add draft ALSA input plugin
I've created an elementary input plugin that plays sound from the soundcard, so you can use MPD to listen to anything connected to the line-in jack, or to Video4Linux FM radio cards that send audio through the soundcard. There has been a small number of posts here in the past requesting line-in input, so here is a first, simplistic stab at it. The patch adds a new sheme, alsa://, which causes mpd to play data read directly from a souncdard. It defaults to hw:0,0, but you can pass any ALSA device name in the URI. So, using mpc for example: mpc add alsa:// mpc play will play from device hw:0,0. To use a diffferent device: mpc add alsa://hw:1,0
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parent
e30fa7d15e
commit
35f85ddd86
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@ -802,6 +802,14 @@ INPUT_LIBS = \
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$(DESPOTIFY_LIBS) \
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$(DESPOTIFY_LIBS) \
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$(MMS_LIBS)
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$(MMS_LIBS)
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if HAVE_ALSA
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libinput_a_SOURCES += \
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src/input/AlsaInputPlugin.cxx \
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src/input/AlsaInputPlugin.hxx
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INPUT_LIBS += $(ALSA_LIBS)
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endif
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if ENABLE_CURL
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if ENABLE_CURL
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libinput_a_SOURCES += \
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libinput_a_SOURCES += \
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src/input/CurlInputPlugin.cxx src/input/CurlInputPlugin.hxx \
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src/input/CurlInputPlugin.cxx src/input/CurlInputPlugin.hxx \
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2
NEWS
2
NEWS
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@ -1,6 +1,8 @@
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ver 0.19 (not yet released)
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ver 0.19 (not yet released)
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* protocol
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* protocol
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- new commands "addtagid", "cleartagid"
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- new commands "addtagid", "cleartagid"
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* input
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- alsa: new input plugin
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* new resampler option using libsoxr
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* new resampler option using libsoxr
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ver 0.18.6 (not yet released)
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ver 0.18.6 (not yet released)
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@ -22,6 +22,10 @@
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#include "util/Macros.hxx"
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#include "util/Macros.hxx"
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#include "input/FileInputPlugin.hxx"
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#include "input/FileInputPlugin.hxx"
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#ifdef HAVE_ALSA
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#include "input/AlsaInputPlugin.hxx"
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#endif
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#ifdef ENABLE_ARCHIVE
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#ifdef ENABLE_ARCHIVE
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#include "input/ArchiveInputPlugin.hxx"
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#include "input/ArchiveInputPlugin.hxx"
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#endif
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#endif
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@ -48,6 +52,9 @@
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const InputPlugin *const input_plugins[] = {
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const InputPlugin *const input_plugins[] = {
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&input_plugin_file,
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&input_plugin_file,
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#ifdef HAVE_ALSA
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&input_plugin_alsa,
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#endif
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#ifdef ENABLE_ARCHIVE
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#ifdef ENABLE_ARCHIVE
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&input_plugin_archive,
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&input_plugin_archive,
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#endif
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#endif
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@ -0,0 +1,206 @@
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/*
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* Copyright (C) 2003-2013 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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/*
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* ALSA code based on an example by Paul Davis released under GPL here:
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* http://equalarea.com/paul/alsa-audio.html
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* and one by Matthias Nagorni, also GPL, here:
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* http://alsamodular.sourceforge.net/alsa_programming_howto.html
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*/
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#include "config.h"
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#include "AlsaInputPlugin.hxx"
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#include "InputPlugin.hxx"
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#include "InputStream.hxx"
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#include "util/Domain.hxx"
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#include "util/Error.hxx"
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#include "util/StringUtil.hxx"
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#include "Log.hxx"
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#include <alsa/asoundlib.h>
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static constexpr Domain alsa_input_domain("alsa");
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static constexpr const char *default_device = "hw:0,0";
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// this value chosen to balance between limiting latency and avoiding stutter
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static constexpr int max_frames_to_buffer = 64;
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// the following defaults are because the PcmDecoderPlugin forces CD format
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static constexpr snd_pcm_format_t default_format = SND_PCM_FORMAT_S16;
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static constexpr int default_channels = 2; // stereo
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static constexpr unsigned int default_rate = 44100; // cd quality
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struct AlsaInputStream {
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InputStream base;
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snd_pcm_t *capture_handle;
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size_t frame_size;
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size_t max_bytes_to_read;
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AlsaInputStream(const char *uri, Mutex &mutex, Cond &cond,
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snd_pcm_t *handle)
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:base(input_plugin_alsa, uri, mutex, cond),
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capture_handle(handle) {
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frame_size = snd_pcm_format_width(default_format) / 8 * default_channels;
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max_bytes_to_read = max_frames_to_buffer * frame_size;
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base.mime = strdup("audio/x-mpd-cdda-pcm");
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base.seekable = false;
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base.size = -1;
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base.ready = true;
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}
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~AlsaInputStream() {
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snd_pcm_close(capture_handle);
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}
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};
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static InputStream *
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alsa_input_open(const char *uri, Mutex &mutex, Cond &cond, Error &error)
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{
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int err;
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// check uri is appropriate for alsa input
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if (!StringStartsWith(uri, "alsa://"))
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return nullptr;
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const char *device = uri + 7;
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if (device[0] == '\0')
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device = default_device;
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snd_pcm_t *capture_handle;
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if ((err = snd_pcm_open(&capture_handle, device,
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SND_PCM_STREAM_CAPTURE, 0)) < 0) {
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error.Format(alsa_input_domain, "Failed to open device: %s (%s)", device, snd_strerror(err));
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return nullptr;
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}
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snd_pcm_hw_params_t *hw_params;
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if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
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error.Format(alsa_input_domain, "Cannot allocate hardware parameter structure (%s)", snd_strerror(err));
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snd_pcm_close(capture_handle);
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return nullptr;
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}
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if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {
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error.Format(alsa_input_domain, "Cannot initialize hardware parameter structure (%s)", snd_strerror(err));
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snd_pcm_hw_params_free(hw_params);
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snd_pcm_close(capture_handle);
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return nullptr;
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}
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if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
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error.Format(alsa_input_domain, "Cannot set access type (%s)", snd_strerror (err));
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snd_pcm_hw_params_free (hw_params);
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snd_pcm_close(capture_handle);
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return nullptr;
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}
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if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params, default_format)) < 0) {
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snd_pcm_hw_params_free(hw_params);
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snd_pcm_close(capture_handle);
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error.Format(alsa_input_domain, "Cannot set sample format (%s)", snd_strerror (err));
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return nullptr;
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}
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if ((err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, default_channels)) < 0) {
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snd_pcm_hw_params_free(hw_params);
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snd_pcm_close(capture_handle);
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error.Format(alsa_input_domain, "Cannot set channels (%s)", snd_strerror (err));
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return nullptr;
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}
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if ((err = snd_pcm_hw_params_set_rate(capture_handle, hw_params, default_rate, 0)) < 0) {
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snd_pcm_hw_params_free(hw_params);
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snd_pcm_close(capture_handle);
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error.Format(alsa_input_domain, "Cannot set sample rate (%s)", snd_strerror (err));
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return nullptr;
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}
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if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {
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error.Format(alsa_input_domain, "Cannot set parameters (%s)", snd_strerror (err));
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snd_pcm_hw_params_free(hw_params);
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snd_pcm_close(capture_handle);
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return nullptr;
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}
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snd_pcm_hw_params_free (hw_params);
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// clear any data already in the PCM buffer
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if ((err = snd_pcm_drop(capture_handle)) < 0) {
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error.Format(alsa_input_domain, "Cannot clear PCM buffer (%s)", snd_strerror (err));
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snd_pcm_hw_params_free(hw_params);
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snd_pcm_close(capture_handle);
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return nullptr;
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}
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AlsaInputStream *ais = new AlsaInputStream(uri, mutex, cond, capture_handle);
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return &ais->base;
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}
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static void
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alsa_input_close(InputStream *is)
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{
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AlsaInputStream *ais = (AlsaInputStream*) is;
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delete ais;
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}
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static size_t
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alsa_input_read(InputStream *is, void *ptr, size_t size,
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gcc_unused Error &error)
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{
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AlsaInputStream *ais = (AlsaInputStream*) is;
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int num_frames = max_frames_to_buffer;
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if (size < ais->max_bytes_to_read)
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// calculate number of whole frames that will fit in size bytes
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num_frames = size / ais->frame_size;
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int ret;
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while ((ret = snd_pcm_readi(ais->capture_handle, ptr,
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num_frames)) < 0) {
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snd_pcm_prepare(ais->capture_handle);
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LogDebug(alsa_input_domain, "Buffer Overrun");
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}
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size_t nbytes = ret == max_frames_to_buffer
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? ais->max_bytes_to_read
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: ret * ais->frame_size;
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is->offset += nbytes;
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return nbytes;
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}
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static bool
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alsa_input_eof(gcc_unused InputStream *is)
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{
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return false;
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};
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const struct InputPlugin input_plugin_alsa = {
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.name = "alsa",
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.init = nullptr,
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.finish = nullptr,
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.open = alsa_input_open,
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.close = alsa_input_close,
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.check = nullptr,
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.update = nullptr,
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.tag = nullptr,
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.available = nullptr,
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.read = alsa_input_read,
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.eof = alsa_input_eof,
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.seek = nullptr
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};
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@ -0,0 +1,28 @@
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/*
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* Copyright (C) 2003-2013 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#ifndef MPD_ALSA_INPUT_PLUGIN_HXX
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#define MPD_ALSA_INPUT_PLUGIN_HXX
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#include "InputPlugin.hxx"
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extern const struct InputPlugin input_plugin_alsa;
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#endif
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@ -54,6 +54,9 @@ static const char *remoteUrlPrefixes[] = {
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#endif
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#endif
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#ifdef ENABLE_DESPOTIFY
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#ifdef ENABLE_DESPOTIFY
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"spt://",
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"spt://",
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#endif
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#ifdef HAVE_ALSA
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"alsa://",
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#endif
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#endif
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NULL
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NULL
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};
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};
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