frodovdr
/
vdr-plugin-softhddevice
mirror of https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
1
0
Fork 0
Code Issues Releases Wiki Activity

Add OSS Mixer support.

This commit is contained in:
Johns 2012-01-04 16:59:48 +01:00
parent 0f449c2394
commit aba14813c0
4 changed files with 118 additions and 17 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
ChangeLog
View File

@ -1,7 +1,10 @@
User johns
Date:
New audio driver oss.
Release Version 0.1.5
Adds OSS mixer support.
Fix bug: audio new stream is not thread safe.
New audio driver OSS.
Fix bug: needed down sampling of 3/5/6 to 2 channels not reported.
Search audio sync inside PES packets, for insane dvb streams.
Use only the needed number of surfaces.

2
README.txt
View File

@ -85,6 +85,8 @@ Setup: environment
oss dsp device name
OSS_MIXERDEV=/dev/mixer
oss mixer device name
OSS_MIXER_CHANNEL=pcm
oss mixer channel name
Setup: /etc/vdr/setup.conf
------

2
Todo
View File

@ -68,6 +68,8 @@ audio/alsa:
CodecAudioOpen can fail "can't open audio codec" and does Fatal exit.
audio/oss:
alsa oss emulation mixer "pcm" not working
ring buffer overflow with alsa oss emulation
playback of recording
play back is too fast

126
audio.c
View File

@ -96,9 +96,9 @@
// Variables
//----------------------------------------------------------------------------
static const char *AudioPCMDevice; ///< alsa PCM device name
static const char *AudioMixerDevice; ///< alsa mixer device name
static const char *AudioMixerChannel; ///< alsa mixer channel name
static const char *AudioPCMDevice; ///< alsa/oss PCM device name
static const char *AudioMixerDevice; ///< alsa/oss mixer device name
static const char *AudioMixerChannel; ///< alsa/oss mixer channel name
static volatile char AudioRunning; ///< thread running / stopped
static int AudioPaused; ///< audio paused
static unsigned AudioSampleRate; ///< audio sample rate in hz
@ -635,7 +635,7 @@ static void AlsaInitMixer(void)
AlsaMixer = alsa_mixer;
AlsaMixerElem = alsa_mixer_elem;
} else {
Error(_("audio/alsa: can't open alsa mixer '%s'\n"), device);
Error(_("audio/alsa: can't open mixer '%s'\n"), device);
}
}
@ -854,10 +854,14 @@ static int AlsaSetup(int *freq, int *channels)
AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
// min 333ms
if (AlsaStartThreshold < (*freq * *channels * 2U) / 3) {
AlsaStartThreshold = (*freq * *channels * 2U) / 3;
if (AlsaStartThreshold < (*freq * *channels * AudioBytesProSample) / 3U) {
AlsaStartThreshold = (*freq * *channels * AudioBytesProSample) / 3U;
}
Debug(3, "audio/alsa: delay %u ms\n", (AlsaStartThreshold * 1000)
// no bigger, than the buffer
if (AlsaStartThreshold > RingBufferFreeBytes(AlsaRingBuffer)) {
AlsaStartThreshold = RingBufferFreeBytes(AlsaRingBuffer);
}
Info(_("audio/alsa: delay %u ms\n"), (AlsaStartThreshold * 1000)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
return ret;
@ -926,6 +930,7 @@ static void AlsaExit(void)
static int OssPcmFildes = -1; ///< pcm file descriptor
static int OssMixerFildes = -1; ///< mixer file descriptor
static int OssMixerChannel; ///< mixer channel index
static RingBuffer *OssRingBuffer; ///< audio ring buffer
static unsigned OssStartThreshold; ///< start play, if filled
@ -995,8 +1000,8 @@ static int OssPlayRingbuffer(void)
if (n < bi.bytes) { // not enough bytes in ring buffer
bi.bytes = n;
}
if (!bi.bytes) { // full or buffer empty
break;
if (bi.bytes <= 0) { // full or buffer empty
break; // bi.bytes could become negative!
}
n = write(OssPcmFildes, p, bi.bytes);
@ -1076,7 +1081,7 @@ static void OssInitPCM(void)
}
}
if ((fildes = open(device, O_WRONLY)) < 0) {
Error(_("audio/oss: can't open device '%s': %s\n"), device,
Error(_("audio/oss: can't open dsp device '%s': %s\n"), device,
strerror(errno));
return;
}
@ -1084,6 +1089,85 @@ static void OssInitPCM(void)
OssPcmFildes = fildes;
}
//----------------------------------------------------------------------------
// OSS Mixer
//----------------------------------------------------------------------------
/**
** Set oss mixer volume (0-100)
**
** @param volume volume (0 .. 100)
*/
static void OssSetVolume(int volume)
{
int v;
if (OssMixerFildes != -1) {
v = (volume * 255) / 100;
v &= 0xff;
v = (v << 8) | v;
if (ioctl(OssMixerFildes, MIXER_WRITE(OssMixerChannel), &v) < 0) {
Error(_("audio/oss: ioctl(MIXER_WRITE): %s\n"), strerror(errno));
}
}
}
/**
** Mixer channel name table.
*/
static const char *OssMixerChannelNames[SOUND_MIXER_NRDEVICES] =
SOUND_DEVICE_NAMES;
/**
** Initialize oss mixer.
*/
static void OssInitMixer(void)
{
const char *device;
const char *channel;
int fildes;
int devmask;
int i;
if (!(device = AudioMixerDevice)) {
if (!(device = getenv("OSS_MIXERDEV"))) {
device = "/dev/mixer";
}
}
if (!(channel = AudioMixerChannel)) {
if (!(channel = getenv("OSS_MIXER_CHANNEL"))) {
channel = "pcm";
}
}
Debug(3, "audio/oss: mixer %s - %s open\n", device, channel);
if ((fildes = open(device, O_RDWR)) < 0) {
Error(_("audio/oss: can't open mixer device '%s': %s\n"), device,
strerror(errno));
return;
}
// search channel name
if (ioctl(fildes, SOUND_MIXER_READ_DEVMASK, &devmask) < 0) {
Error(_("audio/oss: ioctl(SOUND_MIXER_READ_DEVMASK): %s\n"),
strerror(errno));
close(fildes);
return;
}
for (i = 0; i < SOUND_MIXER_NRDEVICES; ++i) {
if (!strcasecmp(OssMixerChannelNames[i], channel)) {
if (devmask & (1 << i)) {
OssMixerFildes = fildes;
OssMixerChannel = i;
return;
}
Error(_("audio/oss: channel '%s' not supported\n"), channel);
break;
}
}
Error(_("audio/oss: channel '%s' not found\n"), channel);
close(fildes);
}
//----------------------------------------------------------------------------
// OSS API
//----------------------------------------------------------------------------
@ -1120,8 +1204,10 @@ static uint64_t OssGetDelay(void)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
pts += ((uint64_t) RingBufferUsedBytes(OssRingBuffer) * 90 * 1000)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample);
Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 " ms\n",
RingBufferUsedBytes(OssRingBuffer), pts / 90);
if (pts > 600 * 90) {
Debug(4, "audio/oss: hw+sw delay %zd %" PRId64 " ms\n",
RingBufferUsedBytes(OssRingBuffer), pts / 90);
}
return pts;
}
@ -1211,7 +1297,7 @@ static int OssSetup(int *freq, int *channels)
Error(_("audio/oss: ioctl(SNDCTL_DSP_GETOSPACE): %s\n"),
strerror(errno));
} else {
Info(_("audio/oss: %d bytes buffered\n"), bi.bytes);
Debug(3, "audio/oss: %d bytes buffered\n", bi.bytes);
}
tmp = -1;
@ -1226,8 +1312,16 @@ static int OssSetup(int *freq, int *channels)
}
// start when enough bytes for initial write
OssStartThreshold = bi.bytes + tmp;
// min 333ms
if (OssStartThreshold < (*freq * *channels * AudioBytesProSample) / 3U) {
OssStartThreshold = (*freq * *channels * AudioBytesProSample) / 3U;
}
// no bigger, than the buffer
if (OssStartThreshold > RingBufferFreeBytes(OssRingBuffer)) {
OssStartThreshold = RingBufferFreeBytes(OssRingBuffer);
}
Debug(3, "audio/alsa: delay %u ms\n", (OssStartThreshold * 1000)
Info(_("audio/oss: delay %u ms\n"), (OssStartThreshold * 1000)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
}
@ -1242,7 +1336,7 @@ static void OssInit(void)
OssRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit
OssInitPCM();
// OssInitMixer();
OssInitMixer();
}
/**
@ -1429,7 +1523,7 @@ void AudioSetVolume(int volume)
AlsaSetVolume(volume);
#endif
#ifdef USE_OSS
#warning "AudioSetVolume not written"
OssSetVolume(volume);
#endif
(void)volume;
}