remove some diagnostics and add in a comment -- using CLOCK_REALTIME with alsa libraries before 1.0.28
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+2
-7
@@ -114,7 +114,6 @@ int frame_size; // in bytes for interleaved stereo
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int alsa_device_initialised; // boolean to ensure the initialisation is only
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// done once
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snd_pcm_t *alsa_handle = NULL;
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int alsa_uses_monotonic_clock = 0;
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static snd_pcm_hw_params_t *alsa_params = NULL;
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static snd_pcm_sw_params_t *alsa_swparams = NULL;
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static snd_ctl_t *ctl = NULL;
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@@ -459,12 +458,6 @@ int actual_open_alsa_device(void) {
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return ret;
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}
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alsa_uses_monotonic_clock = snd_pcm_hw_params_is_monotonic(alsa_params);
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debug(1,"alsa_uses_monotonic_clock is %d",alsa_uses_monotonic_clock);
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debug (1,"sizeof time_t is: %d.", sizeof(time_t));
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debug (1,"sizeof long is: %d.", sizeof(long));
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ret = snd_pcm_sw_params_current(alsa_handle, alsa_swparams);
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if (ret < 0) {
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warn("audio_alsa: Unable to get current sw parameters for device \"%s\": "
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@@ -1158,6 +1151,8 @@ int delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay) {
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if (*state == SND_PCM_STATE_DRAINING)
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debug(1, "alsa: draining with a delay of %d.", delay);
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// It seems that the alsa library uses CLOCK_REALTIME before 1.0.28, even though
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// the check for monotonic returns true. Might have to watch out for this.
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#if SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR < 28
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clock_gettime(CLOCK_REALTIME, &tn);
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#else
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