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Aegisub
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Code Issues Releases Wiki Activity

So apparently Alsa mostly works now. Not thorougly tested. 

Originally committed to SVN as r1146.
This commit is contained in:
Niels Martin Hansen 2007-04-26 20:11:48 +00:00
parent 2e82606c29
commit 7a89405f18
1 changed files with 17 additions and 6 deletions
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23
aegisub/audio_player_alsa.cpp
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@ -194,6 +194,11 @@ void AlsaPlayer::SetUpHardware()
throw _T("Could not set sample format"); throw _T("Could not set sample format");
} }
// Ask for resampling
if (snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1) < 0) {
throw _T("Couldn't enable resampling");
}
// Set sample rate // Set sample rate
rate = provider->GetSampleRate(); rate = provider->GetSampleRate();
real_rate = rate; real_rate = rate;
@ -208,6 +213,7 @@ void AlsaPlayer::SetUpHardware()
if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, provider->GetChannels()) < 0) { if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, provider->GetChannels()) < 0) {
throw _T("Could not set number of channels"); throw _T("Could not set number of channels");
} }
printf("Set sample rate %u (wanted %u)\n", real_rate, rate);
// Set buffer size // Set buffer size
unsigned int wanted_buflen = 1000000; // microseconds unsigned int wanted_buflen = 1000000; // microseconds
@ -260,7 +266,7 @@ void AlsaPlayer::SetUpAsync()
} }
// How full the buffer must be before playback begins // How full the buffer must be before playback begins
if (snd_pcm_sw_params_set_start_threshold(pcm_handle, sw_params, bufsize/2) < 0) { if (snd_pcm_sw_params_set_start_threshold(pcm_handle, sw_params, bufsize - period) < 0) {
throw _T("Failed setting start threshold"); throw _T("Failed setting start threshold");
} }
@ -391,7 +397,7 @@ __int64 AlsaPlayer::GetEndPosition()
// Get current position // Get current position
__int64 AlsaPlayer::GetCurrentPosition() __int64 AlsaPlayer::GetCurrentPosition()
{ {
return cur_frame; // FIXME return cur_frame - bufsize; // FIXME
} }
@ -409,6 +415,8 @@ void AlsaPlayer::async_write_handler(snd_async_handler_t *pcm_callback)
snd_pcm_sframes_t frames = snd_pcm_avail_update(player->pcm_handle); snd_pcm_sframes_t frames = snd_pcm_avail_update(player->pcm_handle);
// TODO: handle underrun
if (player->cur_frame >= player->end_frame) { if (player->cur_frame >= player->end_frame) {
// Past end of stream, add some silence // Past end of stream, add some silence
void *buf = calloc(frames, player->bpf); void *buf = calloc(frames, player->bpf);
@ -418,11 +426,14 @@ void AlsaPlayer::async_write_handler(snd_async_handler_t *pcm_callback)
return; return;
} }
void *buf = malloc(frames * player->bpf); void *buf = malloc(player->period * player->bpf);
player->provider->GetAudioWithVolume(buf, player->cur_frame, frames, player->volume); while (frames >= player->period) {
snd_pcm_writei(player->pcm_handle, buf, frames); player->provider->GetAudioWithVolume(buf, player->cur_frame, player->period, player->volume);
snd_pcm_writei(player->pcm_handle, buf, player->period);
player->cur_frame += player->period;
frames = snd_pcm_avail_update(player->pcm_handle);
}
free(buf); free(buf);
player->cur_frame += frames;
} }