--- mythtv.orig/libs/libmyth/audiooutputalsa.cpp +++ mythtv/libs/libmyth/audiooutputalsa.cpp @@ -1,31 +1,53 @@ -#include -#include -#include -#include -#include "config.h" +/* + * Copyright (C) <=2008 unattributed author(s) + * Copyright (C) 2008 Alan Calvert + * + * 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. + */ using namespace std; #include "mythcontext.h" #include "audiooutputalsa.h" -#define LOC QString("ALSA: ") -#define LOC_WARN QString("ALSA, Warning: ") -#define LOC_ERR QString("ALSA, Error: ") - -// redefine assert as no-op to quiet some compiler warnings -// about assert always evaluating true in alsa headers. -#undef assert -#define assert(x) +#define LOC QString("AudioOutputALSA: ") +#define LOC_WARN QString("AudioOutputALSA Warn: ") +#define LOC_ERR QString("AudioOutputALSA Error: ") AudioOutputALSA::AudioOutputALSA(const AudioSettings &settings) : - AudioOutputBase(settings), - pcm_handle(NULL), - numbadioctls(0), - mixer_handle(NULL), - mixer_control(QString::null) + AudioOutputBase(settings) { - // Set everything up + pcm.device = (audio_passthru) + ? QByteArray(audio_passthru_device.toAscii()) + : QByteArray(audio_main_device.toAscii()); + if (pcm.device.isEmpty()) + pcm.device = "default"; + pcm.handle = NULL; + pcm.logtag = "pcm '" + pcm.device + "' "; + pcm.sample_rate = settings.samplerate; + pcm.use_mmap = false; + + mixer.device = gContext->GetSetting("MixerDevice", "default"); + if (mixer.device.startsWith("ALSA:")) + mixer.device.remove(0, 5); + mixer.handle = NULL; + mixer.control = gContext->GetSetting("MixerControl", "PCM"); + mixer.elem = NULL; + mixer.logtag = "mixer '" + mixer.device + "' "; + Reconfigure(settings); } @@ -36,672 +58,678 @@ bool AudioOutputALSA::OpenDevice() { - snd_pcm_format_t format; - unsigned int buffer_time, period_time; - int err; - - if (pcm_handle != NULL) - CloseDevice(); - - pcm_handle = NULL; - numbadioctls = 0; - - QString real_device = (audio_passthru) ? - audio_passthru_device : audio_main_device; - - VERBOSE(VB_GENERAL, QString("Opening ALSA audio device '%1'.") - .arg(real_device)); - - QByteArray dev_ba = real_device.toLocal8Bit(); - err = snd_pcm_open(&pcm_handle, dev_ba.constData(), - SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); - - if (err < 0) - { - Error(QString("snd_pcm_open(%1): %2") - .arg(real_device).arg(snd_strerror(err))); - - if (pcm_handle) - CloseDevice(); + if (!PrepPCM()) return false; - } - - /* the audio fragment size was computed by using the next lower power of 2 - of the following: - - const int video_frame_rate = 30; - const int bits_per_byte = 8; - int fbytes = (audio_bits * audio_channels * audio_samplerate) / - (bits_per_byte * video_frame_rate); - - For telephony apps, a much shorter fragment size is needed to reduce the - delay, and fragments should be multiples of the RTP packet size (10ms). - 20ms delay should be the max introduced by the driver, which equates - to 320 bytes at 8000 samples/sec and mono 16-bit samples - */ - if (source == AUDIOOUTPUT_TELEPHONY) - { - fragment_size = 320; - buffer_time = 80000; // 80 ms - period_time = buffer_time / 4; // 20ms - } - else + if (!PrepMixer()) { - fragment_size = 1536 * audio_channels * audio_bits / 8; - period_time = 25000; // in usec, interrupt period time - // in usec, for driver buffer alloc (64k max) - buffer_time = period_time * 16; - } - - if (audio_bits == 8) - format = SND_PCM_FORMAT_S8; - else if (audio_bits == 16) - // is the sound data coming in really little-endian or is it - // CPU-endian? -#ifdef WORDS_BIGENDIAN - format = SND_PCM_FORMAT_S16; -#else - format = SND_PCM_FORMAT_S16_LE; -#endif - else if (audio_bits == 24) -#ifdef WORDS_BIGENDIAN - format = SND_PCM_FORMAT_S24; -#else - format = SND_PCM_FORMAT_S24_LE; -#endif - else - { - Error(QString("Unknown sample format: %1 bits.").arg(audio_bits)); + CloseDevice(); return false; } - - err = SetParameters(pcm_handle, - format, audio_channels, audio_samplerate, buffer_time, - period_time); - if (err < 0) + if (AlsaBad(snd_pcm_prepare(pcm.handle), "prepare failed")) { - Error("Unable to set ALSA parameters"); CloseDevice(); return false; - } - - // make us think that soundcard buffer is 4 fragments smaller than - // it really is - audio_buffer_unused = soundcard_buffer_size - (fragment_size * 4); - - if (internal_vol) - OpenMixer(set_initial_vol); - - // Device opened successfully + } + pcm.bytes_per_frame = snd_pcm_frames_to_bytes(pcm.handle, 1); + fragment_size = pcm.bytes_per_frame * pcm.period_size; + soundcard_buffer_size = pcm.bytes_per_frame * pcm.buffer_size; + VERBOSE(VB_AUDIO, LOC + pcm.logtag + + QString("duly opened: fragment size %1, " + "soundcard buffer size %2 (in bytes)") + .arg(fragment_size).arg(soundcard_buffer_size)); return true; } void AudioOutputALSA::CloseDevice() { - CloseMixer(); - if (pcm_handle != NULL) - { - snd_pcm_close(pcm_handle); - pcm_handle = NULL; - } + if (mixer.handle != NULL + && !AlsaBad(snd_mixer_close(mixer.handle), "close mixer failed")) + VERBOSE(VB_AUDIO, LOC + mixer.logtag + "duly closed"); + mixer.handle = NULL; + if (pcm.handle != NULL + && !AlsaBad(snd_pcm_close(pcm.handle), "close pcm failed")) + VERBOSE(VB_AUDIO, LOC + pcm.logtag + "duly closed"); + pcm.handle = NULL; } - void AudioOutputALSA::WriteAudio(unsigned char *aubuf, int size) { - unsigned char *tmpbuf; - int lw = 0; - int frames = size / audio_bytes_per_sample; - - if (pcm_handle == NULL) - { - VERBOSE(VB_IMPORTANT, QString("WriteAudio() called with pcm_handle == NULL!")); + if (audio_actually_paused || pcm.handle == NULL) return; - } - - tmpbuf = aubuf; - - VERBOSE(VB_AUDIO+VB_TIMESTAMP, - QString("WriteAudio: Preparing %1 bytes (%2 frames)") - .arg(size).arg(frames)); - - while (frames > 0) + snd_pcm_state_t state = snd_pcm_state(pcm.handle); + switch (state) { - lw = pcm_write_func(pcm_handle, tmpbuf, frames); - - if (lw >= 0) - { - if (lw < frames) - VERBOSE(VB_AUDIO, QString("WriteAudio: short write %1 bytes (ok)") - .arg(lw * audio_bytes_per_sample)); - - frames -= lw; - tmpbuf += lw * audio_bytes_per_sample; // bytes - } - else if (lw == -EAGAIN) - { - VERBOSE(VB_AUDIO, QString("WriteAudio: device is blocked - waiting")); - - snd_pcm_wait(pcm_handle, 10); - } - else if (lw == -EPIPE && - snd_pcm_state(pcm_handle) == SND_PCM_STATE_XRUN) - { - VERBOSE(VB_IMPORTANT, "WriteAudio: buffer underrun"); - - if ((lw = snd_pcm_prepare(pcm_handle)) < 0) - { - Error(QString("WriteAudio: unable to recover from xrun: %1") - .arg(snd_strerror(lw))); - return; - } - } - else if (lw == -ESTRPIPE) - { - VERBOSE(VB_IMPORTANT, "WriteAudio: device is suspended"); - - while ((lw = snd_pcm_resume(pcm_handle)) == -EAGAIN) - usleep(200); - - if (lw < 0) - { - VERBOSE(VB_IMPORTANT, "WriteAudio: resume failed"); - - if ((lw = snd_pcm_prepare(pcm_handle)) < 0) - { - Error(QString("WriteAudio: unable to recover from suspend: %1") - .arg(snd_strerror(lw))); - return; - } - } - } - else if (lw == -EBADFD) - { - VERBOSE(VB_IMPORTANT, - QString("WriteAudio: device is in a bad state (state = %1)") - .arg(snd_pcm_state(pcm_handle))); - return; - } - else - { - VERBOSE(VB_IMPORTANT, QString("pcm_write_func: %1 (%2)") - .arg(snd_strerror(lw)).arg(lw)); - VERBOSE(VB_IMPORTANT, QString("WriteAudio: snd_pcm_state == %1") - .arg(snd_pcm_state(pcm_handle))); - - // CloseDevice(); - return; - } + case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: + if (!XrunRecovery()) + break; + case SND_PCM_STATE_PREPARED: + if (AlsaBad(snd_pcm_start(pcm.handle), "pcm start failed")) + break; + case SND_PCM_STATE_RUNNING: + if (pcm.use_mmap) + WriteMmap(aubuf, size / pcm.bytes_per_frame); + else + WriteRw(aubuf, size / pcm.bytes_per_frame); + break; + + default: + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + QString("alarming SND_PCM_STATE %1 through WriteAudio()") + .arg(state)); + break; } } int AudioOutputALSA::GetBufferedOnSoundcard(void) const -{ - if (pcm_handle == NULL) - { - VERBOSE(VB_IMPORTANT, QString("getBufferedOnSoundcard() called with pcm_handle == NULL!")); +{ + if (pcm.handle == NULL) return 0; - } - - // this should be more like what you want, previously this function - // was returning the soundcard buffer size -dag - - snd_pcm_sframes_t delay = 0; - - snd_pcm_state_t state = snd_pcm_state(pcm_handle); - if (state == SND_PCM_STATE_RUNNING || - state == SND_PCM_STATE_DRAINING) - { - snd_pcm_delay(pcm_handle, &delay); - } - - if (delay < 0) - delay = 0; - - int buffered = delay * audio_bytes_per_sample; + int buffered = 0; + snd_pcm_sframes_t frames = snd_pcm_avail_update(pcm.handle); + if (frames < 0) + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + QString("GetBufferedOnSoundcard query buffer status " + "failed: %1").arg(snd_strerror(frames))); + else + buffered = (pcm.buffer_size - frames) * pcm.bytes_per_frame; return buffered; } - int AudioOutputALSA::GetSpaceOnSoundcard(void) const { - if (pcm_handle == NULL) - { - VERBOSE(VB_IMPORTANT, QString("GetSpaceOnSoundcard() ") + - "called with pcm_handle == NULL!"); - + if (pcm.handle == NULL) return 0; - } - - snd_pcm_sframes_t avail, delay; - - snd_pcm_state_t state = snd_pcm_state(pcm_handle); - if (state == SND_PCM_STATE_RUNNING || - state == SND_PCM_STATE_DRAINING) - { - snd_pcm_delay(pcm_handle, &delay); - } - - avail = snd_pcm_avail_update(pcm_handle); - if (avail < 0 || - (snd_pcm_uframes_t)avail > (snd_pcm_uframes_t)soundcard_buffer_size) - avail = soundcard_buffer_size; - - int space = (avail * audio_bytes_per_sample) - audio_buffer_unused; - - if (space < 0) - space = 0; + int space = 0; + snd_pcm_sframes_t frames; + if ((frames = snd_pcm_avail_update(pcm.handle)) < 0) + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + QString("GetSpaceOnSoundcard query buffer status failed: %1") + .arg(snd_strerror(frames))); + else + space = frames * pcm.bytes_per_frame; return space; } - -int AudioOutputALSA::SetParameters(snd_pcm_t *handle, - snd_pcm_format_t format, unsigned int channels, - unsigned int rate, unsigned int buffer_time, - unsigned int period_time) -{ - int err, dir; - snd_pcm_hw_params_t *params; - snd_pcm_sw_params_t *swparams; - snd_pcm_uframes_t buffer_size; - snd_pcm_uframes_t period_size; - - VERBOSE(VB_AUDIO, QString("in SetParameters(format=%1, channels=%2, " - "rate=%3, buffer_time=%4, period_time=%5)") - .arg(format).arg(channels).arg(rate).arg(buffer_time).arg(period_time)); - - if (handle == NULL) - { - VERBOSE(VB_IMPORTANT, QString("SetParameters() called with handle == NULL!")); +int AudioOutputALSA::GetVolumeChannel(int channel) const +{ + if (mixer.elem == NULL) return 0; - } - - snd_pcm_hw_params_alloca(¶ms); - snd_pcm_sw_params_alloca(&swparams); - - /* choose all parameters */ - if ((err = snd_pcm_hw_params_any(handle, params)) < 0) - { - Error(QString("Broken configuration for playback; no configurations" - " available: %1").arg(snd_strerror(err))); - return err; - } - /* set the interleaved read/write format, use mmap if available */ - pcm_write_func = &snd_pcm_mmap_writei; - err = snd_pcm_hw_params_set_access( - handle, params, SND_PCM_ACCESS_MMAP_INTERLEAVED); - if (err < 0) + int retvol = 0; + if (channel > SND_MIXER_SCHN_UNKNOWN && channel < audio_channels) { - VERBOSE(VB_GENERAL, LOC_WARN + - "mmap not available, attempting to fall back to slow writes."); - QString old_err = snd_strerror(err); - pcm_write_func = &snd_pcm_writei; - err = snd_pcm_hw_params_set_access( - handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) - { - Error("Interleaved sound types MMAP & RW are not available"); - VERBOSE(VB_IMPORTANT, - QString("MMAP Error: %1\n\t\t\tRW Error: %2") - .arg(old_err).arg(snd_strerror(err))); - return err; + int chk; + snd_mixer_selem_channel_id_t chx = (snd_mixer_selem_channel_id_t)channel; + long mixvol; + chk = snd_mixer_selem_get_playback_volume(mixer.elem, chx, &mixvol); + if (chk < 0) + VERBOSE(VB_IMPORTANT, LOC_ERR + mixer.logtag + + QString("failed to get channel %2 volume: %3") + .arg(channel).arg(snd_strerror(chk))); + else if (mixer.volrange > 0L) + { + retvol = (mixvol - mixer.volmin) * 100.0f / mixer.volrange + .5f; + retvol = max(retvol, 0); + retvol = min(retvol, 100); + VERBOSE(VB_AUDIO+VB_EXTRA, LOC + mixer.logtag + + QString("get volume channel %1: %2 (mixer volume: %3") + .arg(channel).arg(retvol).arg(mixvol)); } } + else + VERBOSE(VB_IMPORTANT, LOC_ERR + mixer.logtag + + QString("get volume, invalid channel: %1").arg(channel)); + return retvol; +} - /* set the sample format */ - if ((err = snd_pcm_hw_params_set_format(handle, params, format)) < 0) - { - Error(QString("Sample format not available: %1") - .arg(snd_strerror(err))); - return err; - } +void AudioOutputALSA::SetVolumeChannel(int channel, int volume) +{ + if (!internal_vol || mixer.elem == NULL) + return; - /* set the count of channels */ - if ((err = snd_pcm_hw_params_set_channels(handle, params, channels)) < 0) + if (channel > SND_MIXER_SCHN_UNKNOWN && channel < audio_channels) { - Error(QString("Channels count (%1) not available: %2") - .arg(channels).arg(snd_strerror(err))); - return err; + snd_mixer_selem_channel_id_t chx = + (snd_mixer_selem_channel_id_t)channel; + long mixvol = volume * mixer.volrange / 100.0f - mixer.volmin + 0.5f; + mixvol = max(mixvol, mixer.volmin); + mixvol = min(mixvol, mixer.volmax); + if (!AlsaBad(snd_mixer_selem_set_playback_volume(mixer.elem, chx, mixvol), + QString("failed to set channel %1 volume -> %2") + .arg(channel).arg(volume))) + VERBOSE(VB_AUDIO, LOC + mixer.logtag + + QString("set channel %1 volume -> %2 (mixer volume: %3)") + .arg(channel).arg(volume).arg(mixvol)); } + else + VERBOSE(VB_IMPORTANT, LOC_ERR + mixer.logtag + + QString("set volume, invalid channel: %1").arg(channel)); +} - /* set the stream rate */ - unsigned int rrate = rate; - if ((err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0)) < 0) +void AudioOutputALSA::Reset(void) +{ + if (pcm.handle != NULL) { - Error(QString("Samplerate (%1Hz) not available: %2") - .arg(rate).arg(snd_strerror(err))); - return err; + AlsaBad(snd_pcm_drop(pcm.handle), "Reset drop failed"); + AlsaBad(snd_pcm_prepare(pcm.handle), "Reset prepare failed"); + AudioOutputBase::Reset(); + VERBOSE(VB_AUDIO+VB_EXTRA, LOC + pcm.logtag + "duly reset"); } +} - if (rrate != rate) - { - Error(QString("Rate doesn't match (requested %1Hz, got %2Hz)") - .arg(rate).arg(rrate)); - return -EINVAL; - } +void AudioOutputALSA::Pause(bool paused) +{ + if (pcm.handle == NULL) + return; - /* set the buffer time */ - if ((err = snd_pcm_hw_params_set_buffer_time_near(handle, params, - &buffer_time, &dir)) < 0) - { - Error(QString("Unable to set buffer time %1 for playback: %2") - .arg(buffer_time).arg(snd_strerror(err))); - return err; + if (paused && !audio_actually_paused) + { + AlsaBad(snd_pcm_drop(pcm.handle), "pause drop failed"); + pauseaudio = paused; + audio_actually_paused = true; + VERBOSE(VB_AUDIO+VB_EXTRA, LOC + pcm.logtag + "actually paused"); + } + else if (!paused && audio_actually_paused) + { + QString tag = "(un)Pause "; + AlsaBad(snd_pcm_drop(pcm.handle), tag + "unpause drop failed"); + AlsaBad(snd_pcm_prepare(pcm.handle), tag + "unpause prepare failed"); + pauseaudio = paused; + audio_actually_paused = false; + VERBOSE(VB_AUDIO+VB_EXTRA, LOC + pcm.logtag + "actually unpaused"); } +} - if ((err = snd_pcm_hw_params_get_buffer_size(params, &buffer_size)) < 0) - { - Error(QString("Unable to get buffer size for playback: %1") - .arg(snd_strerror(err))); - return err; - } else { - VERBOSE(VB_AUDIO, QString("get_buffer_size returned %1").arg(buffer_size)); +void AudioOutputALSA::Drain(void) +{ + AudioOutputBase::Drain(); + if (pcm.handle != NULL) + AlsaBad(snd_pcm_drain(pcm.handle), "pcm drain failed"); +} + +bool AudioOutputALSA::PrepPCM() +{ + if (pcm.handle != NULL) + snd_pcm_close(pcm.handle); + AlsaBad(snd_config_update_free_global(), "failed to update snd config"); + QRegExp built_in_regx("^(front|surround\\d{2}|hdmi|rear|center_lfe|iec958)" + ":(CARD=\\w+,DEV=\\d)|(DEV=\\d,CARD=\\w+)"); + bool is_built_in = QString(pcm.device).contains(built_in_regx); + QRegExp plug_regx("^(plug|plughw):\\w+"); + bool is_plug = QString(pcm.device).contains(plug_regx); + + while (true) + { + pcm.logtag = "pcm '" + pcm.device + "' "; + if (AlsaBad(snd_pcm_open(&pcm.handle, pcm.device.constData(), + SND_PCM_STREAM_PLAYBACK, + SND_PCM_NO_AUTO_CHANNELS), + "failed to open device")) + { + pcm.handle = NULL; + return false; + } + if (AlsaBad(snd_pcm_nonblock(pcm.handle, 1), "set nonblock failed")) + { + snd_pcm_close(pcm.handle); + pcm.handle = NULL; + return false; + } + if (PrepHwparams()) + break; + else if (!is_plug && !is_built_in) // try again with 'plug:' + { + AlsaBad(snd_pcm_close(pcm.handle), "plug retry, close failed"); + pcm.handle = NULL; + pcm.device.prepend("plug:"); + is_plug = true; + continue; + } + else + { + snd_pcm_close(pcm.handle); + pcm.handle = NULL; + return false; + } } - soundcard_buffer_size = buffer_size * audio_bytes_per_sample; - - /* set the period time */ - if ((err = snd_pcm_hw_params_set_period_time_near( - handle, params, &period_time, &dir)) < 0) + if (!PrepSwparams()) { - Error(QString("Unable to set period time %1 for playback: %2") - .arg(period_time).arg(snd_strerror(err))); - return err; - } else { - VERBOSE(VB_AUDIO, QString("set_period_time_near returned %1").arg(period_time)); - } - - if ((err = snd_pcm_hw_params_get_period_size(params, &period_size, - &dir)) < 0) { - Error(QString("Unable to get period size for playback: %1") - .arg(snd_strerror(err))); - return err; - } else { - VERBOSE(VB_AUDIO, QString("get_period_size returned %1").arg(period_size)); + snd_pcm_close(pcm.handle); + return false; } + return true; +} - /* write the parameters to device */ - if ((err = snd_pcm_hw_params(handle, params)) < 0) { - Error(QString("Unable to set hw params for playback: %1") - .arg(snd_strerror(err))); - return err; - } - - /* get the current swparams */ - if ((err = snd_pcm_sw_params_current(handle, swparams)) < 0) - { - Error(QString("Unable to determine current swparams for playback:" - " %1").arg(snd_strerror(err))); - return err; - } - /* start the transfer after period_size */ - if ((err = snd_pcm_sw_params_set_start_threshold(handle, swparams, - period_size)) < 0) - { - Error(QString("Unable to set start threshold mode for playback: %1") - .arg(snd_strerror(err))); - return err; - } +bool AudioOutputALSA::PrepHwparams(void) +{ + bool result = false; + snd_pcm_hw_params_t* hwparams; + snd_pcm_hw_params_alloca(&hwparams); + snd_pcm_access_t axs; + snd_pcm_format_t format; + unsigned int rate, latency, period_time, minchan, maxchan, periods; - /* allow the transfer when at least period_size samples can be processed */ - if ((err = snd_pcm_sw_params_set_avail_min(handle, swparams, - period_size)) < 0) + if (AlsaBad(snd_pcm_hw_params_any(pcm.handle, hwparams), + "no playback configurations available")) + goto fini; + if (AlsaBad(snd_pcm_hw_params_set_periods_integer(pcm.handle, hwparams), + "cannot restrict period size to integral value")) + goto fini; + pcm.use_mmap = false; + axs = SND_PCM_ACCESS_MMAP_INTERLEAVED; + if (!AlsaBad(snd_pcm_hw_params_set_access(pcm.handle, hwparams, axs), + "mmap io access not possible")) + pcm.use_mmap = true; + else { - Error(QString("Unable to set avail min for playback: %1") - .arg(snd_strerror(err))); - return err; + axs = SND_PCM_ACCESS_RW_INTERLEAVED; + if (AlsaBad(snd_pcm_hw_params_set_access(pcm.handle, hwparams, axs), + "failed to set mmap or rw access, both failed")) + goto fini; + } + switch (audio_bits) // ALSA to append _LE/_BE accordingly + { + case 8: + format = SND_PCM_FORMAT_U8; + break; + case 16: + default: + format = SND_PCM_FORMAT_S16; + break; + } + if (AlsaBad(snd_pcm_hw_params_set_format(pcm.handle, hwparams, format), + "failed to set sample format")) + goto fini; + + rate = pcm.sample_rate; + if (AlsaBad(snd_pcm_hw_params_set_rate_near(pcm.handle, hwparams, &rate, NULL), + QString("failed setting sample rate near %1").arg(rate))) + goto fini; + VERBOSE(VB_AUDIO+VB_EXTRA, LOC + pcm.logtag + + QString("requested sample rate %1, set sample rate %2") + .arg(pcm.sample_rate).arg(rate)); + if (rate < pcm.sample_rate * 0.95f || rate > pcm.sample_rate * 1.05f) + { + AlsaBad(-EINVAL, QString("requested sample rate %1, set sample rate %2 " + "- more than 5% off") + .arg(pcm.sample_rate).arg(rate)); + goto fini; + } + if (AlsaBad(snd_pcm_hw_params_set_channels(pcm.handle, hwparams, + audio_channels), + QString("failed to set channels to %1").arg(audio_channels))) + goto fini; + + // latency, period, buffer prep based on Alsa fn() snd_pcm_set_params() + latency = 50000; // 50msec + if (!AlsaBad(snd_pcm_hw_params_set_buffer_time_near(pcm.handle, hwparams, + &latency, NULL), + "initial buffer time/latency setting failed")) + { + if (AlsaBad(snd_pcm_hw_params_get_buffer_size(hwparams, &pcm.buffer_size), + "failed to get buffer size")) + goto fini; + if (AlsaBad(snd_pcm_hw_params_get_buffer_time(hwparams, &latency, NULL), + "failed to get latency setting")) + goto fini; + period_time = latency / 4; + if (AlsaBad(snd_pcm_hw_params_set_period_time_near(pcm.handle, hwparams, + &period_time, NULL), + "failed to set period time")) + goto fini; + if (AlsaBad(snd_pcm_hw_params_get_period_size(hwparams, &pcm.period_size, + NULL), + "failed to get period size")) + goto fini; } - - /* align all transfers to 1 sample */ - if ((err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1)) < 0) + else { - Error(QString("Unable to set transfer align for playback: %1") - .arg(snd_strerror(err))); - return err; - } + VERBOSE(VB_AUDIO, LOC + "hwparams settings, Plan B"); + period_time = latency / 4; + if (AlsaBad(snd_pcm_hw_params_set_period_time_near(pcm.handle, hwparams, + &period_time, NULL), + "failed to set period time")) + goto fini; + if (AlsaBad(snd_pcm_hw_params_get_period_size(hwparams, &pcm.period_size, + NULL), + "failed to get period size")) + goto fini; + pcm.buffer_size = pcm.period_size * 4; + if (AlsaBad(snd_pcm_hw_params_set_buffer_size_near(pcm.handle, hwparams, + &pcm.buffer_size), + "failed to set buffer size")) + goto fini; + if (AlsaBad(snd_pcm_hw_params_get_buffer_size(hwparams, + &pcm.buffer_size), + "failed to get buffer size")) + goto fini; + } + if (AlsaBad(snd_pcm_hw_params (pcm.handle, hwparams), + "failed to set hardware parameters")) + goto fini; + + result = true; + periods = 0; + AlsaBad(snd_pcm_hw_params_get_periods(hwparams, &periods, NULL), + "failed to get periods"); + VERBOSE(VB_AUDIO, LOC + pcm.logtag + + QString("latency %1msec, period size %2, periods %3, " + "buffer size %4 (in frames)") + .arg(latency /1000.0, 0, 'f', 1).arg(pcm.period_size) + .arg(periods).arg(pcm.buffer_size)); + VERBOSE(VB_AUDIO, LOC + pcm.logtag + + QString("channels %1, using %2").arg(audio_channels) + .arg((pcm.use_mmap) ? "mmap write" : "snd_pcm_writei")); + fini: + return result; +} - /* write the parameters to the playback device */ - if ((err = snd_pcm_sw_params(handle, swparams)) < 0) - { - Error(QString("Unable to set sw params for playback: %1") - .arg(snd_strerror(err))); - return err; - } +bool AudioOutputALSA::PrepSwparams(void) +{ + snd_pcm_sw_params_t* swparams; + snd_pcm_sw_params_alloca(&swparams); + snd_pcm_uframes_t boundary; + bool result = false; - if ((err = snd_pcm_prepare(handle)) < 0) - Error(QString("Initial pcm prepare err %1 %2") - .arg(err).arg(snd_strerror(err))); + if (AlsaBad(snd_pcm_sw_params_current(pcm.handle, swparams), + "failed to get swparams")) + goto fini; + + // using explicit start, not auto start + if (AlsaBad(snd_pcm_sw_params_set_start_threshold(pcm.handle, swparams, + INT_MAX), + "failed to set start threshold")) + goto fini; + if (AlsaBad(snd_pcm_sw_params_get_boundary(swparams, &boundary), + "failed to get boundary")) + goto fini; + if (AlsaBad(snd_pcm_sw_params_set_stop_threshold(pcm.handle, swparams, + boundary), + "failed to set stop threshold")) + goto fini; + if (AlsaBad(snd_pcm_sw_params(pcm.handle, swparams), + "failed to set software parameters")) + goto fini; + + result = true; - return 0; + fini: + return result; } - -int AudioOutputALSA::GetVolumeChannel(int channel) const +bool AudioOutputALSA::PrepMixer(void) { - long actual_volume; - - if (mixer_handle == NULL) - return 100; - - QByteArray mix_ctl = mixer_control.toAscii(); - snd_mixer_selem_id_t *sid; - snd_mixer_selem_id_alloca(&sid); - snd_mixer_selem_id_set_index(sid, 0); - snd_mixer_selem_id_set_name(sid, mix_ctl.constData()); + if (pcm.handle == NULL) + return false; - snd_mixer_elem_t *elem = snd_mixer_find_selem(mixer_handle, sid); - if (!elem) + int chk; + if ((chk = snd_mixer_open(&mixer.handle, 0)) < 0) { - VERBOSE(VB_IMPORTANT, QString("Mixer unable to find control %1") - .arg(mixer_control)); - return 100; + VERBOSE(VB_IMPORTANT, LOC_ERR + mixer.logtag + + QString("failed to open: %2").arg(snd_strerror(chk))); + return false; } - - snd_mixer_selem_channel_id_t chan = (snd_mixer_selem_channel_id_t) channel; - if (!snd_mixer_selem_has_playback_channel(elem, chan)) + struct snd_mixer_selem_regopt regopts = + {1, SND_MIXER_SABSTRACT_NONE, mixer.device.constData(), NULL, NULL}; + if ((chk = snd_mixer_selem_register(mixer.handle, ®opts, NULL)) < 0) + { + VERBOSE(VB_IMPORTANT, LOC_ERR + mixer.logtag + + QString("failed to register: %2").arg(snd_strerror(chk))); + snd_mixer_close(mixer.handle); + mixer.handle = NULL; + return false; + } + if ((chk = snd_mixer_load(mixer.handle)) < 0) { - snd_mixer_selem_id_set_index(sid, channel); - if ((elem = snd_mixer_find_selem(mixer_handle, sid)) == NULL) - { - VERBOSE(VB_IMPORTANT, QString("Mixer unable to find control %1 %2") - .arg(mixer_control).arg(channel)); - return 100; - } + VERBOSE(VB_IMPORTANT, LOC_ERR + mixer.logtag + + QString("failed to load: %2").arg(snd_strerror(chk))); + snd_mixer_close(mixer.handle); + mixer.handle = NULL; + return false; } - - ALSAVolumeInfo vinfo = GetVolumeRange(elem); - - snd_mixer_selem_get_playback_volume( - elem, (snd_mixer_selem_channel_id_t)channel, &actual_volume); - - return vinfo.ToMythRange(actual_volume); -} - -void AudioOutputALSA::SetVolumeChannel(int channel, int volume) -{ - SetCurrentVolume(mixer_control, channel, volume); -} - -void AudioOutputALSA::SetCurrentVolume(QString control, int channel, int volume) -{ - VERBOSE(VB_AUDIO, QString("Setting %1 volume to %2") - .arg(control).arg(volume)); - - if (!mixer_handle) - return; // no mixer, nothing to do - - QByteArray ctl = control.toAscii(); - snd_mixer_selem_id_t *sid; - snd_mixer_selem_id_alloca(&sid); - snd_mixer_selem_id_set_index(sid, 0); - snd_mixer_selem_id_set_name(sid, ctl.constData()); - - snd_mixer_elem_t *elem = snd_mixer_find_selem(mixer_handle, sid); - if (!elem) + unsigned int elcount = snd_mixer_get_count(mixer.handle); + snd_mixer_elem_t* elx = snd_mixer_first_elem(mixer.handle); + mixer.elem = NULL; + for (unsigned int ctr = 0; elx != NULL && ctr < elcount; ++ctr) + { + if (!strcmp(mixer.control.constData(), snd_mixer_selem_get_name(elx)) + && !snd_mixer_selem_is_enumerated(elx) + && snd_mixer_selem_has_playback_volume(elx) + && snd_mixer_selem_is_active(elx)) + { + mixer.elem = elx; + mixer.logtag += "control '" + mixer.control + "', "; + VERBOSE(VB_AUDIO+VB_EXTRA, LOC + mixer.logtag + + QString("playback control '%1' selected") + .arg(mixer.control.constData())); + break; + } + elx = snd_mixer_elem_next(elx); + } + if (mixer.elem == NULL) + { + VERBOSE(VB_IMPORTANT, LOC_ERR + mixer.logtag + + QString("playback control '%1' not found") + .arg(mixer.control.constData())); + snd_mixer_close(mixer.handle); + mixer.handle = NULL; + return false; + } + if (AlsaBad(snd_mixer_selem_get_playback_volume_range(mixer.elem, + &mixer.volmin, + &mixer.volmax), + "failed to get volume range")) { - VERBOSE(VB_IMPORTANT, QString("Mixer unable to find control %1") - .arg(control)); - return; + snd_mixer_close(mixer.handle); + mixer.handle = NULL; + return false; } + mixer.volrange = mixer.volmax - mixer.volmin; + VERBOSE(VB_AUDIO+VB_EXTRA, LOC + mixer.logtag + + QString("volume range - min %1, max %2, range %3") + .arg(mixer.volmin).arg(mixer.volmax).arg(mixer.volrange)); + if (set_initial_vol) + { + int initial_vol; + if ( mixer.control == "PCM") + initial_vol = gContext->GetNumSetting("PCMMixerVolume", 80); + else + initial_vol = gContext->GetNumSetting("MasterMixerVolume", 80); + for (int ch = 0; ch < audio_channels; ++ch) + SetVolumeChannel(ch, initial_vol); + } + VERBOSE(VB_AUDIO, LOC + mixer.logtag + "set up ok"); + return true; +} - snd_mixer_selem_channel_id_t chan = (snd_mixer_selem_channel_id_t) channel; - if (!snd_mixer_selem_has_playback_channel(elem, chan)) - { - snd_mixer_selem_id_set_index(sid, channel); - if ((elem = snd_mixer_find_selem(mixer_handle, sid)) == NULL) +void AudioOutputALSA::WriteMmap(unsigned char *data, snd_pcm_uframes_t nframes) +{ + unsigned char* ptr8 = data; + int16_t* ptr16 = (int16_t*)data; + int retries = 0; + int chk; + while ( nframes > 0 && retries < 3) + { + snd_pcm_hwsync(pcm.handle); + snd_pcm_sframes_t avail = snd_pcm_avail_update(pcm.handle); + if (avail < 0) + { + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + "WriteMmap failed to get avail"); + if (Recovery(avail)) + { + ++retries; + continue; + } + else + return; + } + snd_pcm_uframes_t towrite = + (nframes > (snd_pcm_uframes_t)avail) ? avail : nframes; + const snd_pcm_channel_area_t* areas; + snd_pcm_uframes_t offset; + if ((chk = snd_pcm_mmap_begin(pcm.handle, &areas, &offset, &towrite)) + < 0) { - VERBOSE(VB_IMPORTANT, - QString("mixer unable to find control %1 %2") - .arg(control).arg(channel)); + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + "mmap begin failed"); return; } + // get channel bases + unsigned char* samples[audio_channels]; + int steps[audio_channels]; + int ch; + for (ch = 0; ch < audio_channels; ++ch) + { + samples[ch] = (unsigned char*)areas[ch].addr + + (areas[ch].first + areas[ch].step * offset ) / 8; + steps[ch] = areas[ch].step / 8; + } + // wrrrrite ... + unsigned int frm; + switch( audio_bits) + { + case 16: + for (frm = towrite; frm; --frm) + for(ch = 0; ch < audio_channels; ++ch) + { + *(int16_t*)(samples[ch]) = *ptr16++; + samples[ch] += steps[ch]; + } + break; + + case 8: + for (frm = towrite; frm; --frm) + for(ch = 0; ch < audio_channels; ++ch) + { + *samples[ch] = *ptr8++; + samples[ch] += steps[ch]; + } + break; + default: + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + "invalid sample format, DirectWrite"); + return; + } + chk = snd_pcm_mmap_commit(pcm.handle, offset, towrite); + if (chk < 0) + { + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + "mmap commit failed"); + return; + } + if (chk < (int)towrite) + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + QString("short write, %1 / %2, DirectWrite") + .arg(chk).arg(towrite)); + nframes -= towrite; + ++retries; } +} - ALSAVolumeInfo vinfo = GetVolumeRange(elem); - - long set_vol = vinfo.ToALSARange(volume); - - int err = snd_mixer_selem_set_playback_volume(elem, chan, set_vol); - if (err < 0) - { - VERBOSE(VB_IMPORTANT, QString("mixer set channel %1 err %2: %3") - .arg(channel).arg(err).arg(snd_strerror(err))); - } - else - { - VERBOSE(VB_AUDIO, QString("channel %1 vol set to %2") - .arg(channel).arg(set_vol)); - } - - if (snd_mixer_selem_has_playback_switch(elem)) +void AudioOutputALSA::WriteRw(unsigned char *data, snd_pcm_uframes_t nframes) +{ + snd_pcm_uframes_t towrite; + snd_pcm_sframes_t wrote, avail; + int retries = 0; + for (wrote = 0; nframes > 0 && retries < 3; data += wrote * pcm.bytes_per_frame) { - int unmute = (0 != set_vol); - if (snd_mixer_selem_has_playback_switch_joined(elem)) + snd_pcm_hwsync(pcm.handle); + if ((avail = snd_pcm_avail_update(pcm.handle)) < 0) { - // Only mute if all the channels should be muted. - for (int i = 0; i < audio_channels; i++) + VERBOSE(VB_IMPORTANT, LOC_ERR + pcm.logtag + + "WriteRw failed to get avail"); + if (Recovery(avail)) { - if (0 != GetVolumeChannel(i)) - unmute = 1; + ++retries; + continue; } + else + return; } - - err = snd_mixer_selem_set_playback_switch(elem, chan, unmute); - if (err < 0) + towrite = (nframes < (snd_pcm_uframes_t)avail) ? nframes : avail; + wrote = snd_pcm_writei(pcm.handle, (void*)data, towrite); + if (wrote < 0) { - VERBOSE(VB_IMPORTANT, LOC_ERR + - QString("Mixer set playback switch %1 err %2: %3") - .arg(channel).arg(err).arg(snd_strerror(err))); + switch (wrote) // ==> frames written, -EBADFD, -EPIPE, -ESTRPIPE + { + case -EBADFD: + AlsaBad(-EBADFD, "unfit for writing"); + break; + case -EPIPE: + case -ESTRPIPE: + Recovery(wrote); + break; + default: + AlsaBad(wrote, "snd_pcm_writei ==> weird state"); + nframes = 0; + break; + } + wrote = 0; } else { - VERBOSE(VB_AUDIO, LOC + - QString("channel %1 playback switch set to %2") - .arg(channel).arg(unmute)); + if ((snd_pcm_uframes_t)wrote < towrite) + VERBOSE(VB_AUDIO, LOC_ERR + pcm.logtag + + QString("short write, %1 / %2 frames") + .arg(wrote).arg(towrite)); + nframes -= wrote; } + ++retries; } } -void AudioOutputALSA::OpenMixer(bool setstartingvolume) +bool AudioOutputALSA::Recovery(int err) { - int volume; - - mixer_control = gContext->GetSetting("MixerControl", "PCM"); - - SetupMixer(); - - if (mixer_handle != NULL && setstartingvolume) + if (err > 0) + err = -err; + bool isgood = false; + bool suspense = false; + switch (err) { - volume = gContext->GetNumSetting("MasterMixerVolume", 80); - SetCurrentVolume("Master", 0, volume); - SetCurrentVolume("Master", 1, volume); - - volume = gContext->GetNumSetting("PCMMixerVolume", 80); - SetCurrentVolume("PCM", 0, volume); - SetCurrentVolume("PCM", 1, volume); + case -EINTR: + isgood = true; // nuthin to see here + break; + case -ESTRPIPE: + suspense = true; + case -EPIPE: + if (!AlsaBad(snd_pcm_prepare(pcm.handle), + QString("failed to recover from %1") + .arg((suspense) ? "suspend" : "underrun"))) + isgood = true; + break; + default: + break; } + return isgood; } -void AudioOutputALSA::CloseMixer(void) -{ - if (mixer_handle != NULL) - snd_mixer_close(mixer_handle); - mixer_handle = NULL; -} - -void AudioOutputALSA::SetupMixer(void) +bool AudioOutputALSA::XrunRecovery(void) { - int err; - - QString alsadevice = gContext->GetSetting("MixerDevice", "default"); - QString device = alsadevice.remove(QString("ALSA:")); - - if (mixer_handle != NULL) - CloseMixer(); - - VERBOSE(VB_AUDIO, QString("Opening mixer %1").arg(device)); - - // TODO: This is opening card 0. Fix for case of multiple soundcards - if ((err = snd_mixer_open(&mixer_handle, 0)) < 0) - { - Warn(QString("Mixer device open error %1: %2") - .arg(err).arg(snd_strerror(err))); - mixer_handle = NULL; - return; - } - - QByteArray dev = device.toAscii(); - if ((err = snd_mixer_attach(mixer_handle, dev.constData())) < 0) - { - Warn(QString("Mixer attach error %1: %2" - "\n\t\t\tCheck Mixer Name in Setup: '%3'") - .arg(err).arg(snd_strerror(err)).arg(device)); - CloseMixer(); - return; - } - - if ((err = snd_mixer_selem_register(mixer_handle, NULL, NULL)) < 0) + bool isgood = false; + if (pcm.handle != NULL) { - Warn(QString("Mixer register error %1: %2") - .arg(err).arg(snd_strerror(err))); - CloseMixer(); - return; - } - - if ((err = snd_mixer_load(mixer_handle)) < 0) - { - Warn(QString("Mixer load error %1: %2") - .arg(err).arg(snd_strerror(err))); - CloseMixer(); - return; + if (!AlsaBad(snd_pcm_drop(pcm.handle), "pcm drop failed")) + if (!AlsaBad(snd_pcm_prepare(pcm.handle), "pcm prepare failed")) + isgood = true; + VERBOSE(VB_AUDIO, LOC + pcm.logtag + + "xrun recovery " + ((isgood) ? "good" : "not good")); } + return isgood; } -ALSAVolumeInfo AudioOutputALSA::GetVolumeRange(snd_mixer_elem_t *elem) const +bool AudioOutputALSA::AlsaBad(int op_result, QString err_msg, bool warn_only) { - long volume_min, volume_max; - - int err = snd_mixer_selem_get_playback_volume_range( - elem, &volume_min, &volume_max); - - if (err < 0) + bool isbad = (op_result < 0); // (op_result < 0) => true return + if (isbad) { - static bool first_time = true; - if (first_time) - { - VERBOSE(VB_IMPORTANT, - "snd_mixer_selem_get_playback_volume_range()" + ENO); - first_time = false; - } + QString loc_tag = (warn_only) ? LOC_WARN : LOC_ERR; + VERBOSE(VB_IMPORTANT, loc_tag + pcm.logtag + err_msg + + QString(": %1").arg(snd_strerror(op_result))); } - - ALSAVolumeInfo vinfo(volume_min, volume_max); - - VERBOSE(VB_AUDIO, QString("Volume range is %1 to %2, mult=%3") - .arg(vinfo.volume_min).arg(vinfo.volume_max) - .arg(vinfo.range_multiplier)); - - return vinfo; + return isbad; } +/* vim: set expandtab tabstop=4 shiftwidth=4: */ --- mythtv.orig/libs/libmyth/audiooutputalsa.h +++ mythtv/libs/libmyth/audiooutputalsa.h @@ -1,94 +1,82 @@ +/* + * Copyright (C) <=2008 unattributed author(s) + * Copyright (C) 2008 Alan Calvert + * + * 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. + */ + #ifndef AUDIOOUTPUTALSA #define AUDIOOUTPUTALSA -#define ALSA_PCM_NEW_HW_PARAMS_API -#define ALSA_PCM_NEW_SW_PARAMS_API -#include - -#include "audiooutputbase.h" - using namespace std; -class ALSAVolumeInfo -{ - public: - ALSAVolumeInfo(long playback_vol_min, - long playback_vol_max) : - range_multiplier(1.0f), - volume_min(playback_vol_min), volume_max(playback_vol_max) - { - float range = (float) (volume_max - volume_min); - if (range > 0.0f) - range_multiplier = 100.0f / range; - range_multiplier_inv = 1.0f / range_multiplier; - } - - int ToMythRange(long alsa_volume) - { - long toz = alsa_volume - volume_min; - int val = (int) (toz * range_multiplier); - val = (val < 0) ? 0 : val; - val = (val > 100) ? 100 : val; - return val; - } - - long ToALSARange(int myth_volume) - { - float tos = myth_volume * range_multiplier_inv; - long val = (long) (tos + volume_min + 0.5); - val = (val < volume_min) ? volume_min : val; - val = (val > volume_max) ? volume_max : val; - return val; - } - - float range_multiplier; - float range_multiplier_inv; - long volume_min; - long volume_max; -}; +#include +#include "audiooutputbase.h" class AudioOutputALSA : public AudioOutputBase { public: AudioOutputALSA(const AudioSettings &settings); - virtual ~AudioOutputALSA(); + ~AudioOutputALSA(); - // Volume control - virtual int GetVolumeChannel(int channel) const; // Returns 0-100 - virtual void SetVolumeChannel(int channel, int volume); // range 0-100 for vol + void Reset(void); + void Pause(bool paused); + void Drain(void); + int GetVolumeChannel(int channel) const; // volume range 0-100 + void SetVolumeChannel(int channel, int volume); // - protected: - // You need to implement the following functions - virtual bool OpenDevice(void); - virtual void CloseDevice(void); - virtual void WriteAudio(unsigned char *aubuf, int size); - virtual int GetSpaceOnSoundcard(void) const; - virtual int GetBufferedOnSoundcard(void) const; - - private: - inline int SetParameters(snd_pcm_t *handle, - snd_pcm_format_t format, unsigned int channels, - unsigned int rate, unsigned int buffer_time, - unsigned int period_time); - - - // Volume related - void SetCurrentVolume(QString control, int channel, int volume); - void OpenMixer(bool setstartingvolume); - void CloseMixer(void); - void SetupMixer(void); - ALSAVolumeInfo GetVolumeRange(snd_mixer_elem_t *elem) const; + bool OpenDevice(void); + void CloseDevice(void); + void WriteAudio(unsigned char *aubuf, int size); + int GetSpaceOnSoundcard(void) const; + int GetBufferedOnSoundcard(void) const; private: - snd_pcm_t *pcm_handle; - int numbadioctls; - QMutex killAudioLock; - snd_mixer_t *mixer_handle; - QString mixer_control; // e.g. "PCM" - snd_pcm_sframes_t (*pcm_write_func)(snd_pcm_t*, const void*, - snd_pcm_uframes_t); + bool PrepPCM(void); + bool PrepHwparams(void); + bool PrepSwparams(void); + bool PrepMixer(void); + void WriteMmap(unsigned char *data, snd_pcm_uframes_t nframes); + void WriteRw(unsigned char *data, snd_pcm_uframes_t nframes); + bool Recovery(int err); + bool XrunRecovery(void); + bool AlsaBad(int op_result, QString err_msg, bool warn_only = false); + + struct { + QByteArray device; + snd_pcm_t* handle; + unsigned int sample_rate; + snd_pcm_uframes_t period_size; + snd_pcm_uframes_t buffer_size; + int bytes_per_frame; // ! bytes_per_sample, grrr! + bool use_mmap; + QString logtag; + } pcm; + + struct { + QByteArray device; + QByteArray control; + snd_mixer_t* handle; + snd_mixer_elem_t* elem; + long volmin; + long volmax; + long volrange; + QString logtag; + } mixer; }; - #endif - +/* vim: set expandtab tabstop=4 shiftwidth=4: */ --- mythtv.orig/programs/mythfrontend/globalsettings.cpp +++ mythtv/programs/mythfrontend/globalsettings.cpp @@ -19,6 +19,10 @@ #include #include +#ifdef USING_ALSA +#include +#endif + // MythTV headers #include "mythconfig.h" #include "mythcontext.h" @@ -39,11 +43,14 @@ #include "mythdirs.h" #include "mythuihelper.h" +#ifdef USING_ALSA +static AlsaPlaybackPcms alsa_pcms; +#endif + static HostComboBox *AudioOutputDevice() { HostComboBox *gc = new HostComboBox("AudioOutputDevice", true); gc->setLabel(QObject::tr("Audio output device")); - #ifdef USING_ALSA gc->addSelection("ALSA:default", "ALSA:default"); gc->addSelection("ALSA:spdif", "ALSA:spdif"); @@ -52,6 +59,12 @@ gc->addSelection("ALSA:digital", "ALSA:digital"); gc->addSelection("ALSA:mixed-analog", "ALSA:mixed-analog"); gc->addSelection("ALSA:mixed-digital", "ALSA:mixed-digital"); + for (int p = 0; p < alsa_pcms.pcm_count; ++p) + if (strncasecmp(alsa_pcms.pcm_tags[p], "iec958:", 7) != 0) + { + QByteArray sel = "ALSA:" + QByteArray(alsa_pcms.pcm_tags[p]); + gc->addSelection(sel.constData(), sel.constData()); + } #endif #ifdef USING_OSS QDir dev("/dev", "dsp*", QDir::Name, QDir::System); @@ -127,7 +140,14 @@ #ifndef USING_MINGW gc->addSelection("ALSA:iec958:{ AES0 0x02 }", "ALSA:iec958:{ AES0 0x02 }"); #endif - +#ifdef USING_ALSA + for (int p = 0; p < alsa_pcms.pcm_count; ++p) + if (!strncasecmp(alsa_pcms.pcm_tags[p], "iec958:", 7)) + { + QByteArray sel = "ALSA:" + QByteArray(alsa_pcms.pcm_tags[p]); + gc->addSelection(sel.constData(), sel.constData()); + } +#endif gc->setHelpText(QObject::tr("Audio output device to use for AC3 and " "DTS passthrough. Default is the same as Audio output " "device. This value is currently only used with ALSA " @@ -5048,4 +5068,45 @@ addChild(xboxset); } +#ifdef USING_ALSA +AlsaPlaybackPcms::AlsaPlaybackPcms() +{ + for (int p = 0; p < ALSA_MAX_PCMS; ++p) + pcm_tags[p] = NULL; + pcm_count = 0; + snd_ctl_card_info_t *ctl_info = NULL; + if (snd_ctl_card_info_malloc(&ctl_info) == 0) + { + char* io; + void** hints = NULL; + void** n = NULL; + if (!(snd_device_name_hint(-1, "pcm", &hints) < 0)) + { + n = hints; + while (*n != NULL && pcm_count < ALSA_MAX_PCMS) + { + io = snd_device_name_get_hint(*n, "IOID"); + if (io == NULL || (io != NULL && strcmp(io, "Output"))) + pcm_tags[pcm_count++] = snd_device_name_get_hint(*n, "NAME"); + // memory hereby allocated is + // resolved in the destructor + if (io != NULL) + free(io); + n++; + } + snd_device_name_free_hint(hints); + } + } + if (ctl_info != NULL) + snd_ctl_card_info_free(ctl_info); +} + +AlsaPlaybackPcms::~AlsaPlaybackPcms() +{ + for( int p = 0; p < ALSA_MAX_PCMS; ++p) + if (pcm_tags[p] != NULL) + free(pcm_tags[p]); +} + +#endif // vim:set sw=4 ts=4 expandtab: --- mythtv.orig/programs/mythfrontend/globalsettings.h +++ mythtv/programs/mythfrontend/globalsettings.h @@ -165,4 +165,16 @@ }; #endif // USING_IVTV +#ifdef USING_ALSA +#define ALSA_MAX_PCMS 24 + +class AlsaPlaybackPcms +{ + public: + AlsaPlaybackPcms(); + ~AlsaPlaybackPcms(); + char* pcm_tags[ALSA_MAX_PCMS]; + int pcm_count; +}; +#endif // USING_ALSA #endif