aboutsummaryrefslogtreecommitdiffstats
path: root/src/input/plugins/AlsaInputPlugin.cxx
blob: da2d9a9266c08f334293f5155dce610bdccf1820 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
/*
 * 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 "util/Cast.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 : MultiSocketMonitor, DeferredMonitor {
	InputStream base;
	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)
		:MultiSocketMonitor(loop),
		 DeferredMonitor(loop),
		 base(input_plugin_alsa, uri, mutex, cond),
		 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 */
		base.mime = "audio/x-mpd-cdda-pcm";
		base.seekable = false;
		base.size = -1;
		base.ready = true;
		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);

#if GCC_CHECK_VERSION(4,6) || defined(__clang__)
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Winvalid-offsetof"
#endif

	static constexpr AlsaInputStream *Cast(InputStream *is) {
		return ContainerCast(is, AlsaInputStream, base);
	}

#if GCC_CHECK_VERSION(4,6) || defined(__clang__)
#pragma GCC diagnostic pop
#endif

	bool Available() {
		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);

	bool IsEOF() {
		return eof;
	}

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;
	AlsaInputStream *stream = new AlsaInputStream(io_thread_get(),
						      uri, mutex, cond,
						      handle, frame_size);
	return &stream->base;
}

inline size_t
AlsaInputStream::Read(void *ptr, size_t size, Error &error)
{
	assert(ptr != nullptr);

	int num_frames = 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;
	base.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(base.mutex);
	/* wake up the thread that is waiting for more data */
	base.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);
}

static void
alsa_input_close(InputStream *is)
{
	AlsaInputStream *ais = AlsaInputStream::Cast(is);
	delete ais;
}

static bool
alsa_input_available(InputStream *is)
{
	AlsaInputStream *ais = AlsaInputStream::Cast(is);
	return ais->Available();
}

static size_t
alsa_input_read(InputStream *is, void *ptr, size_t size, Error &error)
{
	AlsaInputStream *ais = AlsaInputStream::Cast(is);
	return ais->Read(ptr, size, error);
}

static bool
alsa_input_eof(gcc_unused InputStream *is)
{
	AlsaInputStream *ais = AlsaInputStream::Cast(is);
	return ais->IsEOF();
}

const struct InputPlugin input_plugin_alsa = {
	"alsa",
	nullptr,
	nullptr,
	alsa_input_open,
	alsa_input_close,
	nullptr,
	nullptr,
	nullptr,
	alsa_input_available,
	alsa_input_read,
	alsa_input_eof,
	nullptr,
};