Extremesecrecy
56d66ef7db
Removed tabs (tabs to spaces). Remove the check_deps.sh. removed libwebsocketpp-dev from dependencies. BTW, this is in cmake.yml.
2217 lines
79 KiB
C++
2217 lines
79 KiB
C++
/*
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* MIT License
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*
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* Copyright (c) 2024 Robin E. R. Davies
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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#include "pch.h"
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#include "util.hpp"
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#include "Finally.hpp"
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#include <bit>
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#include <memory>
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#include "ss.hpp"
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#include "AlsaDriver.hpp"
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#include "JackServerSettings.hpp"
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#include <thread>
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#include "RtInversionGuard.hpp"
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#include "PiPedalException.hpp"
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#include "DummyAudioDriver.hpp"
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#include "SchedulerPriority.hpp"
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#include "CrashGuard.hpp"
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#include "CpuUse.hpp"
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#include <alsa/asoundlib.h>
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#include "Lv2Log.hpp"
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#include <limits>
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#include "ss.hpp"
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#undef ALSADRIVER_CONFIG_DBG
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#ifdef ALSADRIVER_CONFIG_DBG
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#include <stdio.h>
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#endif
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using namespace pipedal;
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namespace pipedal
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{
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static bool ShouldForceStereoChannels(snd_pcm_t *pcmHandle, snd_pcm_hw_params_t *hwParams, unsigned int channelsMin, unsigned int channelsMax)
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{
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// The problem: old IC2 drivers seem to return 1-8 channels, but 8 channels is non-functinal. The assumption is that legacy drivers
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// (I2C drivers, particularl, but also the Rpi headphones device, as an interesting example) that don't support channel maps do this.
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// Hypothetically, devices could allow slection of hardware-downmixed surround channels. So deal with this case defensively.
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// The approach: check the channel map and do our best to interpret what we find.
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// No channel map, or any part of the channel map is unknown? Probably the legalcy case we're interested in. Return TRUE
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// If the channel map is a surround format, return true in that case as well.
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// If the channel map is pairwise, return false! (legitimately multi-channel devices should not be forced to stereo).
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// If the channel map is all FL/FR/MONO return false (a hypothetical configuration for a multi-channel device)
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// If the channel map is not all FL/FR/MONO, assume that it's an upmixed/downmixed surround format, and return TRUE.
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// This is high-risk code, because it attempts to anticipate hypothetical device configurations with no actual testing.
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if (channelsMax <= 2)
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return false;
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if (channelsMin == channelsMax)
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return false;
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if (channelsMin > 2)
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return false; // can't imagine what sort of device this is.
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snd_pcm_hw_params_t *test_params;
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snd_pcm_hw_params_alloca(&test_params);
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snd_pcm_hw_params_copy(test_params, hwParams);
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// can we select 2 channels?
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if (snd_pcm_hw_params_set_channels(pcmHandle, test_params, (unsigned int)2) >= 0)
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{
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snd_pcm_chmap_query_t **chmaps = snd_pcm_query_chmaps(pcmHandle);
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if (chmaps == nullptr)
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{
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return true; // probably an old driver. Do it.
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}
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Finally ff([chmaps]()
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{ snd_pcm_free_chmaps(chmaps); });
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for (size_t i = 0; chmaps[i] != nullptr; ++i)
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{
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snd_pcm_chmap_query_t *chmap = chmaps[i];
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if (chmap->map.channels == channelsMax)
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{
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switch (chmap->type)
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{
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case SND_CHMAP_TYPE_NONE:
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default:
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return true; // weird legacy case? Do it.
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case SND_CHMAP_TYPE_PAIRED:
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return false; // A legitimate multi-channel device. definitely don't do it.
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case SND_CHMAP_TYPE_VAR:
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case SND_CHMAP_TYPE_FIXED:
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{
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// we should do it for surround formats. guard against other hypothetical mappings for legitimately multi-channel devices.
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snd_pcm_chmap_position pos0 = (snd_pcm_chmap_position)(chmap->map.pos[0]);
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if (pos0 == snd_pcm_chmap_position::SND_CHMAP_MONO) // hypothetical channel map of all mono channesl.
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{
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return false; // don't do it.
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}
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if (pos0 != snd_pcm_chmap_position::SND_CHMAP_FL && pos0 != snd_pcm_chmap_position::SND_CHMAP_FL) // surround formats always start with FL. Hypothetical quad formats could start with FC.
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{
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return false; // don't do it.
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}
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// accept a hypothetical channel map of mixed FL's and FR's, FC's and MONOs. (Multi-channel with mixed mono and stereo pairs).
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// But otherwise assume it's a surround map, and use a stereo channel configuration instead.
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for (size_t i = 0; i < chmap->map.channels; ++i)
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{
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snd_pcm_chmap_position pos = (snd_pcm_chmap_position)(chmap->map.pos[i]);
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switch (pos)
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{
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case snd_pcm_chmap_position::SND_CHMAP_MONO:
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case snd_pcm_chmap_position::SND_CHMAP_FL:
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case snd_pcm_chmap_position::SND_CHMAP_FR:
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case snd_pcm_chmap_position::SND_CHMAP_FC:
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break; // keep going.
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default:
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return true; // probably a surround sound map.
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}
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}
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return false;
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};
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}
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}
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}
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return true; // no matching channel map(!??). nonsensical case. may as well use the stereo config, which might be more sensible.
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}
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return false;
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}
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struct AudioFormat
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{
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char name[40];
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snd_pcm_format_t pcm_format;
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};
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bool SetPreferredAlsaFormat(
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const char *streamType,
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snd_pcm_t *handle,
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snd_pcm_hw_params_t *hwParams,
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AudioFormat *formats,
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size_t nItems)
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{
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snd_pcm_hw_params_t *test_params;
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snd_pcm_hw_params_alloca(&test_params);
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for (size_t i = 0; i < nItems; ++i)
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{
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snd_pcm_hw_params_copy(test_params, hwParams);
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int err = snd_pcm_hw_params_set_format(handle, test_params, formats[i].pcm_format);
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if (err == 0)
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{
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int err = snd_pcm_hw_params_set_format(handle, hwParams, formats[i].pcm_format);
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if (err == 0)
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{
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return true;
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}
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}
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}
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return false;
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}
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static AudioFormat leFormats[]{
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{"32-bit float little-endian", SND_PCM_FORMAT_FLOAT_LE},
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{"32-bit integer little-endian", SND_PCM_FORMAT_S32_LE},
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{"24-bit little-endian", SND_PCM_FORMAT_S24_LE},
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{"24-bit little-endian in 3bytes format", SND_PCM_FORMAT_S24_3LE},
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{"16-bit little-endian", SND_PCM_FORMAT_S16_LE},
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};
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static AudioFormat beFormats[]{
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{"32-bit float big-endian", SND_PCM_FORMAT_FLOAT_BE},
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{"32-bit integer big-endian", SND_PCM_FORMAT_S32_BE},
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{"24-bit big-endian", SND_PCM_FORMAT_S24_BE},
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{"24-bit big-endian in 3bytes format", SND_PCM_FORMAT_S24_3BE},
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{"16-bit big-endian", SND_PCM_FORMAT_S16_BE},
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};
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[[noreturn]] static void AlsaError(const std::string &message)
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{
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throw PiPedalStateException(message);
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}
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std::string GetAlsaFormatDescription(snd_pcm_format_t format)
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{
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for (size_t i = 0; i < sizeof(beFormats) / sizeof(beFormats[0]); ++i)
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{
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if (beFormats[i].pcm_format == format)
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{
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return beFormats[i].name;
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}
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}
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for (size_t i = 0; i < sizeof(leFormats) / sizeof(leFormats[0]); ++i)
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{
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if (leFormats[i].pcm_format == format)
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{
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return leFormats[i].name;
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}
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}
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return "Unknown format.";
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}
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void SetPreferredAlsaFormat(
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const std::string &alsa_device_name,
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const char *streamType,
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snd_pcm_t *handle,
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snd_pcm_hw_params_t *hwParams)
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{
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int err;
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if (std::endian::native == std::endian::big)
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{
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if (SetPreferredAlsaFormat(streamType, handle, hwParams, beFormats, sizeof(beFormats) / sizeof(beFormats[0])))
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return;
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if (SetPreferredAlsaFormat(streamType, handle, hwParams, leFormats, sizeof(leFormats) / sizeof(leFormats[0])))
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return;
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}
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else
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{
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if (SetPreferredAlsaFormat(streamType, handle, hwParams, leFormats, sizeof(leFormats) / sizeof(leFormats[0])))
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return;
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if (SetPreferredAlsaFormat(streamType, handle, hwParams, beFormats, sizeof(beFormats) / sizeof(beFormats[0])))
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return;
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}
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AlsaError(SS("No supported audio formats (" << alsa_device_name << "/" << streamType << ")"));
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}
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class AlsaDriverImpl : public AudioDriver
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{
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private:
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pipedal::CpuUse cpuUse;
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#ifdef ALSADRIVER_CONFIG_DBG
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snd_output_t *snd_output = nullptr;
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snd_pcm_status_t *snd_status = nullptr;
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#endif
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uint32_t sampleRate = 0;
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uint32_t bufferSize;
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uint32_t numberOfBuffers;
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int playbackChannels = 0;
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int captureChannels = 0;
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uint32_t user_threshold = 0;
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bool soft_mode = false;
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snd_pcm_format_t captureFormat = snd_pcm_format_t::SND_PCM_FORMAT_UNKNOWN;
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uint32_t playbackSampleSize = 0;
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uint32_t captureSampleSize = 0;
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uint32_t playbackFrameSize = 0;
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uint32_t captureFrameSize = 0;
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using CopyFunction = void (AlsaDriverImpl::*)(size_t frames);
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CopyFunction copyInputFn;
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CopyFunction copyOutputFn;
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bool inputSwapped = false;
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bool outputSwapped = false;
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std::vector<float *> activeCaptureBuffers;
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std::vector<float *> activePlaybackBuffers;
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std::vector<float *> captureBuffers;
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std::vector<float *> playbackBuffers;
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std::vector<uint8_t> rawCaptureBuffer;
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std::vector<uint8_t> rawPlaybackBuffer;
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AudioDriverHost *driverHost = nullptr;
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void validate_capture_handle()
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{ // leftover debugging for a buffer overrun :-/
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#ifdef DEBUG
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if (snd_pcm_type(captureHandle) != SND_PCM_TYPE_HW)
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{
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throw std::runtime_error("Capture handle has been overwritten");
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}
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#endif
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}
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public:
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AlsaDriverImpl(AudioDriverHost *driverHost)
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: driverHost(driverHost)
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{
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midiEventMemoryIndex = 0;
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midiEventMemory.resize(MIDI_MEMORY_BUFFER_SIZE);
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midiEvents.resize(MAX_MIDI_EVENT);
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}
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virtual ~AlsaDriverImpl()
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{
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Close();
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#ifdef ALSADRIVER_CONFIG_DBG
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if (snd_output)
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{
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snd_output_close(snd_output);
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snd_output = nullptr;
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}
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if (snd_status)
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{
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snd_pcm_status_free(snd_status);
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snd_status = nullptr;
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}
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#endif
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}
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private:
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void OnShutdown()
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{
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Lv2Log::info("ALSA Audio Server has shut down.");
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}
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static void
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jack_shutdown_fn(void *arg)
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{
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((AlsaDriverImpl *)arg)->OnShutdown();
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}
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static int xrun_callback_fn(void *arg)
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{
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((AudioDriverHost *)arg)->OnUnderrun();
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return 0;
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}
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virtual uint32_t GetSampleRate()
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{
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return this->sampleRate;
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}
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JackServerSettings jackServerSettings;
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std::string alsa_device_name;
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snd_pcm_t *playbackHandle = nullptr;
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snd_pcm_t *captureHandle = nullptr;
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unsigned int periods = 0;
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snd_pcm_hw_params_t *captureHwParams = nullptr;
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snd_pcm_sw_params_t *captureSwParams = nullptr;
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snd_pcm_hw_params_t *playbackHwParams = nullptr;
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snd_pcm_sw_params_t *playbackSwParams = nullptr;
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bool capture_and_playback_not_synced = false;
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std::mutex terminateSync;
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std::atomic<bool> terminateAudio_ = false;
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void terminateAudio(bool terminate)
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{
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this->terminateAudio_ = terminate;
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}
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bool terminateAudio()
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{
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return this->terminateAudio_;
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}
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private:
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void AlsaCleanup()
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{
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if (captureHandle)
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{
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snd_pcm_close(captureHandle);
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captureHandle = nullptr;
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}
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if (playbackHandle)
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{
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snd_pcm_close(playbackHandle);
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playbackHandle = nullptr;
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}
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if (captureHwParams)
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{
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snd_pcm_hw_params_free(captureHwParams);
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captureHwParams = nullptr;
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}
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if (captureSwParams)
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{
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snd_pcm_sw_params_free(captureSwParams);
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captureSwParams = nullptr;
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}
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if (playbackHwParams)
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{
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snd_pcm_hw_params_free(playbackHwParams);
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playbackHwParams = nullptr;
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}
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if (playbackSwParams)
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{
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snd_pcm_sw_params_free(playbackSwParams);
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playbackSwParams = nullptr;
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}
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this->alsaSequencer = nullptr;
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}
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std::string discover_alsa_using_apps()
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{
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return ""; // xxx fix me.
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}
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void AlsaConfigureStream(
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const std::string &alsa_device_name,
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const char *streamType,
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snd_pcm_t *handle,
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snd_pcm_hw_params_t *hwParams,
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snd_pcm_sw_params_t *swParams,
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int *channels,
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unsigned int *periods)
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{
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int err;
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snd_pcm_uframes_t stop_th;
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if ((err = snd_pcm_hw_params_any(handle, hwParams)) < 0)
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{
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AlsaError(SS("No playback configurations available (" << snd_strerror(err) << ")"));
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}
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err = snd_pcm_hw_params_set_access(handle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0)
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{
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AlsaError("snd_pcm_hw_params_set_access failed.");
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}
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SetPreferredAlsaFormat(alsa_device_name, streamType, handle, hwParams);
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unsigned int sampleRate = (unsigned int)this->sampleRate;
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err = snd_pcm_hw_params_set_rate_near(handle, hwParams,
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&sampleRate, NULL);
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this->sampleRate = sampleRate;
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if (err < 0)
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{
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AlsaError(SS("Can't set sample rate to " << this->sampleRate << " (" << alsa_device_name << "/" << streamType << ")"));
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}
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if (!*channels)
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{
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/*if not user-specified, try to find the maximum
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* number of channels */
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unsigned int channels_max = 0;
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unsigned int channels_min = 0;
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err = snd_pcm_hw_params_get_channels_max(hwParams,
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&channels_max);
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if (err < 0)
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{
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AlsaError(SS("Can't get channels_max."));
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}
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err = snd_pcm_hw_params_get_channels_min(hwParams,
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&channels_min);
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if (err < 0)
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{
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AlsaError(SS("Can't get channels_min."));
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}
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*channels = channels_max;
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if (ShouldForceStereoChannels(handle, hwParams, channels_min, channels_max))
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{
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*channels = 2;
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}
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if (*channels >= 1024)
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{
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// The default PCM device has unlimited channels.
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// report 2 channels
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*channels = 2;
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}
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}
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if ((err = snd_pcm_hw_params_set_channels(handle, hwParams,
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*channels)) < 0)
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{
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AlsaError(SS("Can't set channel count to " << *channels << " (" << alsa_device_name << "/" << streamType << ")"));
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}
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snd_pcm_uframes_t effectivePeriodSize = this->bufferSize;
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int dir = 0;
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if ((err = snd_pcm_hw_params_set_period_size_near(handle, hwParams,
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&effectivePeriodSize,
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&dir)) < 0)
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{
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AlsaError(SS("Can't set period size to " << this->bufferSize << " (" << alsa_device_name << "/" << streamType << ")"));
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}
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this->bufferSize = effectivePeriodSize;
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*periods = this->numberOfBuffers;
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dir = 0;
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snd_pcm_hw_params_set_periods_min(handle, hwParams, periods, &dir);
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if (*periods < this->numberOfBuffers)
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*periods = this->numberOfBuffers;
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if (snd_pcm_hw_params_set_periods_near(handle, hwParams,
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periods, NULL) < 0)
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{
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|
AlsaError(SS("Can't set number of periods to " << (*periods) << " (" << alsa_device_name << "/" << streamType << ")"));
|
|
}
|
|
|
|
if (*periods < this->numberOfBuffers)
|
|
{
|
|
AlsaError(SS("Got smaller periods " << *periods << " than " << this->numberOfBuffers));
|
|
}
|
|
|
|
snd_pcm_uframes_t bSize;
|
|
|
|
// if ((err = snd_pcm_hw_params_set_buffer_size(handle, hwParams,
|
|
// *periods *
|
|
// this->bufferSize)) < 0)
|
|
// {
|
|
// AlsaError(SS("Can't set buffer length to " << (*periods * this->bufferSize)));
|
|
// }
|
|
|
|
if ((err = snd_pcm_hw_params(handle, hwParams)) < 0)
|
|
{
|
|
AlsaError(SS("Cannot set hardware parameters for " << alsa_device_name));
|
|
}
|
|
|
|
snd_pcm_sw_params_current(handle, swParams);
|
|
|
|
if (handle == this->captureHandle)
|
|
{
|
|
if ((err = snd_pcm_sw_params_set_start_threshold(handle, swParams,
|
|
0)) < 0)
|
|
{
|
|
AlsaError(SS("Cannot set start mode for " << alsa_device_name));
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if ((err = snd_pcm_sw_params_set_start_threshold(handle, swParams,
|
|
0x7fffffff)) < 0)
|
|
{
|
|
AlsaError(SS("Cannot set start mode for " << alsa_device_name));
|
|
}
|
|
}
|
|
|
|
stop_th = *periods * this->bufferSize;
|
|
if (this->soft_mode)
|
|
{
|
|
stop_th = (snd_pcm_uframes_t)-1;
|
|
}
|
|
|
|
if ((err = snd_pcm_sw_params_set_stop_threshold(
|
|
handle, swParams, stop_th)) < 0)
|
|
{
|
|
AlsaError(SS("ALSA: cannot set stop mode for " << alsa_device_name));
|
|
}
|
|
|
|
if ((err = snd_pcm_sw_params_set_silence_threshold(
|
|
handle, swParams, 0)) < 0)
|
|
{
|
|
AlsaError(SS("Cannot set silence threshold for " << alsa_device_name));
|
|
}
|
|
|
|
if (handle == this->playbackHandle)
|
|
err = snd_pcm_sw_params_set_avail_min(
|
|
handle, swParams,
|
|
this->bufferSize * (*periods - this->numberOfBuffers + 1));
|
|
else
|
|
err = snd_pcm_sw_params_set_avail_min(
|
|
handle, swParams, this->bufferSize);
|
|
|
|
if (err < 0)
|
|
{
|
|
AlsaError(SS("Cannot set avail min for " << alsa_device_name));
|
|
}
|
|
|
|
// err = snd_pcm_sw_params_set_tstamp_mode(handle, swParams, SND_PCM_TSTAMP_ENABLE);
|
|
// if (err < 0)
|
|
// {
|
|
// Lv2Log::info(SS(
|
|
// "Could not enable ALSA time stamp mode for " << alsa_device_name << " (err " << err << ")"));
|
|
// }
|
|
|
|
#if SND_LIB_MAJOR >= 1 && SND_LIB_MINOR >= 1
|
|
err = snd_pcm_sw_params_set_tstamp_type(handle, swParams, SND_PCM_TSTAMP_TYPE_MONOTONIC);
|
|
if (err < 0)
|
|
{
|
|
Lv2Log::info(SS(
|
|
"Could not use monotonic ALSA time stamps for " << alsa_device_name << "(err " << err << ")"));
|
|
}
|
|
#endif
|
|
|
|
if ((err = snd_pcm_sw_params(handle, swParams)) < 0)
|
|
{
|
|
AlsaError(SS("Cannot set software parameters for " << alsa_device_name));
|
|
}
|
|
err = snd_pcm_prepare(handle);
|
|
if (err < 0)
|
|
{
|
|
AlsaError(SS("ALSA prepare failed. " << snd_strerror(err)));
|
|
}
|
|
}
|
|
void SetAlsaParameters(uint32_t bufferSize, uint32_t numberOfBuffers, uint32_t sampleRate)
|
|
{
|
|
this->bufferSize = bufferSize;
|
|
this->numberOfBuffers = numberOfBuffers;
|
|
this->sampleRate = sampleRate;
|
|
|
|
if (this->captureHandle)
|
|
{
|
|
this->alsa_device_name = this->jackServerSettings.GetAlsaInputDevice();
|
|
AlsaConfigureStream(
|
|
this->alsa_device_name,
|
|
"capture",
|
|
captureHandle,
|
|
captureHwParams,
|
|
captureSwParams,
|
|
&captureChannels,
|
|
&this->periods);
|
|
}
|
|
if (this->playbackHandle)
|
|
{
|
|
this->alsa_device_name = this->jackServerSettings.GetAlsaOutputDevice();
|
|
AlsaConfigureStream(
|
|
this->alsa_device_name,
|
|
"playback",
|
|
playbackHandle,
|
|
playbackHwParams,
|
|
playbackSwParams,
|
|
&playbackChannels,
|
|
&this->periods);
|
|
}
|
|
|
|
#ifdef ALSADRIVER_CONFIG_DBG
|
|
snd_pcm_dump(captureHandle, snd_output);
|
|
snd_pcm_dump(playbackHandle, snd_output);
|
|
#endif
|
|
}
|
|
|
|
int32_t EndianSwap(int32_t v)
|
|
{
|
|
int32_t b0 = v & 0xFF;
|
|
int32_t b1 = (v >> 8) & 0xFF;
|
|
int32_t b2 = (v >> 16) & 0xFF;
|
|
int32_t b3 = (v >> 24) & 0xFF;
|
|
|
|
return (b0 << 24) | (b1 << 16) | (b2 << 8) | (b3);
|
|
}
|
|
int16_t EndianSwap(int16_t v)
|
|
{
|
|
int16_t b0 = v & 0xFF;
|
|
int16_t b1 = (v >> 8) & 0xFF;
|
|
|
|
return (b0 << 8) | (b1);
|
|
}
|
|
void EndianSwap(float *p, float v_)
|
|
{
|
|
int32_t v = EndianSwap(*(int32_t *)&v_);
|
|
*(int32_t *)p = v;
|
|
}
|
|
template <typename T>
|
|
static T *getCaptureBuffer(std::vector<uint8_t> &buffer) { return (T *)(buffer.data()); }
|
|
|
|
void CopyCaptureFloatBe(size_t frames)
|
|
{
|
|
int32_t *p = getCaptureBuffer<int32_t>(rawCaptureBuffer);
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int32_t v = EndianSwap(*p);
|
|
++p;
|
|
|
|
*(int32_t *)(buffers[channel] + frame) = v;
|
|
}
|
|
}
|
|
}
|
|
|
|
void CopyCaptureFloatLe(size_t frames)
|
|
{
|
|
float *p = getCaptureBuffer<float>(rawCaptureBuffer);
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = *p++;
|
|
buffers[channel][frame] = v;
|
|
}
|
|
}
|
|
}
|
|
|
|
void CopyCaptureS16Le(size_t frames)
|
|
{
|
|
int16_t *p = getCaptureBuffer<int16_t>(rawCaptureBuffer);
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
constexpr double scale = 1.0f / (std::numeric_limits<int16_t>::max() + 1L);
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int16_t v = *p++;
|
|
buffers[channel][frame] = scale * v;
|
|
}
|
|
}
|
|
}
|
|
void CopyCaptureS16Be(size_t frames)
|
|
{
|
|
int16_t *p = getCaptureBuffer<int16_t>(rawCaptureBuffer);
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
constexpr float scale = 1.0f / (std::numeric_limits<int16_t>::max() + 1L);
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int16_t v = EndianSwap(*p++);
|
|
buffers[channel][frame] = scale * v;
|
|
}
|
|
}
|
|
}
|
|
|
|
void CopyCaptureS32Le(size_t frames)
|
|
{
|
|
int32_t *p = getCaptureBuffer<int32_t>(rawCaptureBuffer);
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
constexpr float scale = 1.0f / (std::numeric_limits<int32_t>::max() + 1L);
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int32_t v = *p++;
|
|
buffers[channel][frame] = scale * v;
|
|
}
|
|
}
|
|
}
|
|
void CopyCaptureS24_3Le(size_t frames)
|
|
{
|
|
uint8_t *p = getCaptureBuffer<uint8_t>(rawCaptureBuffer);
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
constexpr float scale = 1.0f / (std::numeric_limits<int32_t>::max() + 1LL);
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int32_t v = (p[0] << 8) + (p[1] << 16) | (p[2] << 24);
|
|
p += 3;
|
|
buffers[channel][frame] = scale * v;
|
|
}
|
|
}
|
|
}
|
|
void CopyCaptureS24_3Be(size_t frames)
|
|
{
|
|
uint8_t *p = (uint8_t *)rawCaptureBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
constexpr float scale = 1.0f / (std::numeric_limits<int32_t>::max() + 1LL);
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int32_t v = (p[2] << 8) + (p[1] << 16) | (p[0] << 24);
|
|
p += 3;
|
|
buffers[channel][frame] = scale * v;
|
|
}
|
|
}
|
|
}
|
|
void CopyCaptureS24Le(size_t frames)
|
|
{
|
|
int32_t *p = (int32_t *)rawCaptureBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
constexpr float scale = 1.0f / (0x00FFFFFFL + 1L);
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int32_t v = *p++;
|
|
buffers[channel][frame] = scale * v;
|
|
}
|
|
}
|
|
}
|
|
void CopyCaptureS24Be(size_t frames)
|
|
{
|
|
int32_t *p = (int32_t *)rawCaptureBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
constexpr float scale = 1.0f / (0x00FFFFFFL + 1L);
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int32_t v = EndianSwap(*p++);
|
|
buffers[channel][frame] = scale * v;
|
|
}
|
|
}
|
|
}
|
|
void CopyCaptureS32Be(size_t frames)
|
|
{
|
|
int32_t *p = (int32_t *)rawCaptureBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->captureBuffers;
|
|
int channels = this->captureChannels;
|
|
constexpr float scale = 1.0f / (std::numeric_limits<int32_t>::max() + 1L);
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
int32_t v = EndianSwap(*p++);
|
|
buffers[channel][frame] = scale * v;
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackS16Le(size_t frames)
|
|
{
|
|
int16_t *p = (int16_t *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->playbackChannels;
|
|
constexpr float scale = std::numeric_limits<int16_t>::max();
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
if (v > 1.0f)
|
|
v = 1.0f;
|
|
else if (v < -1.0f)
|
|
v = -1.0f;
|
|
*p++ = (int16_t)(scale * v);
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackS16Be(size_t frames)
|
|
{
|
|
int16_t *p = (int16_t *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->playbackChannels;
|
|
constexpr float scale = std::numeric_limits<int16_t>::max();
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
if (v > 1.0f)
|
|
v = 1.0f;
|
|
else if (v < -1.0f)
|
|
v = -1.0f;
|
|
*p++ = EndianSwap((int16_t)(scale * v));
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackS32Le(size_t frames)
|
|
{
|
|
int32_t *p = (int32_t *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->playbackChannels;
|
|
constexpr double scale = std::numeric_limits<int32_t>::max();
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
if (v > 1.0f)
|
|
v = 1.0f;
|
|
else if (v < -1.0f)
|
|
v = -1.0f;
|
|
*p++ = (int32_t)(scale * v);
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackS24Le(size_t frames)
|
|
{
|
|
// 24 bits in low bits of an int32_t.
|
|
|
|
int32_t *p = (int32_t *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->playbackChannels;
|
|
constexpr double scale = 0x00FFFFFF;
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
if (v > 1.0f)
|
|
v = 1.0f;
|
|
else if (v < -1.0f)
|
|
v = -1.0f;
|
|
*p++ = (int32_t)(scale * v);
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackS24Be(size_t frames)
|
|
{
|
|
// 24 bits in low bits of an int32_t.
|
|
|
|
int32_t *p = (int32_t *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->playbackChannels;
|
|
constexpr double scale = 0x00FFFFFF;
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
if (v > 1.0f)
|
|
v = 1.0f;
|
|
else if (v < -1.0f)
|
|
v = -1.0f;
|
|
*p++ = EndianSwap((int32_t)(scale * v));
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackS32Be(size_t frames)
|
|
{
|
|
int32_t *p = (int32_t *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->playbackChannels;
|
|
constexpr double scale = std::numeric_limits<int32_t>::max();
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
if (v > 1.0f)
|
|
v = 1.0f;
|
|
else if (v < -1.0f)
|
|
v = -1.0f;
|
|
*p++ = EndianSwap((int32_t)(scale * v));
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackS24_3Be(size_t frames)
|
|
{
|
|
uint8_t *p = (uint8_t *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->playbackChannels;
|
|
constexpr double scale = std::numeric_limits<int32_t>::max();
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
if (v > 1.0f)
|
|
v = 1.0f;
|
|
else if (v < -1.0f)
|
|
v = -1.0f;
|
|
int32_t iValue = (int32_t)(scale * v);
|
|
p[0] = (uint8_t)(iValue >> 24);
|
|
p[1] = (uint8_t)(iValue >> 16);
|
|
p[2] = (uint8_t)(iValue >> 8);
|
|
|
|
p += 3;
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackS24_3Le(size_t frames)
|
|
{
|
|
uint8_t *p = (uint8_t *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->playbackChannels;
|
|
constexpr double scale = std::numeric_limits<int32_t>::max();
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
if (v > 1.0f)
|
|
v = 1.0f;
|
|
else if (v < -1.0f)
|
|
v = -1.0f;
|
|
int32_t iValue = (int32_t)(scale * v);
|
|
p[0] = (uint8_t)(iValue >> 8);
|
|
p[1] = (uint8_t)(iValue >> 16);
|
|
p[2] = (uint8_t)(iValue >> 24);
|
|
|
|
p += 3;
|
|
}
|
|
}
|
|
}
|
|
|
|
void CopyPlaybackFloatLe(size_t frames)
|
|
{
|
|
float *p = (float *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->captureChannels;
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
*p++ = v;
|
|
}
|
|
}
|
|
}
|
|
void CopyPlaybackFloatBe(size_t frames)
|
|
{
|
|
float *p = (float *)rawPlaybackBuffer.data();
|
|
|
|
std::vector<float *> &buffers = this->playbackBuffers;
|
|
int channels = this->captureChannels;
|
|
for (size_t frame = 0; frame < frames; ++frame)
|
|
{
|
|
for (int channel = 0; channel < channels; ++channel)
|
|
{
|
|
float v = buffers[channel][frame];
|
|
EndianSwap(p, v);
|
|
p++;
|
|
}
|
|
}
|
|
}
|
|
|
|
public:
|
|
void TestFormatEncodeDecode(snd_pcm_format_t captureFormat);
|
|
|
|
private:
|
|
void AllocateBuffers(std::vector<float *> &buffers, size_t n)
|
|
{
|
|
buffers.resize(n);
|
|
for (size_t i = 0; i < n; ++i)
|
|
{
|
|
buffers[i] = new float[this->bufferSize];
|
|
for (size_t j = 0; j < this->bufferSize; ++j)
|
|
{
|
|
buffers[i][j] = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
JackChannelSelection channelSelection;
|
|
bool open = false;
|
|
virtual void Open(const JackServerSettings &jackServerSettings, const JackChannelSelection &channelSelection)
|
|
{
|
|
terminateAudio_ = false;
|
|
if (open)
|
|
{
|
|
throw PiPedalStateException("Already open.");
|
|
}
|
|
this->jackServerSettings = jackServerSettings;
|
|
this->channelSelection = channelSelection;
|
|
|
|
open = true;
|
|
try
|
|
{
|
|
OpenAudio(jackServerSettings, channelSelection);
|
|
std::atomic_thread_fence(std::memory_order::release);
|
|
}
|
|
catch (const std::exception &e)
|
|
{
|
|
std::atomic_thread_fence(std::memory_order::release);
|
|
|
|
Close();
|
|
throw;
|
|
}
|
|
}
|
|
|
|
void PrepareCaptureFunctions(snd_pcm_format_t captureFormat)
|
|
{
|
|
this->captureFormat = captureFormat;
|
|
|
|
switch (captureFormat)
|
|
{
|
|
case SND_PCM_FORMAT_FLOAT_LE:
|
|
captureSampleSize = 4;
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureFloatLe;
|
|
break;
|
|
case SND_PCM_FORMAT_S24_3LE:
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureS24_3Le;
|
|
captureSampleSize = 3;
|
|
break;
|
|
case SND_PCM_FORMAT_S32_LE:
|
|
captureSampleSize = 4;
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureS32Le;
|
|
break;
|
|
case SND_PCM_FORMAT_S24_LE:
|
|
captureSampleSize = 4;
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureS24Le;
|
|
break;
|
|
case SND_PCM_FORMAT_S16_LE:
|
|
captureSampleSize = 2;
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureS16Le;
|
|
break;
|
|
case SND_PCM_FORMAT_FLOAT_BE:
|
|
captureSampleSize = 4;
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureFloatBe;
|
|
captureSampleSize = 4;
|
|
break;
|
|
case SND_PCM_FORMAT_S24_3BE:
|
|
captureSampleSize = 3;
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureS24_3Be;
|
|
break;
|
|
case SND_PCM_FORMAT_S32_BE:
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureS32Be;
|
|
captureSampleSize = 4;
|
|
break;
|
|
case SND_PCM_FORMAT_S24_BE:
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureS24Be;
|
|
captureSampleSize = 4;
|
|
break;
|
|
case SND_PCM_FORMAT_S16_BE:
|
|
copyInputFn = &AlsaDriverImpl::CopyCaptureS16Be;
|
|
captureSampleSize = 2;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
if (copyInputFn == nullptr)
|
|
{
|
|
throw PiPedalStateException(SS("Audio input format not supported. (" << captureFormat << ")"));
|
|
}
|
|
|
|
captureFrameSize = captureSampleSize * captureChannels;
|
|
rawCaptureBuffer.resize(captureFrameSize * bufferSize);
|
|
memset(rawCaptureBuffer.data(), 0, captureFrameSize * bufferSize);
|
|
|
|
AllocateBuffers(captureBuffers, captureChannels);
|
|
}
|
|
|
|
virtual std::string GetConfigurationDescription()
|
|
{
|
|
std::string result = SS(
|
|
"ALSA, "
|
|
<< this->alsa_device_name
|
|
<< ", " << GetAlsaFormatDescription(this->captureFormat)
|
|
<< ", " << this->sampleRate
|
|
<< ", " << this->bufferSize << "x" << this->numberOfBuffers
|
|
<< ", in: " << this->InputBufferCount() << "/" << this->captureChannels
|
|
<< ", out: " << this->OutputBufferCount() << "/" << this->playbackChannels);
|
|
return result;
|
|
}
|
|
void PreparePlaybackFunctions(snd_pcm_format_t playbackFormat)
|
|
{
|
|
copyOutputFn = nullptr;
|
|
switch (playbackFormat)
|
|
{
|
|
case SND_PCM_FORMAT_FLOAT_LE:
|
|
playbackSampleSize = 4;
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackFloatLe;
|
|
break;
|
|
case SND_PCM_FORMAT_S24_3LE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackS24_3Le;
|
|
playbackSampleSize = 3;
|
|
break;
|
|
case SND_PCM_FORMAT_S32_LE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackS32Le;
|
|
playbackSampleSize = 4;
|
|
break;
|
|
case SND_PCM_FORMAT_S24_LE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackS24Le;
|
|
playbackSampleSize = 4;
|
|
break;
|
|
case SND_PCM_FORMAT_S16_LE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackS16Le;
|
|
playbackSampleSize = 2;
|
|
break;
|
|
case SND_PCM_FORMAT_FLOAT_BE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackFloatBe;
|
|
playbackSampleSize = 4;
|
|
break;
|
|
case SND_PCM_FORMAT_S24_3BE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackS24_3Be;
|
|
playbackSampleSize = 3;
|
|
break;
|
|
case SND_PCM_FORMAT_S32_BE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackS32Be;
|
|
playbackSampleSize = 4;
|
|
break;
|
|
case SND_PCM_FORMAT_S24_BE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackS24Be;
|
|
playbackSampleSize = 4;
|
|
break;
|
|
case SND_PCM_FORMAT_S16_BE:
|
|
copyOutputFn = &AlsaDriverImpl::CopyPlaybackS16Be;
|
|
playbackSampleSize = 2;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
if (copyOutputFn == nullptr)
|
|
{
|
|
throw PiPedalStateException(SS("Unsupported audio output format. (" << playbackFormat << ")"));
|
|
}
|
|
|
|
playbackFrameSize = playbackSampleSize * playbackChannels;
|
|
rawPlaybackBuffer.resize(playbackFrameSize * bufferSize);
|
|
memset(rawPlaybackBuffer.data(), 0, playbackFrameSize * bufferSize);
|
|
|
|
AllocateBuffers(playbackBuffers, playbackChannels);
|
|
}
|
|
|
|
void OpenAudio(const JackServerSettings &jackServerSettings, const JackChannelSelection &channelSelection)
|
|
{
|
|
int err;
|
|
|
|
std::string inputName = jackServerSettings.GetAlsaInputDevice();
|
|
std::string outputName = jackServerSettings.GetAlsaOutputDevice();
|
|
|
|
|
|
this->numberOfBuffers = jackServerSettings.GetNumberOfBuffers();
|
|
this->bufferSize = jackServerSettings.GetBufferSize();
|
|
this->user_threshold = jackServerSettings.GetBufferSize();
|
|
|
|
try
|
|
{
|
|
|
|
this->alsa_device_name = outputName;
|
|
err = snd_pcm_open(&playbackHandle, outputName.c_str(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
|
|
if (err < 0)
|
|
{
|
|
switch (errno)
|
|
{
|
|
case EBUSY:
|
|
{
|
|
std::string apps = discover_alsa_using_apps();
|
|
std::string message;
|
|
if (apps.size() != 0)
|
|
{
|
|
message =
|
|
SS("Device " << alsa_device_name << " in use. The following applications are using your soundcard: " << apps
|
|
<< ". Stop them as neccesary before trying to pipedald.");
|
|
}
|
|
else
|
|
{
|
|
message =
|
|
SS("Device " << alsa_device_name << " in use. Stop the application using it before trying to restart pipedald. ");
|
|
}
|
|
Lv2Log::error(message);
|
|
throw PiPedalStateException(std::move(message));
|
|
}
|
|
break;
|
|
case EPERM:
|
|
throw PiPedalStateException(SS("Permission denied opening device '" << alsa_device_name << "'"));
|
|
default:
|
|
throw PiPedalStateException(SS("Unexepected error (" << errno << ") opening device '" << alsa_device_name << "'"));
|
|
}
|
|
}
|
|
if (this->playbackHandle)
|
|
{
|
|
snd_pcm_nonblock(playbackHandle, 0);
|
|
}
|
|
|
|
this->alsa_device_name = inputName;
|
|
err = snd_pcm_open(&captureHandle, inputName.c_str(), SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
|
|
|
|
if (err < 0)
|
|
{
|
|
switch (errno)
|
|
{
|
|
case EBUSY:
|
|
{
|
|
std::string apps = discover_alsa_using_apps();
|
|
std::string message;
|
|
if (apps.size() != 0)
|
|
{
|
|
message =
|
|
SS("Device " << alsa_device_name << " in use. The following applications are using your soundcard: " << apps
|
|
<< ". Stop them as neccesary before trying to restart pipedald.");
|
|
}
|
|
else
|
|
{
|
|
message =
|
|
SS("Device " << alsa_device_name << " in use. Stop the application using it before trying to restart pipedald. ");
|
|
}
|
|
Lv2Log::error(message);
|
|
throw PiPedalStateException(std::move(message));
|
|
}
|
|
break;
|
|
case EPERM:
|
|
throw PiPedalStateException(SS("Permission denied opening device '" << alsa_device_name << "'"));
|
|
default:
|
|
throw PiPedalStateException(SS("Unexepected error (" << errno << ") opening device '" << alsa_device_name << "'"));
|
|
}
|
|
}
|
|
if (this->captureHandle)
|
|
{
|
|
snd_pcm_nonblock(captureHandle, 0);
|
|
}
|
|
|
|
if ((err = snd_pcm_hw_params_malloc(&captureHwParams)) < 0)
|
|
{
|
|
throw PiPedalStateException("Failed to allocate captureHwParams");
|
|
}
|
|
if ((err = snd_pcm_sw_params_malloc(&captureSwParams)) < 0)
|
|
{
|
|
throw PiPedalStateException("Failed to allocate captureSwParams");
|
|
}
|
|
if ((err = snd_pcm_hw_params_malloc(&playbackHwParams)) < 0)
|
|
{
|
|
throw PiPedalStateException("Failed to allocate playbackHwParams");
|
|
}
|
|
if ((err = snd_pcm_sw_params_malloc(&playbackSwParams)) < 0)
|
|
{
|
|
throw PiPedalStateException("Failed to allocate playbackSwParams");
|
|
}
|
|
|
|
SetAlsaParameters(jackServerSettings.GetBufferSize(), jackServerSettings.GetNumberOfBuffers(), jackServerSettings.GetSampleRate());
|
|
capture_and_playback_not_synced = false;
|
|
|
|
if (captureHandle && playbackHandle)
|
|
{
|
|
if (snd_pcm_link(playbackHandle,
|
|
captureHandle) != 0)
|
|
{
|
|
capture_and_playback_not_synced = true;
|
|
}
|
|
}
|
|
|
|
snd_pcm_format_t captureFormat;
|
|
snd_pcm_hw_params_get_format(captureHwParams, &captureFormat);
|
|
copyInputFn = nullptr;
|
|
|
|
PrepareCaptureFunctions(captureFormat);
|
|
|
|
snd_pcm_format_t playbackFormat;
|
|
snd_pcm_hw_params_get_format(playbackHwParams, &playbackFormat);
|
|
|
|
PreparePlaybackFunctions(playbackFormat);
|
|
}
|
|
catch (const std::exception &e)
|
|
{
|
|
AlsaCleanup();
|
|
throw;
|
|
}
|
|
}
|
|
|
|
void FillOutputBuffer()
|
|
{
|
|
validate_capture_handle();
|
|
|
|
memset(rawPlaybackBuffer.data(), 0, playbackFrameSize * bufferSize);
|
|
int retry = 0;
|
|
while (true)
|
|
{
|
|
auto avail = snd_pcm_avail(this->playbackHandle);
|
|
if (avail < 0)
|
|
{
|
|
if (++retry >= 5) // kinda sus code. let's make sure we don't spin forever.
|
|
{
|
|
throw std::runtime_error("Timed out trying to fill the audio output buffer.");
|
|
}
|
|
int err = snd_pcm_prepare(playbackHandle);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalStateException(SS("Audio playback failed. " << snd_strerror(err)));
|
|
}
|
|
std::this_thread::sleep_for(std::chrono::milliseconds(100));
|
|
continue;
|
|
}
|
|
if (avail == 0)
|
|
break;
|
|
if (avail > this->bufferSize)
|
|
avail = this->bufferSize;
|
|
|
|
ssize_t err = WriteBuffer(playbackHandle, rawPlaybackBuffer.data(), avail);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalStateException(SS("Audio playback failed. " << snd_strerror(err)));
|
|
}
|
|
}
|
|
validate_capture_handle();
|
|
}
|
|
void recover_from_output_underrun(snd_pcm_t *capture_handle, snd_pcm_t *playback_handle, int err)
|
|
{
|
|
validate_capture_handle();
|
|
if (err == -EPIPE)
|
|
{
|
|
err = snd_pcm_prepare(playback_handle);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalStateException(SS("Can't recover from ALSA output underrun. (" << snd_strerror(err) << ")"));
|
|
}
|
|
FillOutputBuffer();
|
|
}
|
|
else
|
|
{
|
|
throw PiPedalStateException(SS("Can't recover from ALSA output error. (" << snd_strerror(err) << ")"));
|
|
}
|
|
validate_capture_handle();
|
|
}
|
|
void recover_from_input_underrun(snd_pcm_t *capture_handle, snd_pcm_t *playback_handle, int err)
|
|
{
|
|
validate_capture_handle();
|
|
|
|
if (err == -EPIPE)
|
|
{
|
|
|
|
// Unlink the streams before recovery
|
|
snd_pcm_unlink(capture_handle);
|
|
|
|
err = snd_pcm_drop(capture_handle);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalStateException(SS("Can't recover from ALSA underrun. (" << snd_strerror(err) << ")"));
|
|
}
|
|
err = snd_pcm_drop(playback_handle);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalStateException(SS("Can't recover from ALSA underrun. (" << snd_strerror(err) << ")"));
|
|
}
|
|
|
|
// Prepare both streams
|
|
if ((err = snd_pcm_prepare(playback_handle)) < 0)
|
|
{
|
|
throw std::runtime_error(SS("Cannot prepare playback stream: " << snd_strerror(err)));
|
|
}
|
|
if ((err = snd_pcm_prepare(capture_handle)) < 0)
|
|
{
|
|
throw std::runtime_error(SS("Cannot prepare capture stream: " << snd_strerror(err)));
|
|
}
|
|
|
|
// Fill the playback buffer with silence
|
|
FillOutputBuffer();
|
|
|
|
// Resynchronize the streams
|
|
if ((err = snd_pcm_link(capture_handle, playback_handle)) < 0)
|
|
{
|
|
throw std::runtime_error(SS("Cannot relink streams: " << snd_strerror(err)));
|
|
}
|
|
|
|
// Start the streams
|
|
if ((err = snd_pcm_start(capture_handle)) < 0)
|
|
{
|
|
throw std::runtime_error(SS("Cannot restart capture stream: " << snd_strerror(err)));
|
|
}
|
|
validate_capture_handle();
|
|
}
|
|
else if (err == ESTRPIPE)
|
|
{
|
|
audioRunning = false;
|
|
validate_capture_handle();
|
|
|
|
while ((err = snd_pcm_resume(capture_handle)) == -EAGAIN)
|
|
{
|
|
sleep(1);
|
|
}
|
|
if (err < 0)
|
|
{
|
|
err = snd_pcm_prepare(capture_handle);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalStateException(SS("Can't recover from ALSA suspend. (" << snd_strerror(err) << ")"));
|
|
}
|
|
}
|
|
audioRunning = true;
|
|
validate_capture_handle();
|
|
}
|
|
else
|
|
{
|
|
throw std::runtime_error(SS("Can't restart audio: " << snd_strerror(err)));
|
|
}
|
|
}
|
|
|
|
void DumpStatus(snd_pcm_t *handle)
|
|
{
|
|
#ifdef ALSADRIVER_CONFIG_DBG
|
|
snd_pcm_status(handle, snd_status);
|
|
snd_pcm_status_dump(snd_status, snd_output);
|
|
#endif
|
|
}
|
|
|
|
std::unique_ptr<std::jthread> audioThread;
|
|
bool audioRunning;
|
|
|
|
bool block = false;
|
|
|
|
snd_pcm_sframes_t ReadBuffer(snd_pcm_t *handle, uint8_t *buffer, snd_pcm_uframes_t frames)
|
|
{
|
|
// transcode to jack format.
|
|
// expand running status if neccessary.
|
|
// deal with regular and sysex messages split across
|
|
// buffer boundaries (but discard them)
|
|
snd_pcm_sframes_t framesRead;
|
|
|
|
auto state = snd_pcm_state(handle);
|
|
auto frame_bytes = this->captureFrameSize;
|
|
do
|
|
{
|
|
framesRead = snd_pcm_readi(handle, buffer, frames);
|
|
if (framesRead < 0)
|
|
{
|
|
return framesRead;
|
|
}
|
|
if (framesRead > 0)
|
|
{
|
|
buffer += framesRead * frame_bytes;
|
|
frames -= framesRead;
|
|
}
|
|
if (framesRead == 0)
|
|
{
|
|
snd_pcm_wait(captureHandle, 1);
|
|
}
|
|
} while (frames > 0);
|
|
return framesRead;
|
|
}
|
|
|
|
protected:
|
|
void ReadMidiData(uint32_t audioFrame)
|
|
{
|
|
AlsaMidiMessage message;
|
|
|
|
midiEventCount = 0;
|
|
while (alsaSequencer->ReadMessage(message, 0))
|
|
{
|
|
size_t messageSize = message.size;
|
|
if (messageSize == 0)
|
|
{
|
|
continue;
|
|
}
|
|
if (midiEventMemoryIndex + messageSize >= this->midiEventMemory.size())
|
|
{
|
|
continue;
|
|
}
|
|
if (midiEventCount >= this->midiEvents.size())
|
|
{
|
|
midiEvents.resize(midiEventCount * 2);
|
|
}
|
|
// for now, prevent META event messages from propagating.
|
|
if (message.data[0] == 0xFF && message.size > 1)
|
|
{
|
|
continue;
|
|
}
|
|
MidiEvent *pEvent = midiEvents.data() + midiEventCount++;
|
|
pEvent->time = audioFrame;
|
|
pEvent->size = messageSize;
|
|
pEvent->buffer = midiEventMemory.data() + midiEventMemoryIndex;
|
|
|
|
memcpy(
|
|
midiEventMemory.data() + midiEventMemoryIndex,
|
|
message.data,
|
|
message.size);
|
|
midiEventMemoryIndex += messageSize;
|
|
}
|
|
}
|
|
|
|
private:
|
|
long WriteBuffer(snd_pcm_t *handle, uint8_t *buf, size_t frames)
|
|
{
|
|
long framesRead;
|
|
auto frame_bytes = this->playbackFrameSize;
|
|
|
|
while (frames > 0)
|
|
{
|
|
framesRead = snd_pcm_writei(handle, buf, frames);
|
|
if (framesRead == -EAGAIN)
|
|
continue;
|
|
if (framesRead < 0)
|
|
return framesRead;
|
|
buf += framesRead * frame_bytes;
|
|
frames -= framesRead;
|
|
}
|
|
return 0;
|
|
}
|
|
void AudioThread()
|
|
{
|
|
SetThreadName("alsaDriver");
|
|
try
|
|
{
|
|
SetThreadPriority(SchedulerPriority::RealtimeAudio);
|
|
|
|
bool ok = true;
|
|
|
|
auto playbackState = snd_pcm_state(playbackHandle);
|
|
|
|
FillOutputBuffer();
|
|
|
|
int err;
|
|
if ((err = snd_pcm_start(captureHandle)) < 0)
|
|
{
|
|
throw PiPedalStateException("Unable to start ALSA capture.");
|
|
}
|
|
|
|
CrashGuardLock crashGuardLock;
|
|
|
|
cpuUse.SetStartTime(cpuUse.Now());
|
|
while (true)
|
|
{
|
|
validate_capture_handle();
|
|
cpuUse.UpdateCpuUse();
|
|
|
|
if (terminateAudio())
|
|
{
|
|
break;
|
|
}
|
|
this->midiEventCount = 0;
|
|
|
|
// snd_pcm_wait(captureHandle, 1);
|
|
ssize_t framesToRead = bufferSize;
|
|
ssize_t framesRead = 0;
|
|
bool xrun = false;
|
|
validate_capture_handle();
|
|
|
|
while (framesToRead != 0)
|
|
{
|
|
ReadMidiData((uint32_t)framesRead);
|
|
|
|
ssize_t thisTime = framesToRead;
|
|
ssize_t nFrames;
|
|
if ((nFrames = ReadBuffer(
|
|
captureHandle,
|
|
this->rawCaptureBuffer.data() + this->captureFrameSize * framesRead,
|
|
framesToRead)) < 0)
|
|
{
|
|
this->driverHost->OnUnderrun();
|
|
recover_from_input_underrun(captureHandle, playbackHandle, nFrames);
|
|
xrun = true;
|
|
break;
|
|
}
|
|
framesRead += nFrames;
|
|
framesToRead -= nFrames;
|
|
}
|
|
validate_capture_handle();
|
|
|
|
if (xrun)
|
|
{
|
|
continue;
|
|
}
|
|
cpuUse.AddSample(ProfileCategory::Read);
|
|
if (framesRead == 0)
|
|
continue;
|
|
if (framesRead != bufferSize)
|
|
{
|
|
throw PiPedalStateException("Invalid read.");
|
|
}
|
|
|
|
(this->*copyInputFn)(framesRead);
|
|
cpuUse.AddSample(ProfileCategory::Driver);
|
|
|
|
this->driverHost->OnProcess(framesRead);
|
|
|
|
cpuUse.AddSample(ProfileCategory::Execute);
|
|
|
|
(this->*copyOutputFn)(framesRead);
|
|
cpuUse.AddSample(ProfileCategory::Driver);
|
|
// process.
|
|
|
|
ssize_t err = WriteBuffer(playbackHandle, rawPlaybackBuffer.data(), framesRead);
|
|
|
|
if (err < 0)
|
|
{
|
|
this->driverHost->OnUnderrun();
|
|
recover_from_output_underrun(captureHandle, playbackHandle, err);
|
|
}
|
|
cpuUse.AddSample(ProfileCategory::Write);
|
|
}
|
|
}
|
|
catch (const std::exception &e)
|
|
{
|
|
Lv2Log::error(e.what());
|
|
Lv2Log::error("ALSA audio thread terminated abnormally.");
|
|
}
|
|
this->driverHost->OnAlsaDriverStopped();
|
|
|
|
// if we terminated abnormally, pump messages until we have been terminated.
|
|
if (!terminateAudio())
|
|
{
|
|
// zero out input buffers.
|
|
for (size_t i = 0; i < this->captureBuffers.size(); ++i)
|
|
{
|
|
float *pBuffer = captureBuffers[i];
|
|
for (size_t j = 0; j < this->bufferSize; ++j)
|
|
{
|
|
pBuffer[j] = 0;
|
|
}
|
|
}
|
|
try
|
|
{
|
|
while (!terminateAudio())
|
|
{
|
|
std::this_thread::sleep_for(std::chrono::milliseconds(10));
|
|
// zero out input buffers.
|
|
this->driverHost->OnProcess(this->bufferSize);
|
|
}
|
|
}
|
|
catch (const std::exception &e)
|
|
{
|
|
}
|
|
}
|
|
this->driverHost->OnAudioTerminated();
|
|
}
|
|
|
|
bool alsaActive = false;
|
|
|
|
static int IndexFromPortName(const std::string &s)
|
|
{
|
|
auto pos = s.find_last_of('_');
|
|
if (pos == std::string::npos)
|
|
{
|
|
throw std::invalid_argument("Bad port name.");
|
|
}
|
|
const char *p = s.c_str() + (pos + 1);
|
|
|
|
int v = atoi(p);
|
|
if (v < 0)
|
|
{
|
|
throw std::invalid_argument("Bad port name.");
|
|
}
|
|
return v;
|
|
}
|
|
|
|
bool activated = false;
|
|
virtual void Activate()
|
|
{
|
|
if (activated)
|
|
{
|
|
throw PiPedalStateException("Already activated.");
|
|
}
|
|
activated = true;
|
|
|
|
this->activeCaptureBuffers.resize(channelSelection.GetInputAudioPorts().size());
|
|
|
|
int ix = 0;
|
|
for (auto &x : channelSelection.GetInputAudioPorts())
|
|
{
|
|
int sourceIndex = IndexFromPortName(x);
|
|
if (sourceIndex >= captureBuffers.size())
|
|
{
|
|
Lv2Log::error(SS("Invalid audio input port: " << x));
|
|
}
|
|
else
|
|
{
|
|
this->activeCaptureBuffers[ix++] = this->captureBuffers[sourceIndex];
|
|
}
|
|
}
|
|
|
|
this->activePlaybackBuffers.resize(channelSelection.GetOutputAudioPorts().size());
|
|
|
|
ix = 0;
|
|
for (auto &x : channelSelection.GetOutputAudioPorts())
|
|
{
|
|
int sourceIndex = IndexFromPortName(x);
|
|
if (sourceIndex >= playbackBuffers.size())
|
|
{
|
|
Lv2Log::error(SS("Invalid audio output port: " << x));
|
|
}
|
|
else
|
|
{
|
|
this->activePlaybackBuffers[ix++] = this->playbackBuffers[sourceIndex];
|
|
}
|
|
}
|
|
|
|
audioThread = std::make_unique<std::jthread>([this]()
|
|
{ AudioThread(); });
|
|
}
|
|
|
|
virtual void Deactivate()
|
|
{
|
|
if (!activated)
|
|
{
|
|
return;
|
|
}
|
|
activated = false;
|
|
terminateAudio(true);
|
|
if (audioThread)
|
|
{
|
|
this->audioThread = 0; // jthread joins.
|
|
}
|
|
Lv2Log::debug("Audio thread joined.");
|
|
}
|
|
|
|
static constexpr size_t MIDI_MEMORY_BUFFER_SIZE = 32 * 1024;
|
|
static constexpr size_t MAX_MIDI_EVENT = 4 * 1024;
|
|
|
|
size_t midiEventCount = 0;
|
|
std::vector<MidiEvent> midiEvents;
|
|
size_t midiEventMemoryIndex = 0;
|
|
std::vector<uint8_t> midiEventMemory;
|
|
AlsaSequencer::ptr alsaSequencer;
|
|
|
|
public:
|
|
virtual void SetAlsaSequencer(AlsaSequencer::ptr alsaSequencer) override
|
|
{
|
|
this->alsaSequencer = alsaSequencer;
|
|
}
|
|
|
|
virtual size_t InputBufferCount() const { return activeCaptureBuffers.size(); }
|
|
virtual float *GetInputBuffer(size_t channel) override
|
|
{
|
|
return activeCaptureBuffers[channel];
|
|
}
|
|
|
|
virtual size_t GetMidiInputEventCount() override
|
|
{
|
|
return midiEventCount;
|
|
}
|
|
virtual MidiEvent *GetMidiEvents() override
|
|
{
|
|
return this->midiEvents.data();
|
|
}
|
|
|
|
virtual size_t OutputBufferCount() const { return activePlaybackBuffers.size(); }
|
|
virtual float *GetOutputBuffer(size_t channel) override
|
|
{
|
|
return activePlaybackBuffers[channel];
|
|
}
|
|
|
|
void FreeBuffers(std::vector<float *> &buffer)
|
|
{
|
|
for (size_t i = 0; i < buffer.size(); ++i)
|
|
{
|
|
delete[] buffer[i];
|
|
buffer[i] = 0;
|
|
}
|
|
buffer.clear();
|
|
}
|
|
void DeleteBuffers()
|
|
{
|
|
activeCaptureBuffers.clear();
|
|
activePlaybackBuffers.clear();
|
|
FreeBuffers(this->playbackBuffers);
|
|
FreeBuffers(this->captureBuffers);
|
|
}
|
|
virtual void Close()
|
|
{
|
|
std::atomic_thread_fence(std::memory_order::acquire);
|
|
|
|
if (!open)
|
|
{
|
|
return;
|
|
}
|
|
open = false;
|
|
Deactivate();
|
|
AlsaCleanup();
|
|
DeleteBuffers();
|
|
std::atomic_thread_fence(std::memory_order::release);
|
|
}
|
|
|
|
virtual float CpuUse()
|
|
{
|
|
return cpuUse.GetCpuUse();
|
|
}
|
|
|
|
virtual float CpuOverhead()
|
|
{
|
|
return cpuUse.GetCpuOverhead();
|
|
}
|
|
};
|
|
|
|
AudioDriver *CreateAlsaDriver(AudioDriverHost *driverHost)
|
|
{
|
|
return new AlsaDriverImpl(driverHost);
|
|
}
|
|
|
|
bool GetAlsaChannels(const JackServerSettings &jackServerSettings,
|
|
std::vector<std::string> &inputAudioPorts,
|
|
std::vector<std::string> &outputAudioPorts)
|
|
{
|
|
if (jackServerSettings.IsDummyAudioDevice())
|
|
{
|
|
auto nChannels = GetDummyAudioChannels(jackServerSettings.GetAlsaInputDevice());
|
|
|
|
inputAudioPorts.clear();
|
|
outputAudioPorts.clear();
|
|
for (uint32_t i = 0; i < nChannels; ++i)
|
|
{
|
|
inputAudioPorts.push_back(std::string(SS("system::capture_" << i)));
|
|
outputAudioPorts.push_back(std::string(SS("system::playback_" << i)));
|
|
}
|
|
return true;
|
|
}
|
|
|
|
snd_pcm_t *playbackHandle = nullptr;
|
|
snd_pcm_t *captureHandle = nullptr;
|
|
snd_pcm_hw_params_t *playbackHwParams = nullptr;
|
|
snd_pcm_hw_params_t *captureHwParams = nullptr;
|
|
|
|
Finally ff_playbackHandle{
|
|
[&playbackHandle]()
|
|
{
|
|
if (playbackHandle)
|
|
{
|
|
int rc = snd_pcm_close(playbackHandle);
|
|
if (rc < 0)
|
|
{
|
|
throw std::runtime_error("snd_pcm_close failed.");
|
|
}
|
|
playbackHandle = nullptr;
|
|
}
|
|
}};
|
|
Finally ff_captureHandle{
|
|
[&captureHandle]()
|
|
{
|
|
if (captureHandle)
|
|
{
|
|
int rc = snd_pcm_close(captureHandle);
|
|
if (rc < 0)
|
|
{
|
|
throw std::runtime_error("snd_pcm_close failed.");
|
|
}
|
|
|
|
captureHandle = nullptr;
|
|
}
|
|
}};
|
|
Finally ff_playbackHwParams{
|
|
[&playbackHwParams]()
|
|
{
|
|
if (playbackHwParams)
|
|
{
|
|
snd_pcm_hw_params_free(playbackHwParams);
|
|
}
|
|
}};
|
|
Finally ff_captureHwParams{
|
|
[&captureHwParams]()
|
|
{
|
|
if (captureHwParams)
|
|
{
|
|
snd_pcm_hw_params_free(captureHwParams);
|
|
}
|
|
}};
|
|
|
|
std::string inDev = jackServerSettings.GetAlsaInputDevice();
|
|
std::string outDev = jackServerSettings.GetAlsaOutputDevice();
|
|
bool result = false;
|
|
|
|
try
|
|
{
|
|
int err;
|
|
for (int retry = 0; retry < 2; ++retry)
|
|
{
|
|
err = snd_pcm_open(&playbackHandle, outDev.c_str(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
|
|
if (err < 0) // field report of a device that is present, but won't immediately open.
|
|
{
|
|
sleep(1);
|
|
continue;
|
|
}
|
|
break;
|
|
}
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalStateException(SS(outDev << " playback device not found. "
|
|
<< "(" << snd_strerror(err) << ")"));
|
|
}
|
|
|
|
for (int retry = 0; retry < 15; ++retry)
|
|
{
|
|
err = snd_pcm_open(&captureHandle, inDev.c_str(), SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
|
|
if (err == -EBUSY)
|
|
{
|
|
sleep(1);
|
|
continue;
|
|
}
|
|
break;
|
|
}
|
|
if (err < 0)
|
|
throw PiPedalStateException(SS(inDev << " capture device not found."));
|
|
|
|
if (snd_pcm_hw_params_malloc(&playbackHwParams) < 0)
|
|
{
|
|
throw PiPedalLogicException("Out of memory.");
|
|
}
|
|
if (snd_pcm_hw_params_malloc(&captureHwParams) < 0)
|
|
{
|
|
throw PiPedalLogicException("Out of memory.");
|
|
}
|
|
|
|
snd_pcm_hw_params_any(playbackHandle, playbackHwParams);
|
|
snd_pcm_hw_params_any(captureHandle, captureHwParams);
|
|
|
|
SetPreferredAlsaFormat(inDev, "capture", captureHandle, captureHwParams);
|
|
SetPreferredAlsaFormat(outDev, "output", playbackHandle, playbackHwParams);
|
|
|
|
unsigned int sampleRate = jackServerSettings.GetSampleRate();
|
|
err = snd_pcm_hw_params_set_rate_near(playbackHandle, playbackHwParams, &sampleRate, 0);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalLogicException("Sample rate not supported.");
|
|
}
|
|
sampleRate = jackServerSettings.GetSampleRate();
|
|
err = snd_pcm_hw_params_set_rate_near(captureHandle, captureHwParams, &sampleRate, 0);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalLogicException("Sample rate not supported.");
|
|
}
|
|
|
|
unsigned int playbackChannels, captureChannels;
|
|
|
|
err = snd_pcm_hw_params_get_channels_max(playbackHwParams, &playbackChannels);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalLogicException("No output channels.");
|
|
}
|
|
unsigned int channelsMin;
|
|
err = snd_pcm_hw_params_get_channels_min(playbackHwParams, &channelsMin);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalLogicException("No output channels.");
|
|
}
|
|
if (ShouldForceStereoChannels(playbackHandle, playbackHwParams, channelsMin, playbackChannels))
|
|
{
|
|
playbackChannels = 2;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_get_channels_max(captureHwParams, &captureChannels);
|
|
if (err < 0)
|
|
{
|
|
throw PiPedalLogicException("No input channels.");
|
|
}
|
|
err = snd_pcm_hw_params_get_channels_min(captureHwParams, &channelsMin);
|
|
if (err >= 0)
|
|
{
|
|
if (ShouldForceStereoChannels(captureHandle, captureHwParams, channelsMin, captureChannels))
|
|
{
|
|
captureChannels = 2;
|
|
}
|
|
}
|
|
|
|
inputAudioPorts.clear();
|
|
for (unsigned int i = 0; i < captureChannels; ++i)
|
|
{
|
|
inputAudioPorts.push_back(SS("system::capture_" << i));
|
|
}
|
|
|
|
outputAudioPorts.clear();
|
|
for (unsigned int i = 0; i < playbackChannels; ++i)
|
|
{
|
|
outputAudioPorts.push_back(SS("system::playback_" << i));
|
|
}
|
|
|
|
result = true;
|
|
}
|
|
catch (const std::exception &e)
|
|
{
|
|
result = false;
|
|
throw;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
static void AlsaAssert(bool value)
|
|
{
|
|
if (!value)
|
|
throw PiPedalStateException("Assert failed.");
|
|
}
|
|
|
|
#ifdef JUNK
|
|
static void ExpectEvent(AlsaDriverImpl::AlsaMidiDeviceImpl &m, int event, const std::vector<uint8_t> message)
|
|
{
|
|
MidiEvent e;
|
|
m.GetMidiInputEvent(&e, event);
|
|
AlsaAssert(e.size == message.size());
|
|
for (size_t i = 0; i < message.size(); ++i)
|
|
{
|
|
AlsaAssert(message[i] == e.buffer[i]);
|
|
}
|
|
}
|
|
#endif
|
|
|
|
void AlsaDriverImpl::TestFormatEncodeDecode(snd_pcm_format_t captureFormat)
|
|
{
|
|
this->alsa_device_name = "Test";
|
|
this->numberOfBuffers = 3;
|
|
this->bufferSize = 64;
|
|
this->user_threshold = this->bufferSize;
|
|
this->sampleRate = 44100;
|
|
this->captureChannels = 2;
|
|
this->playbackChannels = 2;
|
|
|
|
PrepareCaptureFunctions(captureFormat);
|
|
PreparePlaybackFunctions(captureFormat);
|
|
|
|
// make sure encode decode round-trips with reasonable accuracy.
|
|
|
|
for (size_t i = 0; i < bufferSize; ++i)
|
|
{
|
|
for (size_t c = 0; c < captureChannels; ++c)
|
|
{
|
|
// provide a rich set of approximately readable bits in the output.
|
|
float value = 1.0f * i / bufferSize + 1.0f * (i) / (128.0 * 256.0);
|
|
|
|
// only 16-bits of precision in data for 16-bit formats
|
|
if (captureFormat != snd_pcm_format_t::SND_PCM_FORMAT_S16_BE && captureFormat != snd_pcm_format_t::SND_PCM_FORMAT_S16_LE)
|
|
{
|
|
value += 1.0f * (c) / (128.0 * 256.0 * 256.0);
|
|
}
|
|
this->playbackBuffers[c][i] = value;
|
|
}
|
|
}
|
|
|
|
(this->*copyOutputFn)(bufferSize);
|
|
|
|
assert(captureFrameSize == playbackFrameSize);
|
|
memcpy(this->rawCaptureBuffer.data(), this->rawPlaybackBuffer.data(), captureFrameSize * bufferSize);
|
|
|
|
(this->*copyInputFn)(bufferSize);
|
|
|
|
for (size_t i = 0; i < bufferSize; ++i)
|
|
{
|
|
for (size_t c = 0; c < captureChannels; ++c)
|
|
{
|
|
float error =
|
|
this->captureBuffers[c][i] - this->playbackBuffers[c][i];
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|
|
|
assert(std::abs(error) < 4e-5);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AlsaFormatEncodeDecodeTest(AudioDriverHost *testDriverHost)
|
|
{
|
|
static snd_pcm_format_t formats[] = {
|
|
snd_pcm_format_t::SND_PCM_FORMAT_S16_LE,
|
|
snd_pcm_format_t::SND_PCM_FORMAT_S16_BE,
|
|
snd_pcm_format_t::SND_PCM_FORMAT_S32_LE,
|
|
snd_pcm_format_t::SND_PCM_FORMAT_S32_BE,
|
|
snd_pcm_format_t::SND_PCM_FORMAT_S24_3BE,
|
|
snd_pcm_format_t::SND_PCM_FORMAT_S24_3LE,
|
|
snd_pcm_format_t::SND_PCM_FORMAT_FLOAT_BE,
|
|
snd_pcm_format_t::SND_PCM_FORMAT_FLOAT_LE,
|
|
};
|
|
|
|
for (auto format : formats)
|
|
{
|
|
// Check audio encode/decode.
|
|
std::unique_ptr<AlsaDriverImpl> alsaDriver{
|
|
(AlsaDriverImpl *)new AlsaDriverImpl(testDriverHost)};
|
|
|
|
alsaDriver->TestFormatEncodeDecode(format);
|
|
}
|
|
}
|
|
void MidiDecoderTest()
|
|
{
|
|
#ifdef JUNK
|
|
AlsaDriverImpl::AlsaMidiDeviceImpl midiState;
|
|
|
|
MidiEvent event;
|
|
|
|
// Running status decoding.
|
|
{
|
|
static uint8_t m0[] = {0x80, 0x1, 0x2, 0x3, 0x4, 0x5};
|
|
midiState.NextEventBuffer();
|
|
midiState.ProcessInputBuffer(m0, sizeof(m0));
|
|
AlsaAssert(midiState.GetMidiInputEventCount() == 2);
|
|
AlsaAssert(midiState.GetMidiInputEvent(&event, 0));
|
|
|
|
ExpectEvent(midiState, 0, {0x80, 0x1, 0x2});
|
|
ExpectEvent(midiState, 1, {0x80, 0x3, 0x4});
|
|
|
|
static uint8_t m1[] = {0x06, 0xC0, 0x1, 0x2};
|
|
midiState.NextEventBuffer();
|
|
midiState.ProcessInputBuffer(m1, sizeof(m1));
|
|
AlsaAssert(midiState.GetMidiInputEventCount() == 3);
|
|
ExpectEvent(midiState, 0, {0x80, 0x05, 0x06});
|
|
ExpectEvent(midiState, 1, {0xC0, 0x1});
|
|
ExpectEvent(midiState, 2, {0xC0, 0x2});
|
|
}
|
|
|
|
// SYSEX.
|
|
{
|
|
static uint8_t m0[] = {0xF0, 0x76, 0xF7, 0xA};
|
|
midiState.NextEventBuffer();
|
|
midiState.ProcessInputBuffer(m0, 4);
|
|
AlsaAssert(midiState.GetMidiInputEventCount() == 2);
|
|
AlsaAssert(midiState.GetMidiInputEvent(&event, 0));
|
|
AlsaAssert(event.size == 2);
|
|
AlsaAssert(event.buffer[0] == 0xF0);
|
|
AlsaAssert(event.buffer[1] == 0x76);
|
|
}
|
|
|
|
// SPLIT SYSEX
|
|
{
|
|
static uint8_t m0[] = {0xF0, 0x76, 0x3B};
|
|
midiState.NextEventBuffer();
|
|
midiState.ProcessInputBuffer(m0, sizeof(m0));
|
|
AlsaAssert(midiState.GetMidiInputEventCount() == 0);
|
|
static uint8_t m1[] = {0x77, 0xF7};
|
|
midiState.NextEventBuffer();
|
|
midiState.ProcessInputBuffer(m1, sizeof(m1));
|
|
AlsaAssert(midiState.GetMidiInputEventCount() == 2);
|
|
|
|
AlsaAssert(midiState.GetMidiInputEvent(&event, 0));
|
|
AlsaAssert(event.size == 0x4);
|
|
AlsaAssert(event.buffer[0] == 0xF0);
|
|
AlsaAssert(event.buffer[1] == 0x76);
|
|
AlsaAssert(event.buffer[2] == 0x3B);
|
|
AlsaAssert(event.buffer[3] == 0x77);
|
|
}
|
|
#endif
|
|
}
|
|
|
|
void FreeAlsaGlobals()
|
|
{
|
|
snd_config_update_free_global(); // to get a clean Valgrind report.
|
|
}
|
|
} // namespace
|