diff options
Diffstat (limited to '')
-rw-r--r-- | src/output/plugins/AlsaOutputPlugin.cxx | 275 |
1 files changed, 137 insertions, 138 deletions
diff --git a/src/output/plugins/AlsaOutputPlugin.cxx b/src/output/plugins/AlsaOutputPlugin.cxx index 28c374a00..22e13d7dc 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" @@ -131,17 +132,48 @@ struct AlsaOutput { mode(0), writei(snd_pcm_writei) { } + ~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 config_param ¶m, Error &error); + static AlsaOutput *Create(const config_param ¶m, 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 ¶m, Error &error) { @@ -178,8 +210,8 @@ AlsaOutput::Configure(const config_param ¶m, Error &error) return true; } -static AudioOutput * -alsa_init(const config_param ¶m, Error &error) +inline AlsaOutput * +AlsaOutput::Create(const config_param ¶m, Error &error) { AlsaOutput *ad = new AlsaOutput(); @@ -188,35 +220,20 @@ alsa_init(const config_param ¶m, Error &error) 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, |