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4e9a8e5883
This reverts commit 4d67d1bd16
.
Using the extended API for the capture path depends on a fix in
PulseAudio
(https://gitlab.freedesktop.org/pulseaudio/pulseaudio/merge_requests/49).
Until then, let's go back to the standard API.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/552
97 lines
3.4 KiB
C
97 lines
3.4 KiB
C
/*
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* GStreamer pulseaudio plugin
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*
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* Copyright (c) 2004-2008 Lennart Poettering
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*
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* gst-pulse is free software; you can redistribute it and/or modify
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* it under the terms of the GNU Lesser General Public License as
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* published by the Free Software Foundation; either version 2.1 of the
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* License, or (at your option) any later version.
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*
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* gst-pulse is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with gst-pulse; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301
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* USA.
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*/
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#ifndef __GST_PULSEUTIL_H__
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#define __GST_PULSEUTIL_H__
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <pulse/pulseaudio.h>
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#include <gst/audio/gstaudioringbuffer.h>
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#include <gst/audio/gstaudiosink.h>
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#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
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# define _PULSE_FORMATS "{ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, " \
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"S24LE, S24BE, S24_32LE, S24_32BE, U8 }"
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#else
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# define _PULSE_FORMATS "{ S16BE, S16LE, F32BE, F32LE, S32BE, S32LE, " \
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"S24BE, S24LE, S24_32BE, S24_32LE, U8 }"
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#endif
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/* NOTE! that we do NOT actually support rate=MAX. This must be fixed up using
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* gst_pulse_fix_pcm_caps() before being used. */
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#define _PULSE_CAPS_LINEAR \
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"audio/x-raw, " \
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"format = (string) " _PULSE_FORMATS ", " \
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"layout = (string) interleaved, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 32 ]; "
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#define _PULSE_CAPS_ALAW \
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"audio/x-alaw, " \
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"rate = (int) [ 1, MAX], " \
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"channels = (int) [ 1, 32 ]; "
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#define _PULSE_CAPS_MULAW \
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"audio/x-mulaw, " \
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"rate = (int) [ 1, MAX], " \
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"channels = (int) [ 1, 32 ]; "
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#define _PULSE_CAPS_AC3 "audio/x-ac3, framed = (boolean) true; "
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#define _PULSE_CAPS_EAC3 "audio/x-eac3, framed = (boolean) true; "
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#define _PULSE_CAPS_DTS "audio/x-dts, framed = (boolean) true, " \
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"block-size = (int) { 512, 1024, 2048 }; "
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#define _PULSE_CAPS_MP3 "audio/mpeg, mpegversion = (int) 1, " \
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"mpegaudioversion = (int) [ 1, 3 ], parsed = (boolean) true;"
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#define _PULSE_CAPS_AAC "audio/mpeg, mpegversion = (int) { 2, 4 }, " \
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"framed = (boolean) true, stream-format = (string) adts;"
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#define _PULSE_CAPS_PCM \
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_PULSE_CAPS_LINEAR \
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_PULSE_CAPS_ALAW \
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_PULSE_CAPS_MULAW
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gboolean gst_pulse_fill_sample_spec (GstAudioRingBufferSpec * spec,
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pa_sample_spec * ss);
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gboolean gst_pulse_fill_format_info (GstAudioRingBufferSpec * spec,
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pa_format_info ** f, guint * channels);
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const char * gst_pulse_sample_format_to_caps_format (pa_sample_format_t sf);
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gchar *gst_pulse_client_name (void);
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pa_channel_map *gst_pulse_gst_to_channel_map (pa_channel_map * map,
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const GstAudioRingBufferSpec * spec);
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GstAudioRingBufferSpec *gst_pulse_channel_map_to_gst (const pa_channel_map * map,
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GstAudioRingBufferSpec * spec);
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void gst_pulse_cvolume_from_linear (pa_cvolume *v, unsigned channels, gdouble volume);
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pa_proplist *gst_pulse_make_proplist (const GstStructure *properties);
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GstStructure *gst_pulse_make_structure (pa_proplist *properties);
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GstCaps * gst_pulse_format_info_to_caps (pa_format_info * format);
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GstCaps * gst_pulse_fix_pcm_caps (GstCaps * incaps);
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#endif
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