Implemented support for the new mixer thread in the lowlevel sound module
'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
parent
bfb5ec8cf9
commit
9c1070cbed
@ -27,6 +27,10 @@
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#if BX_HAVE_ALSASOUND && BX_SUPPORT_SOUNDLOW
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#ifndef WIN32
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#include <pthread.h>
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#endif
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#define LOG_THIS
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bx_sound_alsa_c::bx_sound_alsa_c()
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@ -202,6 +206,9 @@ int bx_sound_alsa_c::closemidioutput()
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int bx_sound_alsa_c::openwaveoutput(const char *wavedev)
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{
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set_pcm_params(real_pcm_param);
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pcm_callback_id = register_wave_callback(this, pcm_callback);
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BX_INIT_MUTEX(mixer_mutex);
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start_mixer_thread();
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return BX_SOUNDLOW_OK;
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}
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@ -216,7 +223,7 @@ int bx_sound_alsa_c::alsa_pcm_open(bx_bool mode, int frequency, int bits, bx_boo
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alsa_pcm[mode].audio_bufsize = 0;
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if (alsa_pcm[mode].handle == NULL) {
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ret = snd_pcm_open(&alsa_pcm[mode].handle, "default", mode ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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ret = snd_pcm_open(&alsa_pcm[mode].handle, "default", mode ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 0);
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if (ret < 0) {
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return BX_SOUNDLOW_ERR;
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}
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@ -226,16 +233,18 @@ int bx_sound_alsa_c::alsa_pcm_open(bx_bool mode, int frequency, int bits, bx_boo
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snd_pcm_hw_params_any(alsa_pcm[mode].handle, params);
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snd_pcm_hw_params_set_access(alsa_pcm[mode].handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
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if ((frequency == wave_ch[mode].oldfreq) &&
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(bits == wave_ch[mode].oldbits) &&
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(stereo == wave_ch[mode].oldstereo) &&
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(format == wave_ch[mode].oldformat))
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return BX_SOUNDLOW_OK; // nothing to do
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if (mode == 1) {
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if ((frequency == wavein_param.samplerate) &&
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(bits == wavein_param.bits) &&
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(stereo == (wavein_param.channels == 2)) &&
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(format == wavein_param.format))
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return BX_SOUNDLOW_OK; // nothing to do
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wave_ch[mode].oldfreq = frequency;
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wave_ch[mode].oldbits = bits;
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wave_ch[mode].oldstereo = stereo;
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wave_ch[mode].oldformat = format;
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wavein_param.samplerate = frequency;
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wavein_param.bits = bits;
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wavein_param.channels = stereo + 1;
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wavein_param.format = format;
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}
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freq = (unsigned int)frequency;
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@ -283,82 +292,26 @@ int bx_sound_alsa_c::set_pcm_params(bx_pcm_param_t param)
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return alsa_pcm_open(0, param.samplerate, param.bits, param.channels == 2, param.format);
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}
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int bx_sound_alsa_c::alsa_pcm_write()
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int bx_sound_alsa_c::get_waveout_packetsize()
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{
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int ret;
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if (alsa_pcm[0].buffer == NULL) {
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alsa_pcm[0].buffer = (char *)malloc(alsa_pcm[0].alsa_bufsize);
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}
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while (alsa_pcm[0].audio_bufsize >= alsa_pcm[0].alsa_bufsize) {
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memcpy(alsa_pcm[0].buffer, audio_buffer[0], alsa_pcm[0].alsa_bufsize);
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ret = snd_pcm_writei(alsa_pcm[0].handle, alsa_pcm[0].buffer, alsa_pcm[0].frames);
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if (ret == -EAGAIN)
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continue;
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if (ret == -EPIPE) {
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/* EPIPE means underrun */
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BX_ERROR(("ALSA: underrun occurred"));
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snd_pcm_prepare(alsa_pcm[0].handle);
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} else if (ret < 0) {
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BX_ERROR(("ALSA: error from writei: %s", snd_strerror(ret)));
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} else if (ret != (int)alsa_pcm[0].frames) {
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BX_ERROR(("ALSA: short write, write %d frames", ret));
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}
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alsa_pcm[0].audio_bufsize -= alsa_pcm[0].alsa_bufsize;
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memmove(audio_buffer[0], audio_buffer[0]+alsa_pcm[0].alsa_bufsize, alsa_pcm[0].audio_bufsize);
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}
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if ((alsa_pcm[0].audio_bufsize == 0) && (alsa_pcm[0].buffer != NULL)) {
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free(alsa_pcm[0].buffer);
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alsa_pcm[0].buffer = NULL;
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}
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return BX_SOUNDLOW_OK;
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return alsa_pcm[0].alsa_bufsize;
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}
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int bx_sound_alsa_c::sendwavepacket(int length, Bit8u data[], bx_pcm_param_t *src_param)
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int bx_sound_alsa_c::waveout(int length, Bit8u data[])
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{
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int len2;
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if (memcmp(src_param, &emu_pcm_param, sizeof(bx_pcm_param_t)) != 0) {
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emu_pcm_param = *src_param;
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cvt_mult = (src_param->bits == 8) ? 2 : 1;
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if (src_param->channels == 1) cvt_mult <<= 1;
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if (src_param->samplerate != real_pcm_param.samplerate) {
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real_pcm_param.samplerate = src_param->samplerate;
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set_pcm_params(real_pcm_param);
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}
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if (!alsa_pcm[0].handle || (length > alsa_pcm[0].alsa_bufsize)) {
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return BX_SOUNDLOW_ERR;
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}
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len2 = length * cvt_mult;
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if (!alsa_pcm[0].handle) {
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// Alert indicating that caller is probably erroneous
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BX_ERROR(("sendwavepacket(): pcm is not open"));
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return BX_SOUNDLOW_ERR;
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int ret = snd_pcm_writei(alsa_pcm[0].handle, data, alsa_pcm[0].frames);
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if (ret == -EPIPE) {
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/* EPIPE means underrun */
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BX_ERROR(("ALSA: underrun occurred"));
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snd_pcm_prepare(alsa_pcm[0].handle);
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} else if (ret < 0) {
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BX_ERROR(("ALSA: error from writei: %s", snd_strerror(ret)));
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} else if (ret != (int)alsa_pcm[0].frames) {
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BX_ERROR(("ALSA: short write, write %d frames", ret));
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}
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if ((alsa_pcm[0].audio_bufsize+len2) > BX_SOUND_ALSA_BUFSIZE) {
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len2 = BX_SOUND_ALSA_BUFSIZE - alsa_pcm[0].audio_bufsize;
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BX_ERROR(("ALSA: audio buffer overflow"));
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}
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if (len2 > 0) {
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convert_pcm_data(data, length, audio_buffer[0]+alsa_pcm[0].audio_bufsize, len2, src_param);
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alsa_pcm[0].audio_bufsize += len2;
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}
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if (alsa_pcm[0].audio_bufsize < alsa_pcm[0].alsa_bufsize) {
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return BX_SOUNDLOW_OK;
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} else {
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return alsa_pcm_write();
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}
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}
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int bx_sound_alsa_c::stopwaveplayback()
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{
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if (alsa_pcm[0].handle && alsa_pcm[0].audio_bufsize > 0) {
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if (alsa_pcm[0].audio_bufsize < alsa_pcm[0].alsa_bufsize) {
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memset(audio_buffer[0]+alsa_pcm[0].audio_bufsize, 0, alsa_pcm[0].alsa_bufsize-alsa_pcm[0].audio_bufsize);
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alsa_pcm[0].audio_bufsize = alsa_pcm[0].alsa_bufsize;
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}
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alsa_pcm_write();
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}
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return BX_SOUNDLOW_OK;
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}
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@ -369,7 +322,6 @@ int bx_sound_alsa_c::closewaveoutput()
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snd_pcm_close(alsa_pcm[0].handle);
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alsa_pcm[0].handle = NULL;
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}
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return BX_SOUNDLOW_OK;
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}
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@ -380,6 +332,7 @@ int bx_sound_alsa_c::openwaveinput(const char *wavedev, sound_record_handler_t r
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record_timer_index = bx_pc_system.register_timer(this, record_timer_handler, 1, 1, 0, "soundlnx");
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// record timer: inactive, continuous, frequency variable
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}
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wavein_param.samplerate = 0;
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return BX_SOUNDLOW_OK;
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}
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@ -422,10 +375,10 @@ int bx_sound_alsa_c::getwavepacket(int length, Bit8u data[])
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} else if (ret != (int)alsa_pcm[1].frames) {
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BX_ERROR(("short read, read %d frames", ret));
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}
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memcpy(audio_buffer[1]+alsa_pcm[1].audio_bufsize, alsa_pcm[1].buffer, alsa_pcm[1].alsa_bufsize);
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memcpy(audio_buffer+alsa_pcm[1].audio_bufsize, alsa_pcm[1].buffer, alsa_pcm[1].alsa_bufsize);
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alsa_pcm[1].audio_bufsize += alsa_pcm[1].alsa_bufsize;
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}
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memcpy(data, audio_buffer[1], length);
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memcpy(data, audio_buffer, length);
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alsa_pcm[1].audio_bufsize -= length;
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if ((alsa_pcm[1].audio_bufsize <= 0) && (alsa_pcm[1].buffer != NULL)) {
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free(alsa_pcm[1].buffer);
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@ -470,4 +423,14 @@ void bx_sound_alsa_c::record_timer(void)
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record_handler(this, record_packet_size);
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}
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int bx_sound_alsa_c::register_wave_callback(void *arg, get_wave_cb_t wd_cb)
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{
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if (cb_count < BX_MAX_WAVE_CALLBACKS) {
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get_wave[cb_count].device = arg;
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get_wave[cb_count].cb = wd_cb;
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return cb_count++;
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}
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return -1;
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}
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#endif
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@ -42,8 +42,8 @@ public:
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virtual int openwaveoutput(const char *wavedev);
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virtual int set_pcm_params(bx_pcm_param_t param);
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virtual int sendwavepacket(int length, Bit8u data[], bx_pcm_param_t *src_param);
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virtual int stopwaveplayback();
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virtual int get_waveout_packetsize();
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virtual int waveout(int length, Bit8u data[]);
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virtual int closewaveoutput();
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virtual int openwaveinput(const char *wavedev, sound_record_handler_t rh);
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@ -54,11 +54,12 @@ public:
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static void record_timer_handler(void *);
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void record_timer(void);
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virtual int register_wave_callback(void *, get_wave_cb_t wd_cb);
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private:
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int alsa_seq_open(const char *alsadev);
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int alsa_seq_output(int delta, int command, int length, Bit8u data[]);
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int alsa_pcm_open(bx_bool input, int frequency, int bits, bx_bool stereo, int format);
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int alsa_pcm_write();
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struct {
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snd_seq_t *handle;
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@ -70,11 +71,8 @@ private:
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int alsa_bufsize, audio_bufsize;
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char *buffer;
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} alsa_pcm[2];
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struct {
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int oldfreq, oldbits, oldformat;
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bx_bool oldstereo;
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} wave_ch[2];
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Bit8u audio_buffer[2][BX_SOUND_ALSA_BUFSIZE];
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bx_pcm_param_t wavein_param;
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Bit8u audio_buffer[BX_SOUND_ALSA_BUFSIZE];
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};
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#endif
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