diff options
Diffstat (limited to 'src/output/plugins/AlsaOutputPlugin.cxx')
| -rw-r--r-- | src/output/plugins/AlsaOutputPlugin.cxx | 895 |
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..33a090274 --- /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 ¶m, 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 ¶m, 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 ¶m, 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_dsd(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_dsd(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_dsd(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_dsd(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, +}; |