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-rw-r--r--src/output/plugins/AlsaOutputPlugin.cxx301
1 files changed, 150 insertions, 151 deletions
diff --git a/src/output/plugins/AlsaOutputPlugin.cxx b/src/output/plugins/AlsaOutputPlugin.cxx
index 28c374a00..8a7bb9643 100644
--- a/src/output/plugins/AlsaOutputPlugin.cxx
+++ b/src/output/plugins/AlsaOutputPlugin.cxx
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@@ -20,6 +20,7 @@
#include "config.h"
#include "AlsaOutputPlugin.hxx"
#include "../OutputAPI.hxx"
+#include "../Wrapper.hxx"
#include "mixer/MixerList.hxx"
#include "pcm/PcmExport.hxx"
#include "config/ConfigError.hxx"
@@ -131,92 +132,108 @@ struct AlsaOutput {
mode(0), writei(snd_pcm_writei) {
}
- bool Configure(const config_param &param, Error &error);
+ ~AlsaOutput() {
+ /* free libasound's config cache */
+ snd_config_update_free_global();
+ }
+
+ gcc_pure
+ const char *GetDevice() {
+ return device.empty() ? default_device : device.c_str();
+ }
+
+ bool Configure(const ConfigBlock &block, Error &error);
+ static AlsaOutput *Create(const ConfigBlock &block, Error &error);
+
+ bool Enable(Error &error);
+ void Disable();
+
+ bool Open(AudioFormat &audio_format, Error &error);
+ void Close();
+
+ size_t Play(const void *chunk, size_t size, Error &error);
+ void Drain();
+ void Cancel();
+
+private:
+ bool SetupDop(AudioFormat audio_format,
+ bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
+ Error &error);
+ bool SetupOrDop(AudioFormat &audio_format, Error &error);
+
+ int Recover(int err);
+
+ /**
+ * Write silence to the ALSA device.
+ */
+ void WriteSilence(snd_pcm_uframes_t nframes) {
+ writei(pcm, silence, nframes);
+ }
+
};
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)
+AlsaOutput::Configure(const ConfigBlock &block, Error &error)
{
- if (!base.Configure(param, error))
+ if (!base.Configure(block, error))
return false;
- device = param.GetBlockValue("device", "");
+ device = block.GetBlockValue("device", "");
- use_mmap = param.GetBlockValue("use_mmap", false);
+ use_mmap = block.GetBlockValue("use_mmap", false);
- dop = param.GetBlockValue("dop", false) ||
+ dop = block.GetBlockValue("dop", false) ||
/* legacy name from MPD 0.18 and older: */
- param.GetBlockValue("dsd_usb", false);
+ block.GetBlockValue("dsd_usb", false);
- buffer_time = param.GetBlockValue("buffer_time",
+ buffer_time = block.GetBlockValue("buffer_time",
MPD_ALSA_BUFFER_TIME_US);
- period_time = param.GetBlockValue("period_time", 0u);
+ period_time = block.GetBlockValue("period_time", 0u);
#ifdef SND_PCM_NO_AUTO_RESAMPLE
- if (!param.GetBlockValue("auto_resample", true))
+ if (!block.GetBlockValue("auto_resample", true))
mode |= SND_PCM_NO_AUTO_RESAMPLE;
#endif
#ifdef SND_PCM_NO_AUTO_CHANNELS
- if (!param.GetBlockValue("auto_channels", true))
+ if (!block.GetBlockValue("auto_channels", true))
mode |= SND_PCM_NO_AUTO_CHANNELS;
#endif
#ifdef SND_PCM_NO_AUTO_FORMAT
- if (!param.GetBlockValue("auto_format", true))
+ if (!block.GetBlockValue("auto_format", true))
mode |= SND_PCM_NO_AUTO_FORMAT;
#endif
return true;
}
-static AudioOutput *
-alsa_init(const config_param &param, Error &error)
+inline AlsaOutput *
+AlsaOutput::Create(const ConfigBlock &block, Error &error)
{
AlsaOutput *ad = new AlsaOutput();
- if (!ad->Configure(param, error)) {
+ if (!ad->Configure(block, error)) {
delete ad;
return nullptr;
}
- return &ad->base;
+ return ad;
}
-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)
+inline bool
+AlsaOutput::Enable(gcc_unused Error &error)
{
- AlsaOutput *ad = (AlsaOutput *)ao;
-
- ad->pcm_export.Construct();
+ pcm_export.Construct();
return true;
}
-static void
-alsa_output_disable(AudioOutput *ao)
+inline void
+AlsaOutput::Disable()
{
- AlsaOutput *ad = (AlsaOutput *)ao;
-
- ad->pcm_export.Destruct();
+ pcm_export.Destruct();
}
static bool
@@ -450,7 +467,7 @@ configure_hw:
if (err < 0) {
FormatWarning(alsa_output_domain,
"Cannot set mmap'ed mode on ALSA device \"%s\": %s",
- alsa_device(ad), snd_strerror(-err));
+ ad->GetDevice(), snd_strerror(-err));
LogWarning(alsa_output_domain,
"Falling back to direct write mode");
ad->use_mmap = false;
@@ -472,7 +489,7 @@ configure_hw:
if (err < 0) {
error.Format(alsa_output_domain, err,
"ALSA device \"%s\" does not support format %s: %s",
- alsa_device(ad),
+ ad->GetDevice(),
sample_format_to_string(audio_format.format),
snd_strerror(-err));
return false;
@@ -489,7 +506,7 @@ configure_hw:
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,
+ ad->GetDevice(), (int)audio_format.channels,
snd_strerror(-err));
return false;
}
@@ -500,7 +517,7 @@ configure_hw:
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);
+ ad->GetDevice(), audio_format.sample_rate);
return false;
}
audio_format.sample_rate = sample_rate;
@@ -631,16 +648,16 @@ configure_hw:
error:
error.Format(alsa_output_domain, err,
"Error opening ALSA device \"%s\" (%s): %s",
- alsa_device(ad), cmd, snd_strerror(-err));
+ ad->GetDevice(), 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)
+inline bool
+AlsaOutput::SetupDop(const AudioFormat audio_format,
+ bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
+ Error &error)
{
- assert(ad->dop);
+ assert(dop);
assert(audio_format.format == SampleFormat::DSD);
/* pass 24 bit to alsa_setup() */
@@ -651,7 +668,7 @@ alsa_setup_dop(AlsaOutput *ad, const AudioFormat audio_format,
const AudioFormat check = dop_format;
- if (!alsa_setup(ad, dop_format, packed_r, reverse_endian_r, error))
+ if (!alsa_setup(this, dop_format, packed_r, reverse_endian_r, error))
return false;
/* if the device allows only 32 bit, shift all DoP
@@ -668,102 +685,91 @@ alsa_setup_dop(AlsaOutput *ad, const AudioFormat audio_format,
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;
+ GetDevice());
+ delete[] silence;
return false;
}
return true;
}
-static bool
-alsa_setup_or_dop(AlsaOutput *ad, AudioFormat &audio_format,
- Error &error)
+inline bool
+AlsaOutput::SetupOrDop(AudioFormat &audio_format, Error &error)
{
bool shift8 = false, packed, reverse_endian;
- const bool dop = ad->dop &&
+ const bool dop2 = 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,
+ const bool success = dop2
+ ? SetupDop(audio_format,
+ &shift8, &packed, &reverse_endian,
+ error)
+ : alsa_setup(this, 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);
+ pcm_export->Open(audio_format.format,
+ audio_format.channels,
+ dop2, shift8, packed, reverse_endian);
return true;
}
-static bool
-alsa_open(AudioOutput *ao, AudioFormat &audio_format, Error &error)
+inline bool
+AlsaOutput::Open(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);
+ int err = snd_pcm_open(&pcm, GetDevice(),
+ SND_PCM_STREAM_PLAYBACK, mode);
if (err < 0) {
error.Format(alsa_output_domain, err,
"Failed to open ALSA device \"%s\": %s",
- alsa_device(ad), snd_strerror(err));
+ GetDevice(), 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)));
+ snd_pcm_name(pcm),
+ snd_pcm_type_name(snd_pcm_type(pcm)));
- if (!alsa_setup_or_dop(ad, audio_format, error)) {
- snd_pcm_close(ad->pcm);
+ if (!SetupOrDop(audio_format, error)) {
+ snd_pcm_close(pcm);
return false;
}
- ad->in_frame_size = audio_format.GetFrameSize();
- ad->out_frame_size = ad->pcm_export->GetFrameSize(audio_format);
+ in_frame_size = audio_format.GetFrameSize();
+ out_frame_size = pcm_export->GetFrameSize(audio_format);
- ad->must_prepare = false;
+ 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)
+inline int
+AlsaOutput::Recover(int err)
{
if (err == -EPIPE) {
FormatDebug(alsa_output_domain,
- "Underrun on ALSA device \"%s\"", alsa_device(ad));
+ "Underrun on ALSA device \"%s\"",
+ GetDevice());
} else if (err == -ESTRPIPE) {
FormatDebug(alsa_output_domain,
"ALSA device \"%s\" was suspended",
- alsa_device(ad));
+ GetDevice());
}
- switch (snd_pcm_state(ad->pcm)) {
+ switch (snd_pcm_state(pcm)) {
case SND_PCM_STATE_PAUSED:
- err = snd_pcm_pause(ad->pcm, /* disable */ 0);
+ err = snd_pcm_pause(pcm, /* disable */ 0);
break;
case SND_PCM_STATE_SUSPENDED:
- err = snd_pcm_resume(ad->pcm);
+ err = snd_pcm_resume(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);
+ period_position = 0;
+ err = snd_pcm_prepare(pcm);
break;
case SND_PCM_STATE_DISCONNECTED:
break;
@@ -779,67 +785,58 @@ alsa_recover(AlsaOutput *ad, int err)
return err;
}
-static void
-alsa_drain(AudioOutput *ao)
+inline void
+AlsaOutput::Drain()
{
- AlsaOutput *ad = (AlsaOutput *)ao;
-
- if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
+ if (snd_pcm_state(pcm) != SND_PCM_STATE_RUNNING)
return;
- if (ad->period_position > 0) {
+ if (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);
+ period_frames - period_position;
+ WriteSilence(nframes);
}
- snd_pcm_drain(ad->pcm);
+ snd_pcm_drain(pcm);
- ad->period_position = 0;
+ period_position = 0;
}
-static void
-alsa_cancel(AudioOutput *ao)
+inline void
+AlsaOutput::Cancel()
{
- AlsaOutput *ad = (AlsaOutput *)ao;
+ period_position = 0;
+ must_prepare = true;
- ad->period_position = 0;
- ad->must_prepare = true;
-
- snd_pcm_drop(ad->pcm);
+ snd_pcm_drop(pcm);
}
-static void
-alsa_close(AudioOutput *ao)
+inline void
+AlsaOutput::Close()
{
- AlsaOutput *ad = (AlsaOutput *)ao;
-
- snd_pcm_close(ad->pcm);
- delete[] ad->silence;
+ snd_pcm_close(pcm);
+ delete[] silence;
}
-static size_t
-alsa_play(AudioOutput *ao, const void *chunk, size_t size,
- Error &error)
+inline size_t
+AlsaOutput::Play(const void *chunk, size_t size, Error &error)
{
- AlsaOutput *ad = (AlsaOutput *)ao;
-
assert(size > 0);
- assert(size % ad->in_frame_size == 0);
+ assert(size % in_frame_size == 0);
- if (ad->must_prepare) {
- ad->must_prepare = false;
+ if (must_prepare) {
+ must_prepare = false;
- int err = snd_pcm_prepare(ad->pcm);
+ int err = snd_pcm_prepare(pcm);
if (err < 0) {
error.Set(alsa_output_domain, err, snd_strerror(-err));
return 0;
}
}
- const auto e = ad->pcm_export->Export({chunk, size});
+ const auto e = 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
@@ -852,43 +849,45 @@ alsa_play(AudioOutput *ao, const void *chunk, size_t size,
chunk = e.data;
size = e.size;
- assert(size % ad->out_frame_size == 0);
+ assert(size % out_frame_size == 0);
- size /= ad->out_frame_size;
+ size /= out_frame_size;
assert(size > 0);
while (true) {
- snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
+ snd_pcm_sframes_t ret = writei(pcm, chunk, size);
if (ret > 0) {
- ad->period_position = (ad->period_position + ret)
- % ad->period_frames;
+ period_position = (period_position + ret)
+ % period_frames;
- size_t bytes_written = ret * ad->out_frame_size;
- return ad->pcm_export->CalcSourceSize(bytes_written);
+ size_t bytes_written = ret * out_frame_size;
+ return pcm_export->CalcSourceSize(bytes_written);
}
if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
- alsa_recover(ad, ret) < 0) {
+ Recover(ret) < 0) {
error.Set(alsa_output_domain, ret, snd_strerror(-ret));
return 0;
}
}
}
+typedef AudioOutputWrapper<AlsaOutput> Wrapper;
+
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,
+ &Wrapper::Init,
+ &Wrapper::Finish,
+ &Wrapper::Enable,
+ &Wrapper::Disable,
+ &Wrapper::Open,
+ &Wrapper::Close,
nullptr,
nullptr,
- alsa_play,
- alsa_drain,
- alsa_cancel,
+ &Wrapper::Play,
+ &Wrapper::Drain,
+ &Wrapper::Cancel,
nullptr,
&alsa_mixer_plugin,
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-03 22:40:18 +0000