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Implemented support for the new mixer thread in the lowlevel sound module

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'alsa'. Cleaned up the code a little bit.
TODO #1: Implement mixer thread support in the OSX sound driver.
TODO #2: Code cleanups in the Bochs sound code.
TODO #3: Resampling support and improvements in the wave recording code.
This commit is contained in:
Volker Ruppert 2015-02-07 18:49:09 +00:00
parent bfb5ec8cf9
commit 9c1070cbed
2 changed files with 52 additions and 91 deletions
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117
bochs/iodev/sound/soundalsa.cc
View File

@ -27,6 +27,10 @@
#if BX_HAVE_ALSASOUND && BX_SUPPORT_SOUNDLOW
#ifndef WIN32
#include <pthread.h>
#endif
#define LOG_THIS
bx_sound_alsa_c::bx_sound_alsa_c()
@ -202,6 +206,9 @@ int bx_sound_alsa_c::closemidioutput()
int bx_sound_alsa_c::openwaveoutput(const char *wavedev)
{
set_pcm_params(real_pcm_param);
pcm_callback_id = register_wave_callback(this, pcm_callback);
BX_INIT_MUTEX(mixer_mutex);
start_mixer_thread();
return BX_SOUNDLOW_OK;
}
@ -216,7 +223,7 @@ int bx_sound_alsa_c::alsa_pcm_open(bx_bool mode, int frequency, int bits, bx_boo
alsa_pcm[mode].audio_bufsize = 0;
if (alsa_pcm[mode].handle == NULL) {
ret = snd_pcm_open(&alsa_pcm[mode].handle, "default", mode ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
ret = snd_pcm_open(&alsa_pcm[mode].handle, "default", mode ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 0);
if (ret < 0) {
return BX_SOUNDLOW_ERR;
}
@ -226,16 +233,18 @@ int bx_sound_alsa_c::alsa_pcm_open(bx_bool mode, int frequency, int bits, bx_boo
snd_pcm_hw_params_any(alsa_pcm[mode].handle, params);
snd_pcm_hw_params_set_access(alsa_pcm[mode].handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if ((frequency == wave_ch[mode].oldfreq) &&
(bits == wave_ch[mode].oldbits) &&
(stereo == wave_ch[mode].oldstereo) &&
(format == wave_ch[mode].oldformat))
if (mode == 1) {
if ((frequency == wavein_param.samplerate) &&
(bits == wavein_param.bits) &&
(stereo == (wavein_param.channels == 2)) &&
(format == wavein_param.format))
return BX_SOUNDLOW_OK; // nothing to do
wave_ch[mode].oldfreq = frequency;
wave_ch[mode].oldbits = bits;
wave_ch[mode].oldstereo = stereo;
wave_ch[mode].oldformat = format;
wavein_param.samplerate = frequency;
wavein_param.bits = bits;
wavein_param.channels = stereo + 1;
wavein_param.format = format;
}
freq = (unsigned int)frequency;
@ -283,18 +292,17 @@ int bx_sound_alsa_c::set_pcm_params(bx_pcm_param_t param)
return alsa_pcm_open(0, param.samplerate, param.bits, param.channels == 2, param.format);
}
int bx_sound_alsa_c::alsa_pcm_write()
int bx_sound_alsa_c::get_waveout_packetsize()
{
int ret;
if (alsa_pcm[0].buffer == NULL) {
alsa_pcm[0].buffer = (char *)malloc(alsa_pcm[0].alsa_bufsize);
return alsa_pcm[0].alsa_bufsize;
}
while (alsa_pcm[0].audio_bufsize >= alsa_pcm[0].alsa_bufsize) {
memcpy(alsa_pcm[0].buffer, audio_buffer[0], alsa_pcm[0].alsa_bufsize);
ret = snd_pcm_writei(alsa_pcm[0].handle, alsa_pcm[0].buffer, alsa_pcm[0].frames);
if (ret == -EAGAIN)
continue;
int bx_sound_alsa_c::waveout(int length, Bit8u data[])
{
if (!alsa_pcm[0].handle || (length > alsa_pcm[0].alsa_bufsize)) {
return BX_SOUNDLOW_ERR;
}
int ret = snd_pcm_writei(alsa_pcm[0].handle, data, alsa_pcm[0].frames);
if (ret == -EPIPE) {
/* EPIPE means underrun */
BX_ERROR(("ALSA: underrun occurred"));
@ -304,61 +312,6 @@ int bx_sound_alsa_c::alsa_pcm_write()
} else if (ret != (int)alsa_pcm[0].frames) {
BX_ERROR(("ALSA: short write, write %d frames", ret));
}
alsa_pcm[0].audio_bufsize -= alsa_pcm[0].alsa_bufsize;
memmove(audio_buffer[0], audio_buffer[0]+alsa_pcm[0].alsa_bufsize, alsa_pcm[0].audio_bufsize);
}
if ((alsa_pcm[0].audio_bufsize == 0) && (alsa_pcm[0].buffer != NULL)) {
free(alsa_pcm[0].buffer);
alsa_pcm[0].buffer = NULL;
}
return BX_SOUNDLOW_OK;
}
int bx_sound_alsa_c::sendwavepacket(int length, Bit8u data[], bx_pcm_param_t *src_param)
{
int len2;
if (memcmp(src_param, &emu_pcm_param, sizeof(bx_pcm_param_t)) != 0) {
emu_pcm_param = *src_param;
cvt_mult = (src_param->bits == 8) ? 2 : 1;
if (src_param->channels == 1) cvt_mult <<= 1;
if (src_param->samplerate != real_pcm_param.samplerate) {
real_pcm_param.samplerate = src_param->samplerate;
set_pcm_params(real_pcm_param);
}
}
len2 = length * cvt_mult;
if (!alsa_pcm[0].handle) {
// Alert indicating that caller is probably erroneous
BX_ERROR(("sendwavepacket(): pcm is not open"));
return BX_SOUNDLOW_ERR;
}
if ((alsa_pcm[0].audio_bufsize+len2) > BX_SOUND_ALSA_BUFSIZE) {
len2 = BX_SOUND_ALSA_BUFSIZE - alsa_pcm[0].audio_bufsize;
BX_ERROR(("ALSA: audio buffer overflow"));
}
if (len2 > 0) {
convert_pcm_data(data, length, audio_buffer[0]+alsa_pcm[0].audio_bufsize, len2, src_param);
alsa_pcm[0].audio_bufsize += len2;
}
if (alsa_pcm[0].audio_bufsize < alsa_pcm[0].alsa_bufsize) {
return BX_SOUNDLOW_OK;
} else {
return alsa_pcm_write();
}
}
int bx_sound_alsa_c::stopwaveplayback()
{
if (alsa_pcm[0].handle && alsa_pcm[0].audio_bufsize > 0) {
if (alsa_pcm[0].audio_bufsize < alsa_pcm[0].alsa_bufsize) {
memset(audio_buffer[0]+alsa_pcm[0].audio_bufsize, 0, alsa_pcm[0].alsa_bufsize-alsa_pcm[0].audio_bufsize);
alsa_pcm[0].audio_bufsize = alsa_pcm[0].alsa_bufsize;
}
alsa_pcm_write();
}
return BX_SOUNDLOW_OK;
}
@ -369,7 +322,6 @@ int bx_sound_alsa_c::closewaveoutput()
snd_pcm_close(alsa_pcm[0].handle);
alsa_pcm[0].handle = NULL;
}
return BX_SOUNDLOW_OK;
}
@ -380,6 +332,7 @@ int bx_sound_alsa_c::openwaveinput(const char *wavedev, sound_record_handler_t r
record_timer_index = bx_pc_system.register_timer(this, record_timer_handler, 1, 1, 0, "soundlnx");
// record timer: inactive, continuous, frequency variable
}
wavein_param.samplerate = 0;
return BX_SOUNDLOW_OK;
}
@ -422,10 +375,10 @@ int bx_sound_alsa_c::getwavepacket(int length, Bit8u data[])
} else if (ret != (int)alsa_pcm[1].frames) {
BX_ERROR(("short read, read %d frames", ret));
}
memcpy(audio_buffer[1]+alsa_pcm[1].audio_bufsize, alsa_pcm[1].buffer, alsa_pcm[1].alsa_bufsize);
memcpy(audio_buffer+alsa_pcm[1].audio_bufsize, alsa_pcm[1].buffer, alsa_pcm[1].alsa_bufsize);
alsa_pcm[1].audio_bufsize += alsa_pcm[1].alsa_bufsize;
}
memcpy(data, audio_buffer[1], length);
memcpy(data, audio_buffer, length);
alsa_pcm[1].audio_bufsize -= length;
if ((alsa_pcm[1].audio_bufsize <= 0) && (alsa_pcm[1].buffer != NULL)) {
free(alsa_pcm[1].buffer);
@ -470,4 +423,14 @@ void bx_sound_alsa_c::record_timer(void)
record_handler(this, record_packet_size);
}
int bx_sound_alsa_c::register_wave_callback(void *arg, get_wave_cb_t wd_cb)
{
if (cb_count < BX_MAX_WAVE_CALLBACKS) {
get_wave[cb_count].device = arg;
get_wave[cb_count].cb = wd_cb;
return cb_count++;
}
return -1;
}
#endif

14
bochs/iodev/sound/soundalsa.h
View File

@ -42,8 +42,8 @@ public:
virtual int openwaveoutput(const char *wavedev);
virtual int set_pcm_params(bx_pcm_param_t param);
virtual int sendwavepacket(int length, Bit8u data[], bx_pcm_param_t *src_param);
virtual int stopwaveplayback();
virtual int get_waveout_packetsize();
virtual int waveout(int length, Bit8u data[]);
virtual int closewaveoutput();
virtual int openwaveinput(const char *wavedev, sound_record_handler_t rh);
@ -54,11 +54,12 @@ public:
static void record_timer_handler(void *);
void record_timer(void);
virtual int register_wave_callback(void *, get_wave_cb_t wd_cb);
private:
int alsa_seq_open(const char *alsadev);
int alsa_seq_output(int delta, int command, int length, Bit8u data[]);
int alsa_pcm_open(bx_bool input, int frequency, int bits, bx_bool stereo, int format);
int alsa_pcm_write();
struct {
snd_seq_t *handle;
@ -70,11 +71,8 @@ private:
int alsa_bufsize, audio_bufsize;
char *buffer;
} alsa_pcm[2];
struct {
int oldfreq, oldbits, oldformat;
bx_bool oldstereo;
} wave_ch[2];
Bit8u audio_buffer[2][BX_SOUND_ALSA_BUFSIZE];
bx_pcm_param_t wavein_param;
Bit8u audio_buffer[BX_SOUND_ALSA_BUFSIZE];
};
#endif