aboutsummaryrefslogtreecommitdiffstats
path: root/src/input/plugins/AlsaInputPlugin.cxx
diff options
context:
space:
mode:
Diffstat (limited to 'src/input/plugins/AlsaInputPlugin.cxx')
-rw-r--r--src/input/plugins/AlsaInputPlugin.cxx383
1 files changed, 383 insertions, 0 deletions
diff --git a/src/input/plugins/AlsaInputPlugin.cxx b/src/input/plugins/AlsaInputPlugin.cxx
new file mode 100644
index 000000000..82b96f7df
--- /dev/null
+++ b/src/input/plugins/AlsaInputPlugin.cxx
@@ -0,0 +1,383 @@
+/*
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+/*
+ * ALSA code based on an example by Paul Davis released under GPL here:
+ * http://equalarea.com/paul/alsa-audio.html
+ * and one by Matthias Nagorni, also GPL, here:
+ * http://alsamodular.sourceforge.net/alsa_programming_howto.html
+ */
+
+#include "config.h"
+#include "AlsaInputPlugin.hxx"
+#include "../InputPlugin.hxx"
+#include "../InputStream.hxx"
+#include "util/Domain.hxx"
+#include "util/Error.hxx"
+#include "util/StringUtil.hxx"
+#include "util/ReusableArray.hxx"
+
+#include "Log.hxx"
+#include "event/MultiSocketMonitor.hxx"
+#include "event/DeferredMonitor.hxx"
+#include "event/Call.hxx"
+#include "thread/Mutex.hxx"
+#include "thread/Cond.hxx"
+#include "IOThread.hxx"
+
+#include <alsa/asoundlib.h>
+
+#include <assert.h>
+#include <string.h>
+
+static constexpr Domain alsa_input_domain("alsa");
+
+static constexpr const char *default_device = "hw:0,0";
+
+// the following defaults are because the PcmDecoderPlugin forces CD format
+static constexpr snd_pcm_format_t default_format = SND_PCM_FORMAT_S16;
+static constexpr int default_channels = 2; // stereo
+static constexpr unsigned int default_rate = 44100; // cd quality
+
+/**
+ * This value should be the same as the read buffer size defined in
+ * PcmDecoderPlugin.cxx:pcm_stream_decode().
+ * We use it to calculate how many audio frames to buffer in the alsa driver
+ * before reading from the device. snd_pcm_readi() blocks until that many
+ * frames are ready.
+ */
+static constexpr size_t read_buffer_size = 4096;
+
+class AlsaInputStream final
+ : public InputStream,
+ MultiSocketMonitor, DeferredMonitor {
+ snd_pcm_t *capture_handle;
+ size_t frame_size;
+ int frames_to_read;
+ bool eof;
+
+ /**
+ * Is somebody waiting for data? This is set by method
+ * Available().
+ */
+ std::atomic_bool waiting;
+
+ ReusableArray<pollfd> pfd_buffer;
+
+public:
+ AlsaInputStream(EventLoop &loop,
+ const char *_uri, Mutex &_mutex, Cond &_cond,
+ snd_pcm_t *_handle, int _frame_size)
+ :InputStream(_uri, _mutex, _cond),
+ MultiSocketMonitor(loop),
+ DeferredMonitor(loop),
+ capture_handle(_handle),
+ frame_size(_frame_size),
+ eof(false)
+ {
+ assert(_uri != nullptr);
+ assert(_handle != nullptr);
+
+ /* this mime type forces use of the PcmDecoderPlugin.
+ Needs to be generalised when/if that decoder is
+ updated to support other audio formats */
+ SetMimeType("audio/x-mpd-cdda-pcm");
+ InputStream::SetReady();
+
+ frames_to_read = read_buffer_size / frame_size;
+
+ snd_pcm_start(capture_handle);
+
+ DeferredMonitor::Schedule();
+ }
+
+ ~AlsaInputStream() {
+ snd_pcm_close(capture_handle);
+ }
+
+ using DeferredMonitor::GetEventLoop;
+
+ static InputStream *Create(const char *uri, Mutex &mutex, Cond &cond,
+ Error &error);
+
+ /* virtual methods from InputStream */
+
+ bool IsEOF() override {
+ return eof;
+ }
+
+ bool IsAvailable() override {
+ if (snd_pcm_avail(capture_handle) > frames_to_read)
+ return true;
+
+ if (!waiting.exchange(true))
+ SafeInvalidateSockets();
+
+ return false;
+ }
+
+ size_t Read(void *ptr, size_t size, Error &error) override;
+
+private:
+ static snd_pcm_t *OpenDevice(const char *device, int rate,
+ snd_pcm_format_t format, int channels,
+ Error &error);
+
+ int Recover(int err);
+
+ void SafeInvalidateSockets() {
+ DeferredMonitor::Schedule();
+ }
+
+ virtual void RunDeferred() override {
+ InvalidateSockets();
+ }
+
+ virtual int PrepareSockets() override;
+ virtual void DispatchSockets() override;
+};
+
+inline InputStream *
+AlsaInputStream::Create(const char *uri, Mutex &mutex, Cond &cond,
+ Error &error)
+{
+ const char *const scheme = "alsa://";
+ if (!StringStartsWith(uri, scheme))
+ return nullptr;
+
+ const char *device = uri + strlen(scheme);
+ if (strlen(device) == 0)
+ device = default_device;
+
+ /* placeholders - eventually user-requested audio format will
+ be passed via the URI. For now we just force the
+ defaults */
+ int rate = default_rate;
+ snd_pcm_format_t format = default_format;
+ int channels = default_channels;
+
+ snd_pcm_t *handle = OpenDevice(device, rate, format, channels,
+ error);
+ if (handle == nullptr)
+ return nullptr;
+
+ int frame_size = snd_pcm_format_width(format) / 8 * channels;
+ return new AlsaInputStream(io_thread_get(),
+ uri, mutex, cond,
+ handle, frame_size);
+}
+
+size_t
+AlsaInputStream::Read(void *ptr, size_t read_size, Error &error)
+{
+ assert(ptr != nullptr);
+
+ int num_frames = read_size / frame_size;
+ int ret;
+ while ((ret = snd_pcm_readi(capture_handle, ptr, num_frames)) < 0) {
+ if (Recover(ret) < 0) {
+ eof = true;
+ error.Format(alsa_input_domain,
+ "PCM error - stream aborted");
+ return 0;
+ }
+ }
+
+ size_t nbytes = ret * frame_size;
+ offset += nbytes;
+ return nbytes;
+}
+
+int
+AlsaInputStream::PrepareSockets()
+{
+ if (!waiting) {
+ ClearSocketList();
+ return -1;
+ }
+
+ int count = snd_pcm_poll_descriptors_count(capture_handle);
+ if (count < 0) {
+ ClearSocketList();
+ return -1;
+ }
+
+ struct pollfd *pfds = pfd_buffer.Get(count);
+
+ count = snd_pcm_poll_descriptors(capture_handle, pfds, count);
+ if (count < 0)
+ count = 0;
+
+ ReplaceSocketList(pfds, count);
+ return -1;
+}
+
+void
+AlsaInputStream::DispatchSockets()
+{
+ waiting = false;
+
+ const ScopeLock protect(mutex);
+ /* wake up the thread that is waiting for more data */
+ cond.broadcast();
+}
+
+inline int
+AlsaInputStream::Recover(int err)
+{
+ switch(err) {
+ case -EPIPE:
+ LogDebug(alsa_input_domain, "Buffer Overrun");
+ // drop through
+ case -ESTRPIPE:
+ case -EINTR:
+ err = snd_pcm_recover(capture_handle, err, 1);
+ break;
+ default:
+ // something broken somewhere, give up
+ err = -1;
+ }
+ return err;
+}
+
+inline snd_pcm_t *
+AlsaInputStream::OpenDevice(const char *device,
+ int rate, snd_pcm_format_t format, int channels,
+ Error &error)
+{
+ snd_pcm_t *capture_handle;
+ int err;
+ if ((err = snd_pcm_open(&capture_handle, device,
+ SND_PCM_STREAM_CAPTURE, 0)) < 0) {
+ error.Format(alsa_input_domain, "Failed to open device: %s (%s)", device, snd_strerror(err));
+ return nullptr;
+ }
+
+ snd_pcm_hw_params_t *hw_params;
+ if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
+ error.Format(alsa_input_domain, "Cannot allocate hardware parameter structure (%s)", snd_strerror(err));
+ snd_pcm_close(capture_handle);
+ return nullptr;
+ }
+
+ if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {
+ error.Format(alsa_input_domain, "Cannot initialize hardware parameter structure (%s)", snd_strerror(err));
+ snd_pcm_hw_params_free(hw_params);
+ snd_pcm_close(capture_handle);
+ return nullptr;
+ }
+
+ if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
+ error.Format(alsa_input_domain, "Cannot set access type (%s)", snd_strerror (err));
+ snd_pcm_hw_params_free(hw_params);
+ snd_pcm_close(capture_handle);
+ return nullptr;
+ }
+
+ if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params, format)) < 0) {
+ snd_pcm_hw_params_free(hw_params);
+ snd_pcm_close(capture_handle);
+ error.Format(alsa_input_domain, "Cannot set sample format (%s)", snd_strerror (err));
+ return nullptr;
+ }
+
+ if ((err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, channels)) < 0) {
+ snd_pcm_hw_params_free(hw_params);
+ snd_pcm_close(capture_handle);
+ error.Format(alsa_input_domain, "Cannot set channels (%s)", snd_strerror (err));
+ return nullptr;
+ }
+
+ if ((err = snd_pcm_hw_params_set_rate(capture_handle, hw_params, rate, 0)) < 0) {
+ snd_pcm_hw_params_free(hw_params);
+ snd_pcm_close(capture_handle);
+ error.Format(alsa_input_domain, "Cannot set sample rate (%s)", snd_strerror (err));
+ return nullptr;
+ }
+
+ /* period needs to be big enough so that poll() doesn't fire too often,
+ * but small enough that buffer overruns don't occur if Read() is not
+ * invoked often enough.
+ * the calculation here is empirical; however all measurements were
+ * done using 44100:16:2. When we extend this plugin to support
+ * other audio formats then this may need to be revisited */
+ snd_pcm_uframes_t period = read_buffer_size * 2;
+ int direction = -1;
+ if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params,
+ &period, &direction)) < 0) {
+ error.Format(alsa_input_domain, "Cannot set period size (%s)",
+ snd_strerror(err));
+ snd_pcm_hw_params_free(hw_params);
+ snd_pcm_close(capture_handle);
+ return nullptr;
+ }
+
+ if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {
+ error.Format(alsa_input_domain, "Cannot set parameters (%s)",
+ snd_strerror(err));
+ snd_pcm_hw_params_free(hw_params);
+ snd_pcm_close(capture_handle);
+ return nullptr;
+ }
+
+ snd_pcm_hw_params_free (hw_params);
+
+ snd_pcm_sw_params_t *sw_params;
+
+ snd_pcm_sw_params_malloc(&sw_params);
+ snd_pcm_sw_params_current(capture_handle, sw_params);
+
+ if ((err = snd_pcm_sw_params_set_start_threshold(capture_handle, sw_params,
+ period)) < 0) {
+ error.Format(alsa_input_domain,
+ "unable to set start threshold (%s)", snd_strerror(err));
+ snd_pcm_sw_params_free(sw_params);
+ snd_pcm_close(capture_handle);
+ return nullptr;
+ }
+
+ if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0) {
+ error.Format(alsa_input_domain,
+ "unable to install sw params (%s)", snd_strerror(err));
+ snd_pcm_sw_params_free(sw_params);
+ snd_pcm_close(capture_handle);
+ return nullptr;
+ }
+
+ snd_pcm_sw_params_free(sw_params);
+
+ snd_pcm_prepare(capture_handle);
+
+ return capture_handle;
+}
+
+/*######################### Plugin Functions ##############################*/
+
+static InputStream *
+alsa_input_open(const char *uri, Mutex &mutex, Cond &cond, Error &error)
+{
+ return AlsaInputStream::Create(uri, mutex, cond, error);
+}
+
+const struct InputPlugin input_plugin_alsa = {
+ "alsa",
+ nullptr,
+ nullptr,
+ alsa_input_open,
+};