aboutsummaryrefslogtreecommitdiffstats
path: root/src/output/plugins/AlsaOutputPlugin.cxx
diff options
context:
space:
mode:
Diffstat (limited to '')
-rw-r--r--src/output/plugins/AlsaOutputPlugin.cxx895
1 files changed, 895 insertions, 0 deletions
diff --git a/src/output/plugins/AlsaOutputPlugin.cxx b/src/output/plugins/AlsaOutputPlugin.cxx
new file mode 100644
index 000000000..28c374a00
--- /dev/null
+++ b/src/output/plugins/AlsaOutputPlugin.cxx
@@ -0,0 +1,895 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "AlsaOutputPlugin.hxx"
+#include "../OutputAPI.hxx"
+#include "mixer/MixerList.hxx"
+#include "pcm/PcmExport.hxx"
+#include "config/ConfigError.hxx"
+#include "util/Manual.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "util/ConstBuffer.hxx"
+#include "Log.hxx"
+
+#include <alsa/asoundlib.h>
+
+#include <string>
+
+#if SND_LIB_VERSION >= 0x1001c
+/* alsa-lib supports DSD since version 1.0.27.1 */
+#define HAVE_ALSA_DSD
+#endif
+
+static const char default_device[] = "default";
+
+static constexpr unsigned MPD_ALSA_BUFFER_TIME_US = 500000;
+
+static constexpr unsigned MPD_ALSA_RETRY_NR = 5;
+
+typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
+ snd_pcm_uframes_t size);
+
+struct AlsaOutput {
+ AudioOutput base;
+
+ Manual<PcmExport> pcm_export;
+
+ /**
+ * The configured name of the ALSA device; empty for the
+ * default device
+ */
+ std::string device;
+
+ /** use memory mapped I/O? */
+ bool use_mmap;
+
+ /**
+ * Enable DSD over PCM according to the DoP standard standard?
+ *
+ * @see http://dsd-guide.com/dop-open-standard
+ */
+ bool dop;
+
+ /** libasound's buffer_time setting (in microseconds) */
+ unsigned int buffer_time;
+
+ /** libasound's period_time setting (in microseconds) */
+ unsigned int period_time;
+
+ /** the mode flags passed to snd_pcm_open */
+ int mode;
+
+ /** the libasound PCM device handle */
+ snd_pcm_t *pcm;
+
+ /**
+ * a pointer to the libasound writei() function, which is
+ * snd_pcm_writei() or snd_pcm_mmap_writei(), depending on the
+ * use_mmap configuration
+ */
+ alsa_writei_t *writei;
+
+ /**
+ * The size of one audio frame passed to method play().
+ */
+ size_t in_frame_size;
+
+ /**
+ * The size of one audio frame passed to libasound.
+ */
+ size_t out_frame_size;
+
+ /**
+ * The size of one period, in number of frames.
+ */
+ snd_pcm_uframes_t period_frames;
+
+ /**
+ * The number of frames written in the current period.
+ */
+ snd_pcm_uframes_t period_position;
+
+ /**
+ * Do we need to call snd_pcm_prepare() before the next write?
+ * It means that we put the device to SND_PCM_STATE_SETUP by
+ * calling snd_pcm_drop().
+ *
+ * Without this flag, we could easily recover after a failed
+ * optimistic write (returning -EBADFD), but the Raspberry Pi
+ * audio driver is infamous for generating ugly artefacts from
+ * this.
+ */
+ bool must_prepare;
+
+ /**
+ * This buffer gets allocated after opening the ALSA device.
+ * It contains silence samples, enough to fill one period (see
+ * #period_frames).
+ */
+ uint8_t *silence;
+
+ AlsaOutput()
+ :base(alsa_output_plugin),
+ mode(0), writei(snd_pcm_writei) {
+ }
+
+ bool Configure(const config_param &param, Error &error);
+};
+
+static constexpr Domain alsa_output_domain("alsa_output");
+
+static const char *
+alsa_device(const AlsaOutput *ad)
+{
+ return ad->device.empty() ? default_device : ad->device.c_str();
+}
+
+inline bool
+AlsaOutput::Configure(const config_param &param, Error &error)
+{
+ if (!base.Configure(param, error))
+ return false;
+
+ device = param.GetBlockValue("device", "");
+
+ use_mmap = param.GetBlockValue("use_mmap", false);
+
+ dop = param.GetBlockValue("dop", false) ||
+ /* legacy name from MPD 0.18 and older: */
+ param.GetBlockValue("dsd_usb", false);
+
+ buffer_time = param.GetBlockValue("buffer_time",
+ MPD_ALSA_BUFFER_TIME_US);
+ period_time = param.GetBlockValue("period_time", 0u);
+
+#ifdef SND_PCM_NO_AUTO_RESAMPLE
+ if (!param.GetBlockValue("auto_resample", true))
+ mode |= SND_PCM_NO_AUTO_RESAMPLE;
+#endif
+
+#ifdef SND_PCM_NO_AUTO_CHANNELS
+ if (!param.GetBlockValue("auto_channels", true))
+ mode |= SND_PCM_NO_AUTO_CHANNELS;
+#endif
+
+#ifdef SND_PCM_NO_AUTO_FORMAT
+ if (!param.GetBlockValue("auto_format", true))
+ mode |= SND_PCM_NO_AUTO_FORMAT;
+#endif
+
+ return true;
+}
+
+static AudioOutput *
+alsa_init(const config_param &param, Error &error)
+{
+ AlsaOutput *ad = new AlsaOutput();
+
+ if (!ad->Configure(param, error)) {
+ delete ad;
+ return nullptr;
+ }
+
+ return &ad->base;
+}
+
+static void
+alsa_finish(AudioOutput *ao)
+{
+ AlsaOutput *ad = (AlsaOutput *)ao;
+
+ delete ad;
+
+ /* free libasound's config cache */
+ snd_config_update_free_global();
+}
+
+static bool
+alsa_output_enable(AudioOutput *ao, gcc_unused Error &error)
+{
+ AlsaOutput *ad = (AlsaOutput *)ao;
+
+ ad->pcm_export.Construct();
+ return true;
+}
+
+static void
+alsa_output_disable(AudioOutput *ao)
+{
+ AlsaOutput *ad = (AlsaOutput *)ao;
+
+ ad->pcm_export.Destruct();
+}
+
+static bool
+alsa_test_default_device()
+{
+ snd_pcm_t *handle;
+
+ int ret = snd_pcm_open(&handle, default_device,
+ SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+ if (ret) {
+ FormatError(alsa_output_domain,
+ "Error opening default ALSA device: %s",
+ snd_strerror(-ret));
+ return false;
+ } else
+ snd_pcm_close(handle);
+
+ return true;
+}
+
+/**
+ * Convert MPD's #SampleFormat enum to libasound's snd_pcm_format_t
+ * enum. Returns SND_PCM_FORMAT_UNKNOWN if there is no according ALSA
+ * PCM format.
+ */
+static snd_pcm_format_t
+get_bitformat(SampleFormat sample_format)
+{
+ switch (sample_format) {
+ case SampleFormat::UNDEFINED:
+ return SND_PCM_FORMAT_UNKNOWN;
+
+ case SampleFormat::DSD:
+#ifdef HAVE_ALSA_DSD
+ return SND_PCM_FORMAT_DSD_U8;
+#else
+ return SND_PCM_FORMAT_UNKNOWN;
+#endif
+
+ case SampleFormat::S8:
+ return SND_PCM_FORMAT_S8;
+
+ case SampleFormat::S16:
+ return SND_PCM_FORMAT_S16;
+
+ case SampleFormat::S24_P32:
+ return SND_PCM_FORMAT_S24;
+
+ case SampleFormat::S32:
+ return SND_PCM_FORMAT_S32;
+
+ case SampleFormat::FLOAT:
+ return SND_PCM_FORMAT_FLOAT;
+ }
+
+ assert(false);
+ gcc_unreachable();
+}
+
+/**
+ * Determine the byte-swapped PCM format. Returns
+ * SND_PCM_FORMAT_UNKNOWN if the format cannot be byte-swapped.
+ */
+static snd_pcm_format_t
+byteswap_bitformat(snd_pcm_format_t fmt)
+{
+ switch (fmt) {
+ case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
+ case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
+ case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
+ case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
+ case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;
+
+ case SND_PCM_FORMAT_S24_3BE:
+ return SND_PCM_FORMAT_S24_3LE;
+
+ case SND_PCM_FORMAT_S24_3LE:
+ return SND_PCM_FORMAT_S24_3BE;
+
+ case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
+ default: return SND_PCM_FORMAT_UNKNOWN;
+ }
+}
+
+/**
+ * Check if there is a "packed" version of the give PCM format.
+ * Returns SND_PCM_FORMAT_UNKNOWN if not.
+ */
+static snd_pcm_format_t
+alsa_to_packed_format(snd_pcm_format_t fmt)
+{
+ switch (fmt) {
+ case SND_PCM_FORMAT_S24_LE:
+ return SND_PCM_FORMAT_S24_3LE;
+
+ case SND_PCM_FORMAT_S24_BE:
+ return SND_PCM_FORMAT_S24_3BE;
+
+ default:
+ return SND_PCM_FORMAT_UNKNOWN;
+ }
+}
+
+/**
+ * Attempts to configure the specified sample format. On failure,
+ * fall back to the packed version.
+ */
+static int
+alsa_try_format_or_packed(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+ snd_pcm_format_t fmt, bool *packed_r)
+{
+ int err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
+ if (err == 0)
+ *packed_r = false;
+
+ if (err != -EINVAL)
+ return err;
+
+ fmt = alsa_to_packed_format(fmt);
+ if (fmt == SND_PCM_FORMAT_UNKNOWN)
+ return -EINVAL;
+
+ err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
+ if (err == 0)
+ *packed_r = true;
+
+ return err;
+}
+
+/**
+ * Attempts to configure the specified sample format, and tries the
+ * reversed host byte order if was not supported.
+ */
+static int
+alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+ SampleFormat sample_format,
+ bool *packed_r, bool *reverse_endian_r)
+{
+ snd_pcm_format_t alsa_format = get_bitformat(sample_format);
+ if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
+ return -EINVAL;
+
+ int err = alsa_try_format_or_packed(pcm, hwparams, alsa_format,
+ packed_r);
+ if (err == 0)
+ *reverse_endian_r = false;
+
+ if (err != -EINVAL)
+ return err;
+
+ alsa_format = byteswap_bitformat(alsa_format);
+ if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
+ return -EINVAL;
+
+ err = alsa_try_format_or_packed(pcm, hwparams, alsa_format, packed_r);
+ if (err == 0)
+ *reverse_endian_r = true;
+
+ return err;
+}
+
+/**
+ * Configure a sample format, and probe other formats if that fails.
+ */
+static int
+alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+ AudioFormat &audio_format,
+ bool *packed_r, bool *reverse_endian_r)
+{
+ /* try the input format first */
+
+ int err = alsa_output_try_format(pcm, hwparams,
+ audio_format.format,
+ packed_r, reverse_endian_r);
+
+ /* if unsupported by the hardware, try other formats */
+
+ static constexpr SampleFormat probe_formats[] = {
+ SampleFormat::S24_P32,
+ SampleFormat::S32,
+ SampleFormat::S16,
+ SampleFormat::S8,
+ SampleFormat::UNDEFINED,
+ };
+
+ for (unsigned i = 0;
+ err == -EINVAL && probe_formats[i] != SampleFormat::UNDEFINED;
+ ++i) {
+ const SampleFormat mpd_format = probe_formats[i];
+ if (mpd_format == audio_format.format)
+ continue;
+
+ err = alsa_output_try_format(pcm, hwparams, mpd_format,
+ packed_r, reverse_endian_r);
+ if (err == 0)
+ audio_format.format = mpd_format;
+ }
+
+ return err;
+}
+
+/**
+ * Set up the snd_pcm_t object which was opened by the caller. Set up
+ * the configured settings and the audio format.
+ */
+static bool
+alsa_setup(AlsaOutput *ad, AudioFormat &audio_format,
+ bool *packed_r, bool *reverse_endian_r, Error &error)
+{
+ unsigned int sample_rate = audio_format.sample_rate;
+ unsigned int channels = audio_format.channels;
+ int err;
+ const char *cmd = nullptr;
+ unsigned retry = MPD_ALSA_RETRY_NR;
+ unsigned int period_time, period_time_ro;
+ unsigned int buffer_time;
+
+ period_time_ro = period_time = ad->period_time;
+configure_hw:
+ /* configure HW params */
+ snd_pcm_hw_params_t *hwparams;
+ snd_pcm_hw_params_alloca(&hwparams);
+ cmd = "snd_pcm_hw_params_any";
+ err = snd_pcm_hw_params_any(ad->pcm, hwparams);
+ if (err < 0)
+ goto error;
+
+ if (ad->use_mmap) {
+ err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
+ SND_PCM_ACCESS_MMAP_INTERLEAVED);
+ if (err < 0) {
+ FormatWarning(alsa_output_domain,
+ "Cannot set mmap'ed mode on ALSA device \"%s\": %s",
+ alsa_device(ad), snd_strerror(-err));
+ LogWarning(alsa_output_domain,
+ "Falling back to direct write mode");
+ ad->use_mmap = false;
+ } else
+ ad->writei = snd_pcm_mmap_writei;
+ }
+
+ if (!ad->use_mmap) {
+ cmd = "snd_pcm_hw_params_set_access";
+ err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (err < 0)
+ goto error;
+ ad->writei = snd_pcm_writei;
+ }
+
+ err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
+ packed_r, reverse_endian_r);
+ if (err < 0) {
+ error.Format(alsa_output_domain, err,
+ "ALSA device \"%s\" does not support format %s: %s",
+ alsa_device(ad),
+ sample_format_to_string(audio_format.format),
+ snd_strerror(-err));
+ return false;
+ }
+
+ snd_pcm_format_t format;
+ if (snd_pcm_hw_params_get_format(hwparams, &format) == 0)
+ FormatDebug(alsa_output_domain,
+ "format=%s (%s)", snd_pcm_format_name(format),
+ snd_pcm_format_description(format));
+
+ err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
+ &channels);
+ if (err < 0) {
+ error.Format(alsa_output_domain, err,
+ "ALSA device \"%s\" does not support %i channels: %s",
+ alsa_device(ad), (int)audio_format.channels,
+ snd_strerror(-err));
+ return false;
+ }
+ audio_format.channels = (int8_t)channels;
+
+ err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
+ &sample_rate, nullptr);
+ if (err < 0 || sample_rate == 0) {
+ error.Format(alsa_output_domain, err,
+ "ALSA device \"%s\" does not support %u Hz audio",
+ alsa_device(ad), audio_format.sample_rate);
+ return false;
+ }
+ audio_format.sample_rate = sample_rate;
+
+ snd_pcm_uframes_t buffer_size_min, buffer_size_max;
+ snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min);
+ snd_pcm_hw_params_get_buffer_size_max(hwparams, &buffer_size_max);
+ unsigned buffer_time_min, buffer_time_max;
+ snd_pcm_hw_params_get_buffer_time_min(hwparams, &buffer_time_min, 0);
+ snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time_max, 0);
+ FormatDebug(alsa_output_domain, "buffer: size=%u..%u time=%u..%u",
+ (unsigned)buffer_size_min, (unsigned)buffer_size_max,
+ buffer_time_min, buffer_time_max);
+
+ snd_pcm_uframes_t period_size_min, period_size_max;
+ snd_pcm_hw_params_get_period_size_min(hwparams, &period_size_min, 0);
+ snd_pcm_hw_params_get_period_size_max(hwparams, &period_size_max, 0);
+ unsigned period_time_min, period_time_max;
+ snd_pcm_hw_params_get_period_time_min(hwparams, &period_time_min, 0);
+ snd_pcm_hw_params_get_period_time_max(hwparams, &period_time_max, 0);
+ FormatDebug(alsa_output_domain, "period: size=%u..%u time=%u..%u",
+ (unsigned)period_size_min, (unsigned)period_size_max,
+ period_time_min, period_time_max);
+
+ if (ad->buffer_time > 0) {
+ buffer_time = ad->buffer_time;
+ cmd = "snd_pcm_hw_params_set_buffer_time_near";
+ err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm, hwparams,
+ &buffer_time, nullptr);
+ if (err < 0)
+ goto error;
+ } else {
+ err = snd_pcm_hw_params_get_buffer_time(hwparams, &buffer_time,
+ nullptr);
+ if (err < 0)
+ buffer_time = 0;
+ }
+
+ if (period_time_ro == 0 && buffer_time >= 10000) {
+ period_time_ro = period_time = buffer_time / 4;
+
+ FormatDebug(alsa_output_domain,
+ "default period_time = buffer_time/4 = %u/4 = %u",
+ buffer_time, period_time);
+ }
+
+ if (period_time_ro > 0) {
+ period_time = period_time_ro;
+ cmd = "snd_pcm_hw_params_set_period_time_near";
+ err = snd_pcm_hw_params_set_period_time_near(ad->pcm, hwparams,
+ &period_time, nullptr);
+ if (err < 0)
+ goto error;
+ }
+
+ cmd = "snd_pcm_hw_params";
+ err = snd_pcm_hw_params(ad->pcm, hwparams);
+ if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
+ period_time_ro = period_time_ro >> 1;
+ goto configure_hw;
+ } else if (err < 0)
+ goto error;
+ if (retry != MPD_ALSA_RETRY_NR)
+ FormatDebug(alsa_output_domain,
+ "ALSA period_time set to %d", period_time);
+
+ snd_pcm_uframes_t alsa_buffer_size;
+ cmd = "snd_pcm_hw_params_get_buffer_size";
+ err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
+ if (err < 0)
+ goto error;
+
+ snd_pcm_uframes_t alsa_period_size;
+ cmd = "snd_pcm_hw_params_get_period_size";
+ err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
+ nullptr);
+ if (err < 0)
+ goto error;
+
+ /* configure SW params */
+ snd_pcm_sw_params_t *swparams;
+ snd_pcm_sw_params_alloca(&swparams);
+
+ cmd = "snd_pcm_sw_params_current";
+ err = snd_pcm_sw_params_current(ad->pcm, swparams);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params_set_start_threshold";
+ err = snd_pcm_sw_params_set_start_threshold(ad->pcm, swparams,
+ alsa_buffer_size -
+ alsa_period_size);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params_set_avail_min";
+ err = snd_pcm_sw_params_set_avail_min(ad->pcm, swparams,
+ alsa_period_size);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params";
+ err = snd_pcm_sw_params(ad->pcm, swparams);
+ if (err < 0)
+ goto error;
+
+ FormatDebug(alsa_output_domain, "buffer_size=%u period_size=%u",
+ (unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
+
+ if (alsa_period_size == 0)
+ /* this works around a SIGFPE bug that occurred when
+ an ALSA driver indicated period_size==0; this
+ caused a division by zero in alsa_play(). By using
+ the fallback "1", we make sure that this won't
+ happen again. */
+ alsa_period_size = 1;
+
+ ad->period_frames = alsa_period_size;
+ ad->period_position = 0;
+
+ ad->silence = new uint8_t[snd_pcm_frames_to_bytes(ad->pcm,
+ alsa_period_size)];
+ snd_pcm_format_set_silence(format, ad->silence,
+ alsa_period_size * channels);
+
+ return true;
+
+error:
+ error.Format(alsa_output_domain, err,
+ "Error opening ALSA device \"%s\" (%s): %s",
+ alsa_device(ad), cmd, snd_strerror(-err));
+ return false;
+}
+
+static bool
+alsa_setup_dop(AlsaOutput *ad, const AudioFormat audio_format,
+ bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
+ Error &error)
+{
+ assert(ad->dop);
+ assert(audio_format.format == SampleFormat::DSD);
+
+ /* pass 24 bit to alsa_setup() */
+
+ AudioFormat dop_format = audio_format;
+ dop_format.format = SampleFormat::S24_P32;
+ dop_format.sample_rate /= 2;
+
+ const AudioFormat check = dop_format;
+
+ if (!alsa_setup(ad, dop_format, packed_r, reverse_endian_r, error))
+ return false;
+
+ /* if the device allows only 32 bit, shift all DoP
+ samples left by 8 bit and leave the lower 8 bit cleared;
+ the DSD-over-USB documentation does not specify whether
+ this is legal, but there is anecdotical evidence that this
+ is possible (and the only option for some devices) */
+ *shift8_r = dop_format.format == SampleFormat::S32;
+ if (dop_format.format == SampleFormat::S32)
+ dop_format.format = SampleFormat::S24_P32;
+
+ if (dop_format != check) {
+ /* no bit-perfect playback, which is required
+ for DSD over USB */
+ error.Format(alsa_output_domain,
+ "Failed to configure DSD-over-PCM on ALSA device \"%s\"",
+ alsa_device(ad));
+ delete[] ad->silence;
+ return false;
+ }
+
+ return true;
+}
+
+static bool
+alsa_setup_or_dop(AlsaOutput *ad, AudioFormat &audio_format,
+ Error &error)
+{
+ bool shift8 = false, packed, reverse_endian;
+
+ const bool dop = ad->dop &&
+ audio_format.format == SampleFormat::DSD;
+ const bool success = dop
+ ? alsa_setup_dop(ad, audio_format,
+ &shift8, &packed, &reverse_endian,
+ error)
+ : alsa_setup(ad, audio_format, &packed, &reverse_endian,
+ error);
+ if (!success)
+ return false;
+
+ ad->pcm_export->Open(audio_format.format,
+ audio_format.channels,
+ dop, shift8, packed, reverse_endian);
+ return true;
+}
+
+static bool
+alsa_open(AudioOutput *ao, AudioFormat &audio_format, Error &error)
+{
+ AlsaOutput *ad = (AlsaOutput *)ao;
+
+ int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
+ SND_PCM_STREAM_PLAYBACK, ad->mode);
+ if (err < 0) {
+ error.Format(alsa_output_domain, err,
+ "Failed to open ALSA device \"%s\": %s",
+ alsa_device(ad), snd_strerror(err));
+ return false;
+ }
+
+ FormatDebug(alsa_output_domain, "opened %s type=%s",
+ snd_pcm_name(ad->pcm),
+ snd_pcm_type_name(snd_pcm_type(ad->pcm)));
+
+ if (!alsa_setup_or_dop(ad, audio_format, error)) {
+ snd_pcm_close(ad->pcm);
+ return false;
+ }
+
+ ad->in_frame_size = audio_format.GetFrameSize();
+ ad->out_frame_size = ad->pcm_export->GetFrameSize(audio_format);
+
+ ad->must_prepare = false;
+
+ return true;
+}
+
+/**
+ * Write silence to the ALSA device.
+ */
+static void
+alsa_write_silence(AlsaOutput *ad, snd_pcm_uframes_t nframes)
+{
+ ad->writei(ad->pcm, ad->silence, nframes);
+}
+
+static int
+alsa_recover(AlsaOutput *ad, int err)
+{
+ if (err == -EPIPE) {
+ FormatDebug(alsa_output_domain,
+ "Underrun on ALSA device \"%s\"", alsa_device(ad));
+ } else if (err == -ESTRPIPE) {
+ FormatDebug(alsa_output_domain,
+ "ALSA device \"%s\" was suspended",
+ alsa_device(ad));
+ }
+
+ switch (snd_pcm_state(ad->pcm)) {
+ case SND_PCM_STATE_PAUSED:
+ err = snd_pcm_pause(ad->pcm, /* disable */ 0);
+ break;
+ case SND_PCM_STATE_SUSPENDED:
+ err = snd_pcm_resume(ad->pcm);
+ if (err == -EAGAIN)
+ return 0;
+ /* fall-through to snd_pcm_prepare: */
+ case SND_PCM_STATE_SETUP:
+ case SND_PCM_STATE_XRUN:
+ ad->period_position = 0;
+ err = snd_pcm_prepare(ad->pcm);
+ break;
+ case SND_PCM_STATE_DISCONNECTED:
+ break;
+ /* this is no error, so just keep running */
+ case SND_PCM_STATE_RUNNING:
+ err = 0;
+ break;
+ default:
+ /* unknown state, do nothing */
+ break;
+ }
+
+ return err;
+}
+
+static void
+alsa_drain(AudioOutput *ao)
+{
+ AlsaOutput *ad = (AlsaOutput *)ao;
+
+ if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
+ return;
+
+ if (ad->period_position > 0) {
+ /* generate some silence to finish the partial
+ period */
+ snd_pcm_uframes_t nframes =
+ ad->period_frames - ad->period_position;
+ alsa_write_silence(ad, nframes);
+ }
+
+ snd_pcm_drain(ad->pcm);
+
+ ad->period_position = 0;
+}
+
+static void
+alsa_cancel(AudioOutput *ao)
+{
+ AlsaOutput *ad = (AlsaOutput *)ao;
+
+ ad->period_position = 0;
+ ad->must_prepare = true;
+
+ snd_pcm_drop(ad->pcm);
+}
+
+static void
+alsa_close(AudioOutput *ao)
+{
+ AlsaOutput *ad = (AlsaOutput *)ao;
+
+ snd_pcm_close(ad->pcm);
+ delete[] ad->silence;
+}
+
+static size_t
+alsa_play(AudioOutput *ao, const void *chunk, size_t size,
+ Error &error)
+{
+ AlsaOutput *ad = (AlsaOutput *)ao;
+
+ assert(size > 0);
+ assert(size % ad->in_frame_size == 0);
+
+ if (ad->must_prepare) {
+ ad->must_prepare = false;
+
+ int err = snd_pcm_prepare(ad->pcm);
+ if (err < 0) {
+ error.Set(alsa_output_domain, err, snd_strerror(-err));
+ return 0;
+ }
+ }
+
+ const auto e = ad->pcm_export->Export({chunk, size});
+ if (e.size == 0)
+ /* the DoP (DSD over PCM) filter converts two frames
+ at a time and ignores the last odd frame; if there
+ was only one frame (e.g. the last frame in the
+ file), the result is empty; to avoid an endless
+ loop, bail out here, and pretend the one frame has
+ been played */
+ return size;
+
+ chunk = e.data;
+ size = e.size;
+
+ assert(size % ad->out_frame_size == 0);
+
+ size /= ad->out_frame_size;
+ assert(size > 0);
+
+ while (true) {
+ snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
+ if (ret > 0) {
+ ad->period_position = (ad->period_position + ret)
+ % ad->period_frames;
+
+ size_t bytes_written = ret * ad->out_frame_size;
+ return ad->pcm_export->CalcSourceSize(bytes_written);
+ }
+
+ if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
+ alsa_recover(ad, ret) < 0) {
+ error.Set(alsa_output_domain, ret, snd_strerror(-ret));
+ return 0;
+ }
+ }
+}
+
+const struct AudioOutputPlugin alsa_output_plugin = {
+ "alsa",
+ alsa_test_default_device,
+ alsa_init,
+ alsa_finish,
+ alsa_output_enable,
+ alsa_output_disable,
+ alsa_open,
+ alsa_close,
+ nullptr,
+ nullptr,
+ alsa_play,
+ alsa_drain,
+ alsa_cancel,
+ nullptr,
+
+ &alsa_mixer_plugin,
+};