384 lines
10 KiB
C++
384 lines
10 KiB
C++
/*
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* Copyright (C) 2003-2014 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 "util/ReusableArray.hxx"
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#include "Log.hxx"
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#include "event/MultiSocketMonitor.hxx"
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#include "event/DeferredMonitor.hxx"
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#include "event/Call.hxx"
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#include "thread/Mutex.hxx"
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#include "thread/Cond.hxx"
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#include "IOThread.hxx"
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#include <alsa/asoundlib.h>
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#include <assert.h>
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#include <string.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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// 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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/**
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* This value should be the same as the read buffer size defined in
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* PcmDecoderPlugin.cxx:pcm_stream_decode().
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* We use it to calculate how many audio frames to buffer in the alsa driver
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* before reading from the device. snd_pcm_readi() blocks until that many
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* frames are ready.
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*/
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static constexpr size_t read_buffer_size = 4096;
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class AlsaInputStream final
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: public InputStream,
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MultiSocketMonitor, DeferredMonitor {
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snd_pcm_t *capture_handle;
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size_t frame_size;
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int frames_to_read;
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bool eof;
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/**
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* Is somebody waiting for data? This is set by method
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* Available().
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*/
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std::atomic_bool waiting;
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ReusableArray<pollfd> pfd_buffer;
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public:
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AlsaInputStream(EventLoop &loop,
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const char *_uri, Mutex &_mutex, Cond &_cond,
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snd_pcm_t *_handle, int _frame_size)
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:InputStream(_uri, _mutex, _cond),
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MultiSocketMonitor(loop),
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DeferredMonitor(loop),
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capture_handle(_handle),
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frame_size(_frame_size),
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eof(false)
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{
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assert(_uri != nullptr);
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assert(_handle != nullptr);
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/* this mime type forces use of the PcmDecoderPlugin.
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Needs to be generalised when/if that decoder is
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updated to support other audio formats */
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SetMimeType("audio/x-mpd-cdda-pcm");
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InputStream::SetReady();
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frames_to_read = read_buffer_size / frame_size;
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snd_pcm_start(capture_handle);
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DeferredMonitor::Schedule();
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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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using DeferredMonitor::GetEventLoop;
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static InputStream *Create(const char *uri, Mutex &mutex, Cond &cond,
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Error &error);
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/* virtual methods from InputStream */
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bool IsEOF() override {
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return eof;
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}
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bool IsAvailable() override {
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if (snd_pcm_avail(capture_handle) > frames_to_read)
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return true;
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if (!waiting.exchange(true))
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SafeInvalidateSockets();
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return false;
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}
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size_t Read(void *ptr, size_t size, Error &error) override;
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private:
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static snd_pcm_t *OpenDevice(const char *device, int rate,
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snd_pcm_format_t format, int channels,
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Error &error);
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int Recover(int err);
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void SafeInvalidateSockets() {
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DeferredMonitor::Schedule();
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}
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virtual void RunDeferred() override {
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InvalidateSockets();
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}
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virtual int PrepareSockets() override;
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virtual void DispatchSockets() override;
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};
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inline InputStream *
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AlsaInputStream::Create(const char *uri, Mutex &mutex, Cond &cond,
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Error &error)
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{
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const char *const scheme = "alsa://";
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if (!StringStartsWith(uri, scheme))
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return nullptr;
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const char *device = uri + strlen(scheme);
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if (strlen(device) == 0)
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device = default_device;
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/* placeholders - eventually user-requested audio format will
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be passed via the URI. For now we just force the
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defaults */
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int rate = default_rate;
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snd_pcm_format_t format = default_format;
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int channels = default_channels;
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snd_pcm_t *handle = OpenDevice(device, rate, format, channels,
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error);
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if (handle == nullptr)
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return nullptr;
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int frame_size = snd_pcm_format_width(format) / 8 * channels;
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return new AlsaInputStream(io_thread_get(),
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uri, mutex, cond,
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handle, frame_size);
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}
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size_t
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AlsaInputStream::Read(void *ptr, size_t read_size, Error &error)
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{
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assert(ptr != nullptr);
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int num_frames = read_size / frame_size;
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int ret;
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while ((ret = snd_pcm_readi(capture_handle, ptr, num_frames)) < 0) {
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if (Recover(ret) < 0) {
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eof = true;
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error.Format(alsa_input_domain,
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"PCM error - stream aborted");
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return 0;
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}
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}
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size_t nbytes = ret * frame_size;
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offset += nbytes;
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return nbytes;
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}
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int
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AlsaInputStream::PrepareSockets()
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{
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if (!waiting) {
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ClearSocketList();
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return -1;
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}
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int count = snd_pcm_poll_descriptors_count(capture_handle);
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if (count < 0) {
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ClearSocketList();
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return -1;
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}
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struct pollfd *pfds = pfd_buffer.Get(count);
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count = snd_pcm_poll_descriptors(capture_handle, pfds, count);
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if (count < 0)
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count = 0;
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ReplaceSocketList(pfds, count);
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return -1;
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}
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void
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AlsaInputStream::DispatchSockets()
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{
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waiting = false;
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const ScopeLock protect(mutex);
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/* wake up the thread that is waiting for more data */
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cond.broadcast();
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}
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inline int
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AlsaInputStream::Recover(int err)
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{
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switch(err) {
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case -EPIPE:
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LogDebug(alsa_input_domain, "Buffer Overrun");
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// drop through
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case -ESTRPIPE:
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case -EINTR:
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err = snd_pcm_recover(capture_handle, err, 1);
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break;
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default:
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// something broken somewhere, give up
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err = -1;
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}
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return err;
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}
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inline snd_pcm_t *
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AlsaInputStream::OpenDevice(const char *device,
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int rate, snd_pcm_format_t format, int channels,
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Error &error)
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{
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snd_pcm_t *capture_handle;
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int err;
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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, 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, 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, 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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/* period needs to be big enough so that poll() doesn't fire too often,
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* but small enough that buffer overruns don't occur if Read() is not
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* invoked often enough.
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* the calculation here is empirical; however all measurements were
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* done using 44100:16:2. When we extend this plugin to support
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* other audio formats then this may need to be revisited */
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snd_pcm_uframes_t period = read_buffer_size * 2;
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int direction = -1;
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if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params,
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&period, &direction)) < 0) {
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error.Format(alsa_input_domain, "Cannot set period size (%s)",
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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(capture_handle, hw_params)) < 0) {
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error.Format(alsa_input_domain, "Cannot set parameters (%s)",
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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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snd_pcm_sw_params_t *sw_params;
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snd_pcm_sw_params_malloc(&sw_params);
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snd_pcm_sw_params_current(capture_handle, sw_params);
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if ((err = snd_pcm_sw_params_set_start_threshold(capture_handle, sw_params,
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period)) < 0) {
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error.Format(alsa_input_domain,
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"unable to set start threshold (%s)", snd_strerror(err));
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snd_pcm_sw_params_free(sw_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_sw_params(capture_handle, sw_params)) < 0) {
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error.Format(alsa_input_domain,
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"unable to install sw params (%s)", snd_strerror(err));
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snd_pcm_sw_params_free(sw_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_sw_params_free(sw_params);
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snd_pcm_prepare(capture_handle);
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return capture_handle;
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}
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/*######################### Plugin Functions ##############################*/
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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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return AlsaInputStream::Create(uri, mutex, cond, error);
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}
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const struct InputPlugin input_plugin_alsa = {
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"alsa",
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nullptr,
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nullptr,
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alsa_input_open,
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};
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