/*
* 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,
};