ // channels not support?
+ if ((err =
+ AudioSetup(&audio_decoder->HwSampleRate,
+ &audio_decoder->HwChannels, *passthrough))) {
+
+ Debug(3, "codec/audio: audio setup error\n");
+ // FIXME: handle errors
+ audio_decoder->HwChannels = 0;
+ audio_decoder->HwSampleRate = 0;
+ return err;
+ }
+
+ Debug(3, "codec/audio: resample %s %dHz *%d -> %s %dHz *%d\n",
+ av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
+ audio_ctx->channels, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16),
+ audio_decoder->HwSampleRate, audio_decoder->HwChannels);
+
+ return 0;
+}
+
+/**
+** Audio pass-through decoder helper.
+**
+** @param audio_decoder audio decoder data
+** @param avpkt undecoded audio packet
+*/
+static int CodecAudioPassthroughHelper(AudioDecoder * audio_decoder,
+ const AVPacket * avpkt)
+{
+#ifdef USE_PASSTHROUGH
+ const AVCodecContext *audio_ctx;
+
+ audio_ctx = audio_decoder->AudioCtx;
+ // SPDIF/HDMI passthrough
+ if (CodecPassthrough & CodecAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
+ uint16_t *spdif;
+ int spdif_sz;
+
+ spdif = audio_decoder->Spdif;
+ spdif_sz = 6144;
+
+#ifdef USE_AC3_DRIFT_CORRECTION
+ // FIXME: this works with some TVs/AVReceivers
+ // FIXME: write burst size drift correction, which should work with all
+ if (CodecAudioDrift & CORRECT_AC3) {
+ int x;
+
+ x = (audio_decoder->DriftFrac +
+ (audio_decoder->DriftCorr * spdif_sz)) / (10 *
+ audio_decoder->HwSampleRate * 100);
+ audio_decoder->DriftFrac =
+ (audio_decoder->DriftFrac +
+ (audio_decoder->DriftCorr * spdif_sz)) % (10 *
+ audio_decoder->HwSampleRate * 100);
+ // round to word border
+ x *= audio_decoder->HwChannels * 4;
+ if (x < -64) { // limit correction
+ x = -64;
+ } else if (x > 64) {
+ x = 64;
+ }
+ spdif_sz += x;
+ }
+#endif
+
+ // build SPDIF header and append A52 audio to it
+ // avpkt is the original data
+ if (spdif_sz < avpkt->size + 8) {
+ Error(_("codec/audio: decoded data smaller than encoded\n"));
+ return -1;
+ }
+ spdif[0] = htole16(0xF872); // iec 61937 sync word
+ spdif[1] = htole16(0x4E1F);
+ spdif[2] = htole16(IEC61937_AC3 | (avpkt->data[5] & 0x07) << 8);
+ spdif[3] = htole16(avpkt->size * 8);
+ // copy original data for output
+ // FIXME: not 100% sure, if endian is correct on not intel hardware
+ swab(avpkt->data, spdif + 4, avpkt->size);
+ // FIXME: don't need to clear always
+ memset(spdif + 4 + avpkt->size / 2, 0, spdif_sz - 8 - avpkt->size);
+ // don't play with the ac-3 samples
+ AudioEnqueue(spdif, spdif_sz);
+ return 1;
+ }
+ if (CodecPassthrough & CodecEAC3 && audio_ctx->codec_id == CODEC_ID_EAC3) {
+ uint16_t *spdif;
+ int spdif_sz;
+ int repeat;
+
+ // build SPDIF header and append A52 audio to it
+ // avpkt is the original data
+ spdif = audio_decoder->Spdif;
+ spdif_sz = 6144;
+ // 24576 = 4 * 6144
+ if (spdif_sz < audio_decoder->SpdifIndex + avpkt->size + 8) {
+ Error(_("codec/audio: decoded data smaller than encoded\n"));
+ return -1;
+ }
+ // check if we must pack multiple packets
+ repeat = 1;
+ if ((avpkt->data[4] & 0xc0) != 0xc0) { // fscod
+ static const uint8_t eac3_repeat[4] = { 6, 3, 2, 1 };
+
+ // fscod2
+ repeat = eac3_repeat[(avpkt->data[4] & 0x30) >> 4];
+ }
+ //fprintf(stderr, "repeat %d\n", repeat);
+
+ // copy original data for output
+ // pack upto repeat EAC-3 pakets into one IEC 61937 burst
+ // FIXME: not 100% sure, if endian is correct on not intel hardware
+ swab(avpkt->data, spdif + 4 + audio_decoder->SpdifIndex, avpkt->size);
+ audio_decoder->SpdifIndex += avpkt->size;
+ if (++audio_decoder->SpdifCount < repeat) {
+ return 1;
+ }
+
+ spdif[0] = htole16(0xF872); // iec 61937 sync word
+ spdif[1] = htole16(0x4E1F);
+ spdif[2] = htole16(IEC61937_EAC3);
+ spdif[3] = htole16(audio_decoder->SpdifIndex * 8);
+ memset(spdif + 4 + audio_decoder->SpdifIndex / 2, 0,
+ spdif_sz - 8 - audio_decoder->SpdifIndex);
+
+ // don't play with the eac-3 samples
+ AudioEnqueue(spdif, spdif_sz);
+
+ audio_decoder->SpdifIndex = 0;
+ audio_decoder->SpdifCount = 0;
+ return 1;
+ }
+#endif
+ return 0;
+}
+
#ifndef USE_SWRESAMPLE
/**
@@ -1007,8 +1191,10 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
audio_decoder->Drift = drift;
corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
// SPDIF/HDMI passthrough
- if ((CodecAudioDrift & 2) && (!CodecPassthroughAC3
- || audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)) {
+ if ((CodecAudioDrift & CORRECT_AC3) && (!CodecPassthroughAC3
+ || audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
+ && (!CodecPassthroughEAC3
+ || audio_decoder->AudioCtx->codec_id != CODEC_ID_EAC3)) {
audio_decoder->DriftCorr = -corr;
}
@@ -1045,14 +1231,15 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
** Handle audio format changes.
**
** @param audio_decoder audio decoder data
+**
+** @note this is the old not good supported version
*/
static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
{
+ int passthrough;
const AVCodecContext *audio_ctx;
int err;
- int isAC3;
- // FIXME: use swr_convert from swresample (only in ffmpeg!)
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL;
@@ -1064,28 +1251,8 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
}
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;
- isAC3 = 1;
- } else {
- audio_decoder->HwChannels = audio_ctx->channels;
- isAC3 = 0;
- }
+ if ((err = CodecAudioUpdateHelper(audio_decoder, &passthrough))) {
- // channels not support?
- if ((err =
- AudioSetup(&audio_decoder->HwSampleRate,
- &audio_decoder->HwChannels, isAC3))) {
Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
audio_ctx->sample_rate, audio_ctx->channels,
audio_decoder->HwSampleRate, audio_decoder->HwChannels);
@@ -1101,19 +1268,21 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
Error(_("codec/audio: resample setup error\n"));
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
- return;
}
- } else {
- Debug(3, "codec/audio: audio setup error\n");
- // FIXME: handle errors
- audio_decoder->HwChannels = 0;
- audio_decoder->HwSampleRate = 0;
return;
}
+ Debug(3, "codec/audio: audio setup error\n");
+ // FIXME: handle errors
+ audio_decoder->HwChannels = 0;
+ audio_decoder->HwSampleRate = 0;
+ return;
+ }
+ if (passthrough) { // pass-through no conversion allowed
+ return;
}
// prepare audio drift resample
#ifdef USE_AUDIO_DRIFT_CORRECTION
- if ((CodecAudioDrift & 1) && !isAC3) {
+ if (CodecAudioDrift & CORRECT_PCM) {
if (audio_decoder->AvResample) {
Error(_("codec/audio: overwrite resample\n"));
}
@@ -1144,7 +1313,7 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
{
#ifdef USE_AUDIO_DRIFT_CORRECTION
- if ((CodecAudioDrift & 1) && audio_decoder->AvResample) {
+ if ((CodecAudioDrift & CORRECT_PCM) && audio_decoder->AvResample) {
int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
int16_t buftmp[MAX_CHANNELS][(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4];
@@ -1205,12 +1374,16 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
n *= 2;
n *= audio_decoder->HwChannels;
- CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
+ if (!(audio_decoder->Passthrough & CodecPCM)) {
+ CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
+ }
AudioEnqueue(buf, n);
return;
}
#endif
- CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
+ if (!(audio_decoder->Passthrough & CodecPCM)) {
+ CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
+ }
AudioEnqueue(data, count);
}
@@ -1232,6 +1405,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
audio_ctx = audio_decoder->AudioCtx;
+ // FIXME: don't need to decode pass-through codecs
buf_sz = sizeof(buf);
l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
if (avpkt->size != l) {
@@ -1250,7 +1424,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// FIXME: must first play remainings bytes, than change and play new.
- if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
+ if (audio_decoder->Passthrough != CodecPassthrough
|| audio_decoder->SampleRate != audio_ctx->sample_rate
|| audio_decoder->Channels != audio_ctx->channels) {
CodecAudioUpdateFormat(audio_decoder);
@@ -1283,48 +1457,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
CodecAudioEnqueue(audio_decoder, outbuf, outlen);
}
} else {
-#ifdef USE_PASSTHROUGH
- // SPDIF/HDMI passthrough
- if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
- // build SPDIF header and append A52 audio to it
- // avpkt is the original data
- buf_sz = 6144;
-
-#ifdef USE_AC3_DRIFT_CORRECTION
- if (CodecAudioDrift & 2) {
- int x;
-
- x = (audio_decoder->DriftFrac +
- (audio_decoder->DriftCorr * buf_sz)) / (10 *
- audio_decoder->HwSampleRate * 100);
- audio_decoder->DriftFrac =
- (audio_decoder->DriftFrac +
- (audio_decoder->DriftCorr * buf_sz)) % (10 *
- audio_decoder->HwSampleRate * 100);
- x *= audio_decoder->HwChannels * 4;
- if (x < -64) { // limit correction
- x = -64;
- } else if (x > 64) {
- x = 64;
- }
- buf_sz += x;
- }
-#endif
- if (buf_sz < avpkt->size + 8) {
- Error(_
- ("codec/audio: decoded data smaller than encoded\n"));
- return;
- }
- // copy original data for output
- // FIXME: not 100% sure, if endian is correct
- buf[0] = htole16(0xF872); // iec 61937 sync word
- buf[1] = htole16(0x4E1F);
- buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
- buf[3] = htole16(avpkt->size * 8);
- swab(avpkt->data, buf + 4, avpkt->size);
- memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
- // don't play with the ac-3 samples
- AudioEnqueue(buf, buf_sz);
+ if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
#if 0
@@ -1378,7 +1511,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
// DTS HD?
// True HD?
#endif
-#endif
CodecAudioEnqueue(audio_decoder, buf, buf_sz);
}
}
@@ -1461,8 +1593,10 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
audio_decoder->Drift = drift;
corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
// SPDIF/HDMI passthrough
- if ((CodecAudioDrift & 2) && (!CodecPassthroughAC3
- || audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)) {
+ if ((CodecAudioDrift & CORRECT_AC3) && (!(CodecPassthrough & CodecAC3)
+ || audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
+ && (!(CodecPassthrough & CodecEAC3)
+ || audio_decoder->AudioCtx->codec_id != CODEC_ID_EAC3)) {
audio_decoder->DriftCorr = -corr;
}
@@ -1504,49 +1638,27 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
*/
static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
{
+ int passthrough;
const AVCodecContext *audio_ctx;
- int err;
- int isAC3;
-
- audio_ctx = audio_decoder->AudioCtx;
- Debug(3, "codec/audio: format change %s %dHz *%d channels %s\n",
- av_get_sample_fmt_name(audio_ctx->sample_fmt), 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;
- isAC3 = 1;
- } else {
- audio_decoder->HwChannels = audio_ctx->channels;
- isAC3 = 0;
- }
-
- // channels not support?
- if ((err =
- AudioSetup(&audio_decoder->HwSampleRate,
- &audio_decoder->HwChannels, isAC3))) {
- Debug(3, "codec/audio: audio setup error\n");
- // FIXME: handle errors
- audio_decoder->HwChannels = 0;
- audio_decoder->HwSampleRate = 0;
+ if (CodecAudioUpdateHelper(audio_decoder, &passthrough)) {
+ // FIXME: handle swresample format conversions.
return;
}
-
- if (isAC3) { // no AC3 conversion allowed
+ if (passthrough) { // pass-through no conversion allowed
return;
}
- Debug(3, "codec/audio: resample %s %dHz *%d -> %s %dHz *%d\n",
- av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate,
- audio_ctx->channels, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16),
- audio_decoder->HwSampleRate, audio_decoder->HwChannels);
+ audio_ctx = audio_decoder->AudioCtx;
+
+#ifdef DEBUG
+ if (audio_ctx->sample_fmt == AV_SAMPLE_FMT_S16
+ && audio_ctx->sample_rate == audio_decoder->HwSampleRate
+ && !CodecAudioDrift) {
+ // FIXME: use Resample only, when it is needed!
+ fprintf(stderr, "no resample needed\n");
+ }
+#endif
audio_decoder->Resample =
swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout,
@@ -1579,6 +1691,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
audio_ctx = audio_decoder->AudioCtx;
+ // FIXME: don't need to decode pass-through codecs
frame.data[0] = NULL;
n = avcodec_decode_audio4(audio_ctx, &frame, &got_frame,
(AVPacket *) avpkt);
@@ -1602,42 +1715,20 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// format change
- if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
+ if (audio_decoder->Passthrough != CodecPassthrough
|| audio_decoder->SampleRate != audio_ctx->sample_rate
|| audio_decoder->Channels != audio_ctx->channels) {
CodecAudioUpdateFormat(audio_decoder);
-
}
if (!audio_decoder->HwSampleRate || !audio_decoder->HwChannels) {
return; // unsupported sample format
}
-#ifdef USE_PASSTHROUGH
- // SPDIF/HDMI passthrough
- if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
- int16_t spdif[6144 / 2];
- int spdif_sz;
- // build SPDIF header and append A52 audio to it
- // avpkt is the original data
- spdif_sz = 6144;
- if (spdif_sz < avpkt->size + 8) {
- Error(_("codec/audio: decoded data smaller than encoded\n"));
- return;
- }
- // copy original data for output
- spdif[0] = htole16(0xF872); // iec 61937 sync word
- spdif[1] = htole16(0x4E1F);
- spdif[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
- spdif[3] = htole16(avpkt->size * 8);
- // FIXME: not 100% sure, if endian is correct on not intel hardware
- swab(avpkt->data, spdif + 4, avpkt->size);
- memset(spdif + 4 + avpkt->size / 2, 0, spdif_sz - 8 - avpkt->size);
- // don't play with the ac-3 samples
- AudioEnqueue(spdif, spdif_sz);
+ if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
-#endif
+
if (0) {
char strbuf[32];
int data_sz;
@@ -1665,12 +1756,19 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
sizeof(outbuf) / (2 * audio_decoder->HwChannels),
(const uint8_t **)frame.extended_data, frame.nb_samples);
if (n > 0) {
- CodecReorderAudioFrame((int16_t *) outbuf,
- n * 2 * audio_decoder->HwChannels, audio_decoder->HwChannels);
+ if (!(audio_decoder->Passthrough & CodecPCM)) {
+ CodecReorderAudioFrame((int16_t *) outbuf,
+ n * 2 * audio_decoder->HwChannels,
+ audio_decoder->HwChannels);
+ }
AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
}
return;
}
+#ifdef DEBUG
+ // should be never reached
+ fprintf(stderr, "oops\n");
+#endif
}
#endif
diff --git a/codec.h b/codec.h
index 08cc5ac..4a11607 100644
--- a/codec.h
+++ b/codec.h
@@ -1,7 +1,7 @@
///
/// @file codec.h @brief Codec module headerfile
///
-/// Copyright (c) 2009 - 2012 by Johns. All Rights Reserved.
+/// Copyright (c) 2009 - 2013 by Johns. All Rights Reserved.
///
/// Contributor(s):
///
@@ -24,6 +24,16 @@
/// @{
//----------------------------------------------------------------------------
+// Defines
+//----------------------------------------------------------------------------
+
+#define CodecPCM 0x01 ///< PCM bit mask
+#define CodecMPA 0x02 ///< MPA bit mask (planned)
+#define CodecAC3 0x04 ///< AC-3 bit mask
+#define CodecEAC3 0x08 ///< EAC-3 bit mask
+#define CodecDTS 0x10 ///< DTS bit mask (planned)
+
+//----------------------------------------------------------------------------
// Typedefs
//----------------------------------------------------------------------------
diff --git a/softhddev.c b/softhddev.c
index 3cded90..c638ebb 100644
--- a/softhddev.c
+++ b/softhddev.c
@@ -314,7 +314,7 @@ const uint16_t Ac3FrameSizeTable[38][3] = {
};
///
-/// Fast check for AC3 audio.
+/// Fast check for (E)AC3 audio.
///
/// 5 bytes 0x0B77xxxxxx AC3 audio
///
@@ -326,17 +326,11 @@ 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;
}
///
-/// Check for AC-3 audio.
+/// Check for (E)AC-3 audio.
///
/// 0x0B77xxxxxx already checked.
///
@@ -347,20 +341,61 @@ static inline int FastAc3Check(const uint8_t * p)
/// @retval 0 no valid AC-3 audio
/// @retval >0 valid AC-3 audio
///
+/// o AC3 Header
+/// AAAAAAAA AAAAAAAA BBBBBBBB BBBBBBBB CCDDDDDD EEEEEFFF
+///
+/// o a 16x Frame sync, always 0x0B77
+/// o b 16x CRC 16
+/// o c 2x Samplerate
+/// o d 6x Framesize code
+/// o e 5x Bitstream ID
+/// o f 3x Bitstream mode
+///
+/// o EAC3 Header
+/// AAAAAAAA AAAAAAAA BBCCCDDD DDDDDDDD EEFFGGGH IIIII...
+///
+/// o a 16x Frame sync, always 0x0B77
+/// o b 2x Frame type
+/// o c 3x Sub stream ID
+/// o d 10x Framesize - 1 in words
+/// o e 2x Framesize code
+/// o f 2x Framesize code 2
+///
static int Ac3Check(const uint8_t * data, int size)
{
- int fscod;
- int frmsizcod;
int frame_size;
- // crc1 crc1 fscod|frmsizcod
- fscod = data[4] >> 6;
- frmsizcod = data[4] & 0x3F; // invalid is checked by fast
- frame_size = Ac3FrameSizeTable[frmsizcod][fscod] * 2;
+ if (size < 5) { // need 5 bytes to see if AC3/EAC3
+ return -5;
+ }
+
+ if (data[5] > (10 << 3)) { // EAC3
+ if ((data[4] & 0xF0) == 0xF0) { // invalid fscod fscod2
+ return 0;
+ }
+ frame_size = ((data[2] & 0x03) << 8) + data[3] + 1;
+ frame_size *= 2;
+ } else { // AC3
+ int fscod;
+ int frmsizcod;
+
+ // crc1 crc1 fscod|frmsizcod
+ fscod = data[4] >> 6;
+ if (fscod == 0x03) { // invalid sample rate
+ return 0;
+ }
+ frmsizcod = data[4] & 0x3F;
+ if (frmsizcod > 37) { // invalid frame size
+ return 0;
+ }
+ // invalid is checked above
+ frame_size = Ac3FrameSizeTable[frmsizcod][fscod] * 2;
+ }
if (frame_size + 5 > size) {
return -frame_size - 5;
}
+ // FIXME: relaxed checks if codec is already detected
// check if after this frame a new AC-3 frame starts
if (FastAc3Check(data + frame_size)) {
return frame_size;
@@ -617,6 +652,7 @@ static void PesParse(PesDemux * pesdx, const uint8_t * data, int size,
// 4 bytes 0xFFExxxxx Mpeg audio
// 5 bytes 0x0B77xxxxxx AC3 audio
+ // 6 bytes 0x0B77xxxxxxxx EAC3 audio
// 3 bytes 0x56Exxx AAC LATM audio
// 7/9 bytes 0xFFFxxxxxxxxxxx ADTS audio
// PCM audio can't be found
@@ -629,6 +665,9 @@ static void PesParse(PesDemux * pesdx, const uint8_t * data, int size,
if (!r && FastAc3Check(q)) {
r = Ac3Check(q, n);
codec_id = CODEC_ID_AC3;
+ if (r > 0 && q[5] > (10 << 3)) {
+ codec_id = CODEC_ID_EAC3;
+ }
}
if (!r && FastLatmCheck(q)) {
r = LatmCheck(q, n);
@@ -1119,6 +1158,7 @@ int PlayAudio(const uint8_t * data, int size, uint8_t id)
// 4 bytes 0xFFExxxxx Mpeg audio
// 3 bytes 0x56Exxx AAC LATM audio
// 5 bytes 0x0B77xxxxxx AC3 audio
+ // 6 bytes 0x0B77xxxxxxxx EAC3 audio
// 7/9 bytes 0xFFFxxxxxxxxxxx ADTS audio
// PCM audio can't be found
r = 0;
@@ -1134,6 +1174,9 @@ int PlayAudio(const uint8_t * data, int size, uint8_t id)
if ((id == 0xbd || (id & 0xF0) == 0x80) && !r && FastAc3Check(p)) {
r = Ac3Check(p, n);
codec_id = CODEC_ID_AC3;
+ if (r > 0 && p[5] > (10 << 3)) {
+ codec_id = CODEC_ID_EAC3;
+ }
/* faster ac3 detection at end of pes packet (no improvemnts)
if (AudioCodecID == codec_id && -r - 2 == n) {
r = n;
@@ -2787,7 +2830,7 @@ int ProcessArgs(int argc, char *const argv[])
AudioSetChannel(optarg);
continue;
case 'p': // pass-through audio device
- AudioSetDeviceAC3(optarg);
+ AudioSetPassthroughDevice(optarg);
continue;
case 'd': // x11 display name
X11DisplayName = optarg;
diff --git a/softhddev.h b/softhddev.h
index 69fa32e..6e446dc 100644
--- a/softhddev.h
+++ b/softhddev.h
@@ -106,6 +106,8 @@ extern "C"
extern void PipStop(void);
/// Pip play video packet
extern int PipPlayVideo(const uint8_t *, int);
+
+ extern const char *X11DisplayName; ///< x11 display name
#ifdef __cplusplus
}
#endif
diff --git a/softhddevice.cpp b/softhddevice.cpp
index a9c2ab3..9f35371 100644
--- a/softhddevice.cpp
+++ b/softhddevice.cpp
@@ -20,6 +20,8 @@
/// $Id$
//////////////////////////////////////////////////////////////////////////////
+#define __STDC_CONSTANT_MACROS ///< needed for ffmpeg UINT64_C
+
#include <vdr/interface.h>
#include <vdr/plugin.h>
#include <vdr/player.h>
@@ -34,15 +36,15 @@
#include "softhddev.h"
#include "softhddevice.h"
#include "softhddevice_service.h"
+
extern "C"
{
+#include <stdint.h>
+#include <libavcodec/avcodec.h>
+
#include "audio.h"
#include "video.h"
- extern const char *X11DisplayName; ///< x11 display name
-
- extern void CodecSetAudioDrift(int);
- extern void CodecSetAudioPassthrough(int);
- extern void CodecSetAudioDownmix(int);
+#include "codec.h"
}
//////////////////////////////////////////////////////////////////////////////
@@ -126,7 +128,8 @@ static int ConfigAutoCropTolerance; ///< auto crop detection tolerance
static int ConfigVideoAudioDelay; ///< config audio delay
static char ConfigAudioDrift; ///< config audio drift
-static char ConfigAudioPassthrough; ///< config audio pass-through
+static char ConfigAudioPassthrough; ///< config audio pass-through mask
+static char AudioPassthroughState; ///< flag audio pass-through on/off
static char ConfigAudioDownmix; ///< config ffmpeg audio downmix
static char ConfigAudioSoftvol; ///< config use software volume
static char ConfigAudioNormalize; ///< config use normalize volume
@@ -138,7 +141,7 @@ int ConfigAudioBufferTime; ///< config size ms of audio buffer
static char *ConfigX11Display; ///< config x11 display
static char *ConfigAudioDevice; ///< config audio stereo device
-static char *ConfigAC3Device; ///< config audio passthrough device
+static char *ConfigPassthroughDevice; ///< config audio pass-through device
#ifdef USE_PIP
static int ConfigPipX = 100 - 3 - 18; ///< config pip pip x in %
@@ -615,7 +618,9 @@ class cMenuSetupSoft:public cMenuSetupPage
int Audio;
int AudioDelay;
int AudioDrift;
- int AudioPassthrough;
+ int AudioPassthroughPCM;
+ int AudioPassthroughAC3;
+ int AudioPassthroughEAC3;
int AudioDownmix;
int AudioSoftvol;
int AudioNormalize;
@@ -720,9 +725,6 @@ void cMenuSetupSoft::Create(void)
static const char *const audiodrift[] = {
"None", "PCM", "AC-3", "PCM + AC-3"
};
- static const char *const passthrough[] = {
- "None", "AC-3"
- };
static const char *const resolution[RESOLUTIONS] = {
"576i", "720p", "fake 1080i", "1080i"
};
@@ -844,9 +846,13 @@ void cMenuSetupSoft::Create(void)
-1000, 1000));
Add(new cMenuEditStraItem(tr("Audio drift correction"), &AudioDrift, 4,
audiodrift));
- Add(new cMenuEditStraItem(tr("Audio pass-through"), &AudioPassthrough,
- 2, passthrough));
- Add(new cMenuEditBoolItem(tr("Enable AC-3 (decoder) downmix"),
+ Add(new cMenuEditBoolItem(tr("Enable PCM pass-through"),
+ &AudioPassthroughPCM, trVDR("no"), trVDR("yes")));
+ Add(new cMenuEditBoolItem(tr("Enable AC-3 pass-through"),
+ &AudioPassthroughAC3, trVDR("no"), trVDR("yes")));
+ Add(new cMenuEditBoolItem(tr("Enable EAC-3 pass-through"),
+ &AudioPassthroughEAC3, trVDR("no"), trVDR("yes")));
+ Add(new cMenuEditBoolItem(tr("Enable (E)AC-3 (decoder) downmix"),
&AudioDownmix, trVDR("no"), trVDR("yes")));
Add(new cMenuEditBoolItem(tr("Volume control"), &AudioSoftvol,
tr("Hardware"), tr("Software")));
@@ -1031,7 +1037,9 @@ cMenuSetupSoft::cMenuSetupSoft(void)
Audio = 0;
AudioDelay = ConfigVideoAudioDelay;
AudioDrift = ConfigAudioDrift;
- AudioPassthrough = ConfigAudioPassthrough;
+ AudioPassthroughPCM = ConfigAudioPassthrough & CodecPCM;
+ AudioPassthroughAC3 = ConfigAudioPassthrough & CodecAC3;
+ AudioPassthroughEAC3 = ConfigAudioPassthrough & CodecEAC3;
AudioDownmix = ConfigAudioDownmix;
AudioSoftvol = ConfigAudioSoftvol;
AudioNormalize = ConfigAudioNormalize;
@@ -1174,8 +1182,12 @@ void cMenuSetupSoft::Store(void)
VideoSetAudioDelay(ConfigVideoAudioDelay);
SetupStore("AudioDrift", ConfigAudioDrift = AudioDrift);
CodecSetAudioDrift(ConfigAudioDrift);
- SetupStore("AudioPassthrough", ConfigAudioPassthrough = AudioPassthrough);
+ ConfigAudioPassthrough = (AudioPassthroughPCM ? CodecPCM : 0)
+ | (AudioPassthroughAC3 ? CodecAC3 : 0)
+ | (AudioPassthroughEAC3 ? CodecEAC3 : 0);
+ SetupStore("AudioPassthrough", ConfigAudioPassthrough);
CodecSetAudioPassthrough(ConfigAudioPassthrough);
+ AudioPassthroughState = 1;
SetupStore("AudioDownmix", ConfigAudioDownmix = AudioDownmix);
CodecSetAudioDownmix(ConfigAudioDownmix);
SetupStore("AudioSoftvol", ConfigAudioSoftvol = AudioSoftvol);
@@ -1786,18 +1798,23 @@ static void HandleHotkey(int code)
{
switch (code) {
case 10: // disable pass-through
- CodecSetAudioPassthrough(ConfigAudioPassthrough = 0);
+ AudioPassthroughState = 0;
+ CodecSetAudioPassthrough(0);
Skins.QueueMessage(mtInfo, tr("pass-through disabled"));
break;
case 11: // enable pass-through
- CodecSetAudioPassthrough(ConfigAudioPassthrough = 1);
+ // note: you can't enable, without configured pass-through
+ AudioPassthroughState = 1;
+ CodecSetAudioPassthrough(ConfigAudioPassthrough);
Skins.QueueMessage(mtInfo, tr("pass-through enabled"));
break;
case 12: // toggle pass-through
- CodecSetAudioPassthrough(ConfigAudioPassthrough ^= 1);
- if (ConfigAudioPassthrough) {
+ AudioPassthroughState ^= 1;
+ if (AudioPassthroughState) {
+ CodecSetAudioPassthrough(ConfigAudioPassthrough);
Skins.QueueMessage(mtInfo, tr("pass-through enabled"));
} else {
+ CodecSetAudioPassthrough(0);
Skins.QueueMessage(mtInfo, tr("pass-through disabled"));
}
break;
@@ -2902,6 +2919,9 @@ bool cPluginSoftHdDevice::SetupParse(const char *name, const char *value)
}
if (!strcasecmp(name, "AudioPassthrough")) {
CodecSetAudioPassthrough(ConfigAudioPassthrough = atoi(value));
+ if (ConfigAudioPassthrough) {
+ AudioPassthroughState = 1;
+ }
return true;
}
if (!strcasecmp(name, "AudioDownmix")) {
@@ -3272,13 +3292,13 @@ cString cPluginSoftHdDevice::SVDRPCommand(const char *command,
free(tmp);
return "missing option argument";
}
- free(ConfigAC3Device);
- ConfigAC3Device = strdup(o);
- AudioSetDeviceAC3(ConfigAC3Device);
+ free(ConfigPassthroughDevice);
+ ConfigPassthroughDevice = strdup(o);
+ AudioSetPassthroughDevice(ConfigPassthroughDevice);
} else if (!strncmp(s, "-p", 2)) {
- free(ConfigAC3Device);
- ConfigAC3Device = strdup(s + 2);
- AudioSetDeviceAC3(ConfigAC3Device);
+ free(ConfigPassthroughDevice);
+ ConfigPassthroughDevice = strdup(s + 2);
+ AudioSetPassthroughDevice(ConfigPassthroughDevice);
} else if (*s) {
free(tmp);
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-14 16:48:13 +0000