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Merge Olle's comment patch (bug #3097)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -1,9 +1,9 @@
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/*
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* Asterisk -- A telephony toolkit for Linux.
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*
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* Copyright (C) 2002, Linux Support Services
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* Copyright (C) 2002-2004, Digium, inc
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*
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* By Matthew Fredrickson <creslin@linux-support.net>
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* By Matthew Fredrickson <creslin@digium.com>
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License
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@@ -61,7 +61,7 @@
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#define MIN_SWITCH_TIME 600
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static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
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//static int block = O_NONBLOCK;
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/* static int block = O_NONBLOCK; */
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static char indevname[50] = ALSA_INDEV;
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static char outdevname[50] = ALSA_OUTDEV;
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@@ -284,14 +284,14 @@ static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
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snd_pcm_sw_params_t *swparams = NULL;
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struct pollfd pfd;
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snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
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//int period_bytes = 0;
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/* int period_bytes = 0; */
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snd_pcm_uframes_t buffer_size = 0;
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unsigned int rate = DESIRED_RATE;
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#if 0
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unsigned int per_min = 1;
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#endif
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//unsigned int per_max = 8;
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/* unsigned int per_max = 8; */
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snd_pcm_uframes_t start_threshold, stop_threshold;
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err = snd_pcm_open(&handle, dev, stream, O_NONBLOCK);
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@@ -334,7 +334,7 @@ static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
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ast_log(LOG_DEBUG, "Period size is %d\n", err);
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}
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buffer_size = 4096 * 2; //period_size * 16;
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buffer_size = 4096 * 2; /* period_size * 16; */
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err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
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if (err < 0) {
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ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err));
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@@ -542,7 +542,7 @@ static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
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int len = sizpos;
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int pos;
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int res = 0;
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//size_t frames = 0;
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/* size_t frames = 0; */
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snd_pcm_state_t state;
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/* Immediately return if no sound is enabled */
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if (nosound)
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