output/alsa: move functions into the struct
This commit is contained in:
Max Kellermann
parent
f532964fde
commit
153f5854e2
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@ -20,6 +20,7 @@
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#include "config.h"
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#include "AlsaOutputPlugin.hxx"
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#include "../OutputAPI.hxx"
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#include "../Wrapper.hxx"
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#include "mixer/MixerList.hxx"
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#include "pcm/PcmExport.hxx"
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#include "config/ConfigError.hxx"
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@ -131,17 +132,48 @@ struct AlsaOutput {
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mode(0), writei(snd_pcm_writei) {
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}
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~AlsaOutput() {
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/* free libasound's config cache */
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snd_config_update_free_global();
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}
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gcc_pure
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const char *GetDevice() {
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return device.empty() ? default_device : device.c_str();
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}
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bool Configure(const config_param ¶m, Error &error);
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static AlsaOutput *Create(const config_param ¶m, Error &error);
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bool Enable(Error &error);
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void Disable();
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bool Open(AudioFormat &audio_format, Error &error);
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void Close();
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size_t Play(const void *chunk, size_t size, Error &error);
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void Drain();
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void Cancel();
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private:
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bool SetupDop(AudioFormat audio_format,
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bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
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Error &error);
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bool SetupOrDop(AudioFormat &audio_format, Error &error);
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int Recover(int err);
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/**
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* Write silence to the ALSA device.
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*/
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void WriteSilence(snd_pcm_uframes_t nframes) {
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writei(pcm, silence, nframes);
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}
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};
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static constexpr Domain alsa_output_domain("alsa_output");
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static const char *
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alsa_device(const AlsaOutput *ad)
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{
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return ad->device.empty() ? default_device : ad->device.c_str();
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}
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inline bool
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AlsaOutput::Configure(const config_param ¶m, Error &error)
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{
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@ -178,8 +210,8 @@ AlsaOutput::Configure(const config_param ¶m, Error &error)
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return true;
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}
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static AudioOutput *
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alsa_init(const config_param ¶m, Error &error)
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inline AlsaOutput *
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AlsaOutput::Create(const config_param ¶m, Error &error)
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{
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AlsaOutput *ad = new AlsaOutput();
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@ -188,35 +220,20 @@ alsa_init(const config_param ¶m, Error &error)
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return nullptr;
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}
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return &ad->base;
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return ad;
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}
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static void
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alsa_finish(AudioOutput *ao)
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inline bool
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AlsaOutput::Enable(gcc_unused Error &error)
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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delete ad;
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/* free libasound's config cache */
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snd_config_update_free_global();
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}
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static bool
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alsa_output_enable(AudioOutput *ao, gcc_unused Error &error)
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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ad->pcm_export.Construct();
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pcm_export.Construct();
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return true;
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}
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static void
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alsa_output_disable(AudioOutput *ao)
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inline void
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AlsaOutput::Disable()
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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ad->pcm_export.Destruct();
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pcm_export.Destruct();
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}
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static bool
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@ -450,7 +467,7 @@ configure_hw:
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if (err < 0) {
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FormatWarning(alsa_output_domain,
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"Cannot set mmap'ed mode on ALSA device \"%s\": %s",
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alsa_device(ad), snd_strerror(-err));
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ad->GetDevice(), snd_strerror(-err));
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LogWarning(alsa_output_domain,
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"Falling back to direct write mode");
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ad->use_mmap = false;
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@ -472,7 +489,7 @@ configure_hw:
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if (err < 0) {
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error.Format(alsa_output_domain, err,
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"ALSA device \"%s\" does not support format %s: %s",
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alsa_device(ad),
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ad->GetDevice(),
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sample_format_to_string(audio_format.format),
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snd_strerror(-err));
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return false;
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@ -489,7 +506,7 @@ configure_hw:
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if (err < 0) {
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error.Format(alsa_output_domain, err,
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"ALSA device \"%s\" does not support %i channels: %s",
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alsa_device(ad), (int)audio_format.channels,
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ad->GetDevice(), (int)audio_format.channels,
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snd_strerror(-err));
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return false;
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}
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@ -500,7 +517,7 @@ configure_hw:
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if (err < 0 || sample_rate == 0) {
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error.Format(alsa_output_domain, err,
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"ALSA device \"%s\" does not support %u Hz audio",
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alsa_device(ad), audio_format.sample_rate);
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ad->GetDevice(), audio_format.sample_rate);
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return false;
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}
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audio_format.sample_rate = sample_rate;
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@ -631,16 +648,16 @@ configure_hw:
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error:
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error.Format(alsa_output_domain, err,
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"Error opening ALSA device \"%s\" (%s): %s",
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alsa_device(ad), cmd, snd_strerror(-err));
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ad->GetDevice(), cmd, snd_strerror(-err));
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return false;
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}
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static bool
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alsa_setup_dop(AlsaOutput *ad, const AudioFormat audio_format,
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bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
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Error &error)
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inline bool
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AlsaOutput::SetupDop(const AudioFormat audio_format,
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bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
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Error &error)
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{
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assert(ad->dop);
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assert(dop);
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assert(audio_format.format == SampleFormat::DSD);
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/* pass 24 bit to alsa_setup() */
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@ -651,7 +668,7 @@ alsa_setup_dop(AlsaOutput *ad, const AudioFormat audio_format,
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const AudioFormat check = dop_format;
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if (!alsa_setup(ad, dop_format, packed_r, reverse_endian_r, error))
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if (!alsa_setup(this, dop_format, packed_r, reverse_endian_r, error))
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return false;
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/* if the device allows only 32 bit, shift all DoP
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@ -668,102 +685,91 @@ alsa_setup_dop(AlsaOutput *ad, const AudioFormat audio_format,
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for DSD over USB */
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error.Format(alsa_output_domain,
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"Failed to configure DSD-over-PCM on ALSA device \"%s\"",
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alsa_device(ad));
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delete[] ad->silence;
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GetDevice());
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delete[] silence;
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return false;
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}
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return true;
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}
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static bool
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alsa_setup_or_dop(AlsaOutput *ad, AudioFormat &audio_format,
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Error &error)
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inline bool
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AlsaOutput::SetupOrDop(AudioFormat &audio_format, Error &error)
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{
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bool shift8 = false, packed, reverse_endian;
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const bool dop = ad->dop &&
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const bool dop2 = dop &&
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audio_format.format == SampleFormat::DSD;
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const bool success = dop
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? alsa_setup_dop(ad, audio_format,
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&shift8, &packed, &reverse_endian,
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error)
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: alsa_setup(ad, audio_format, &packed, &reverse_endian,
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const bool success = dop2
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? SetupDop(audio_format,
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&shift8, &packed, &reverse_endian,
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error)
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: alsa_setup(this, audio_format, &packed, &reverse_endian,
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error);
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if (!success)
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return false;
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ad->pcm_export->Open(audio_format.format,
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audio_format.channels,
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dop, shift8, packed, reverse_endian);
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pcm_export->Open(audio_format.format,
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audio_format.channels,
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dop2, shift8, packed, reverse_endian);
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return true;
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}
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static bool
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alsa_open(AudioOutput *ao, AudioFormat &audio_format, Error &error)
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inline bool
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AlsaOutput::Open(AudioFormat &audio_format, Error &error)
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
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SND_PCM_STREAM_PLAYBACK, ad->mode);
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int err = snd_pcm_open(&pcm, GetDevice(),
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SND_PCM_STREAM_PLAYBACK, mode);
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if (err < 0) {
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error.Format(alsa_output_domain, err,
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"Failed to open ALSA device \"%s\": %s",
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alsa_device(ad), snd_strerror(err));
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GetDevice(), snd_strerror(err));
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return false;
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}
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FormatDebug(alsa_output_domain, "opened %s type=%s",
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snd_pcm_name(ad->pcm),
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snd_pcm_type_name(snd_pcm_type(ad->pcm)));
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snd_pcm_name(pcm),
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snd_pcm_type_name(snd_pcm_type(pcm)));
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if (!alsa_setup_or_dop(ad, audio_format, error)) {
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snd_pcm_close(ad->pcm);
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if (!SetupOrDop(audio_format, error)) {
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snd_pcm_close(pcm);
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return false;
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}
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ad->in_frame_size = audio_format.GetFrameSize();
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ad->out_frame_size = ad->pcm_export->GetFrameSize(audio_format);
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in_frame_size = audio_format.GetFrameSize();
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out_frame_size = pcm_export->GetFrameSize(audio_format);
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ad->must_prepare = false;
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must_prepare = false;
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return true;
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}
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/**
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* Write silence to the ALSA device.
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*/
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static void
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alsa_write_silence(AlsaOutput *ad, snd_pcm_uframes_t nframes)
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{
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ad->writei(ad->pcm, ad->silence, nframes);
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}
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static int
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alsa_recover(AlsaOutput *ad, int err)
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inline int
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AlsaOutput::Recover(int err)
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{
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if (err == -EPIPE) {
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FormatDebug(alsa_output_domain,
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"Underrun on ALSA device \"%s\"", alsa_device(ad));
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"Underrun on ALSA device \"%s\"",
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GetDevice());
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} else if (err == -ESTRPIPE) {
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FormatDebug(alsa_output_domain,
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"ALSA device \"%s\" was suspended",
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alsa_device(ad));
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GetDevice());
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}
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switch (snd_pcm_state(ad->pcm)) {
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switch (snd_pcm_state(pcm)) {
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case SND_PCM_STATE_PAUSED:
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err = snd_pcm_pause(ad->pcm, /* disable */ 0);
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err = snd_pcm_pause(pcm, /* disable */ 0);
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break;
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case SND_PCM_STATE_SUSPENDED:
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err = snd_pcm_resume(ad->pcm);
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err = snd_pcm_resume(pcm);
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if (err == -EAGAIN)
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return 0;
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/* fall-through to snd_pcm_prepare: */
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case SND_PCM_STATE_SETUP:
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case SND_PCM_STATE_XRUN:
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ad->period_position = 0;
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err = snd_pcm_prepare(ad->pcm);
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period_position = 0;
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err = snd_pcm_prepare(pcm);
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break;
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case SND_PCM_STATE_DISCONNECTED:
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break;
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@ -779,67 +785,58 @@ alsa_recover(AlsaOutput *ad, int err)
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return err;
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}
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static void
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alsa_drain(AudioOutput *ao)
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inline void
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AlsaOutput::Drain()
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
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if (snd_pcm_state(pcm) != SND_PCM_STATE_RUNNING)
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return;
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if (ad->period_position > 0) {
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if (period_position > 0) {
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/* generate some silence to finish the partial
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period */
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snd_pcm_uframes_t nframes =
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ad->period_frames - ad->period_position;
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alsa_write_silence(ad, nframes);
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period_frames - period_position;
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WriteSilence(nframes);
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}
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snd_pcm_drain(ad->pcm);
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snd_pcm_drain(pcm);
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ad->period_position = 0;
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period_position = 0;
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}
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static void
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alsa_cancel(AudioOutput *ao)
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inline void
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AlsaOutput::Cancel()
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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period_position = 0;
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must_prepare = true;
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ad->period_position = 0;
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ad->must_prepare = true;
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snd_pcm_drop(ad->pcm);
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snd_pcm_drop(pcm);
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}
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static void
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alsa_close(AudioOutput *ao)
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inline void
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AlsaOutput::Close()
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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snd_pcm_close(ad->pcm);
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delete[] ad->silence;
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snd_pcm_close(pcm);
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delete[] silence;
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}
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static size_t
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alsa_play(AudioOutput *ao, const void *chunk, size_t size,
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Error &error)
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inline size_t
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AlsaOutput::Play(const void *chunk, size_t size, Error &error)
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{
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AlsaOutput *ad = (AlsaOutput *)ao;
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assert(size > 0);
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assert(size % ad->in_frame_size == 0);
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assert(size % in_frame_size == 0);
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if (ad->must_prepare) {
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ad->must_prepare = false;
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if (must_prepare) {
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must_prepare = false;
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int err = snd_pcm_prepare(ad->pcm);
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int err = snd_pcm_prepare(pcm);
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if (err < 0) {
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error.Set(alsa_output_domain, err, snd_strerror(-err));
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return 0;
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}
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}
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const auto e = ad->pcm_export->Export({chunk, size});
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const auto e = pcm_export->Export({chunk, size});
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if (e.size == 0)
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/* the DoP (DSD over PCM) filter converts two frames
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at a time and ignores the last odd frame; if there
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@ -852,43 +849,45 @@ alsa_play(AudioOutput *ao, const void *chunk, size_t size,
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chunk = e.data;
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size = e.size;
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assert(size % ad->out_frame_size == 0);
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assert(size % out_frame_size == 0);
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size /= ad->out_frame_size;
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size /= out_frame_size;
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assert(size > 0);
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while (true) {
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snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
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snd_pcm_sframes_t ret = writei(pcm, chunk, size);
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if (ret > 0) {
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ad->period_position = (ad->period_position + ret)
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% ad->period_frames;
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period_position = (period_position + ret)
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% period_frames;
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size_t bytes_written = ret * ad->out_frame_size;
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return ad->pcm_export->CalcSourceSize(bytes_written);
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size_t bytes_written = ret * out_frame_size;
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return pcm_export->CalcSourceSize(bytes_written);
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}
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if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
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alsa_recover(ad, ret) < 0) {
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Recover(ret) < 0) {
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error.Set(alsa_output_domain, ret, snd_strerror(-ret));
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return 0;
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}
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}
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}
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typedef AudioOutputWrapper<AlsaOutput> Wrapper;
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const struct AudioOutputPlugin alsa_output_plugin = {
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"alsa",
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alsa_test_default_device,
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alsa_init,
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alsa_finish,
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alsa_output_enable,
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alsa_output_disable,
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alsa_open,
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alsa_close,
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&Wrapper::Init,
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&Wrapper::Finish,
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&Wrapper::Enable,
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&Wrapper::Disable,
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&Wrapper::Open,
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&Wrapper::Close,
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nullptr,
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nullptr,
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alsa_play,
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alsa_drain,
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alsa_cancel,
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&Wrapper::Play,
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&Wrapper::Drain,
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&Wrapper::Cancel,
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nullptr,
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&alsa_mixer_plugin,
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