diff options
Diffstat (limited to 'src/output/AlsaOutputPlugin.cxx')
| -rw-r--r-- | src/output/AlsaOutputPlugin.cxx | 869 |
1 files changed, 0 insertions, 869 deletions
diff --git a/src/output/AlsaOutputPlugin.cxx b/src/output/AlsaOutputPlugin.cxx deleted file mode 100644 index f8aae13a1..000000000 --- a/src/output/AlsaOutputPlugin.cxx +++ /dev/null @@ -1,869 +0,0 @@ -/* - * Copyright (C) 2003-2013 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 "MixerList.hxx" -#include "pcm/PcmExport.hxx" -#include "util/Manual.hxx" -#include "util/Error.hxx" -#include "util/Domain.hxx" -#include "Log.hxx" - -#include <glib.h> -#include <alsa/asoundlib.h> - -#include <string> - -#define ALSA_PCM_NEW_HW_PARAMS_API -#define ALSA_PCM_NEW_SW_PARAMS_API - -static const char default_device[] = "default"; - -static constexpr unsigned MPD_ALSA_BUFFER_TIME_US = 500000; - -#define 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 { - struct audio_output 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 USB according to the dCS suggested - * standard? - * - * @see http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf - */ - bool dsd_usb; - - /** 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). - */ - void *silence; - - AlsaOutput():mode(0), writei(snd_pcm_writei) { - } - - bool Init(const config_param ¶m, Error &error) { - return ao_base_init(&base, &alsa_output_plugin, - param, error); - } - - void Deinit() { - ao_base_finish(&base); - } -}; - -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(); -} - -static void -alsa_configure(AlsaOutput *ad, const config_param ¶m) -{ - ad->device = param.GetBlockValue("device", ""); - - ad->use_mmap = param.GetBlockValue("use_mmap", false); - - ad->dsd_usb = param.GetBlockValue("dsd_usb", false); - - ad->buffer_time = param.GetBlockValue("buffer_time", - MPD_ALSA_BUFFER_TIME_US); - ad->period_time = param.GetBlockValue("period_time", 0u); - -#ifdef SND_PCM_NO_AUTO_RESAMPLE - if (!param.GetBlockValue("auto_resample", true)) - ad->mode |= SND_PCM_NO_AUTO_RESAMPLE; -#endif - -#ifdef SND_PCM_NO_AUTO_CHANNELS - if (!param.GetBlockValue("auto_channels", true)) - ad->mode |= SND_PCM_NO_AUTO_CHANNELS; -#endif - -#ifdef SND_PCM_NO_AUTO_FORMAT - if (!param.GetBlockValue("auto_format", true)) - ad->mode |= SND_PCM_NO_AUTO_FORMAT; -#endif -} - -static struct audio_output * -alsa_init(const config_param ¶m, Error &error) -{ - AlsaOutput *ad = new AlsaOutput(); - - if (!ad->Init(param, error)) { - delete ad; - return nullptr; - } - - alsa_configure(ad, param); - - return &ad->base; -} - -static void -alsa_finish(struct audio_output *ao) -{ - AlsaOutput *ad = (AlsaOutput *)ao; - - ad->Deinit(); - delete ad; - - /* free libasound's config cache */ - snd_config_update_free_global(); -} - -static bool -alsa_output_enable(struct audio_output *ao, gcc_unused Error &error) -{ - AlsaOutput *ad = (AlsaOutput *)ao; - - ad->pcm_export.Construct(); - return true; -} - -static void -alsa_output_disable(struct audio_output *ao) -{ - AlsaOutput *ad = (AlsaOutput *)ao; - - ad->pcm_export.Destruct(); -} - -static bool -alsa_test_default_device(void) -{ - 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; -} - -static snd_pcm_format_t -get_bitformat(SampleFormat sample_format) -{ - switch (sample_format) { - case SampleFormat::UNDEFINED: - case SampleFormat::DSD: - return SND_PCM_FORMAT_UNKNOWN; - - 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(); -} - -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; - } -} - -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; - } -} - -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 const 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; - int 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 = g_malloc(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->dsd_usb); - assert(audio_format.format == SampleFormat::DSD); - - /* pass 24 bit to alsa_setup() */ - - AudioFormat usb_format = audio_format; - usb_format.format = SampleFormat::S24_P32; - usb_format.sample_rate /= 2; - - const AudioFormat check = usb_format; - - if (!alsa_setup(ad, usb_format, packed_r, reverse_endian_r, error)) - return false; - - /* if the device allows only 32 bit, shift all DSD-over-USB - 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 = usb_format.format == SampleFormat::S32; - if (usb_format.format == SampleFormat::S32) - usb_format.format = SampleFormat::S24_P32; - - if (usb_format != check) { - /* no bit-perfect playback, which is required - for DSD over USB */ - error.Format(alsa_output_domain, - "Failed to configure DSD-over-USB on ALSA device \"%s\"", - alsa_device(ad)); - g_free(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 dsd_usb = ad->dsd_usb && - audio_format.format == SampleFormat::DSD; - const bool success = dsd_usb - ? 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, - dsd_usb, shift8, packed, reverse_endian); - return true; -} - -static bool -alsa_open(struct audio_output *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(struct audio_output *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(struct audio_output *ao) -{ - AlsaOutput *ad = (AlsaOutput *)ao; - - ad->period_position = 0; - ad->must_prepare = true; - - snd_pcm_drop(ad->pcm); -} - -static void -alsa_close(struct audio_output *ao) -{ - AlsaOutput *ad = (AlsaOutput *)ao; - - snd_pcm_close(ad->pcm); - g_free(ad->silence); -} - -static size_t -alsa_play(struct audio_output *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 size_t original_size = size; - chunk = ad->pcm_export->Export(chunk, size, size); - if (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 original_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 audio_output_plugin 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, -}; |