From f8bfea8bae44134e8002db869fd4ec137d1c0d6e Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Fri, 24 Jan 2014 16:18:21 +0100 Subject: Input*: move to input/ --- src/input/plugins/AlsaInputPlugin.cxx | 431 ++++++++++++++++++++++++++++++++++ 1 file changed, 431 insertions(+) create mode 100644 src/input/plugins/AlsaInputPlugin.cxx (limited to 'src/input/plugins/AlsaInputPlugin.cxx') diff --git a/src/input/plugins/AlsaInputPlugin.cxx b/src/input/plugins/AlsaInputPlugin.cxx new file mode 100644 index 000000000..da2d9a926 --- /dev/null +++ b/src/input/plugins/AlsaInputPlugin.cxx @@ -0,0 +1,431 @@ +/* + * Copyright (C) 2003-2014 The Music Player Daemon Project + * http://www.musicpd.org + * + * 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. + */ + +/* + * ALSA code based on an example by Paul Davis released under GPL here: + * http://equalarea.com/paul/alsa-audio.html + * and one by Matthias Nagorni, also GPL, here: + * http://alsamodular.sourceforge.net/alsa_programming_howto.html + */ + +#include "config.h" +#include "AlsaInputPlugin.hxx" +#include "../InputPlugin.hxx" +#include "../InputStream.hxx" +#include "util/Domain.hxx" +#include "util/Error.hxx" +#include "util/StringUtil.hxx" +#include "util/ReusableArray.hxx" +#include "util/Cast.hxx" +#include "Log.hxx" +#include "event/MultiSocketMonitor.hxx" +#include "event/DeferredMonitor.hxx" +#include "event/Call.hxx" +#include "thread/Mutex.hxx" +#include "thread/Cond.hxx" +#include "IOThread.hxx" + +#include + +#include +#include + +static constexpr Domain alsa_input_domain("alsa"); + +static constexpr const char *default_device = "hw:0,0"; + +// the following defaults are because the PcmDecoderPlugin forces CD format +static constexpr snd_pcm_format_t default_format = SND_PCM_FORMAT_S16; +static constexpr int default_channels = 2; // stereo +static constexpr unsigned int default_rate = 44100; // cd quality + +/** + * This value should be the same as the read buffer size defined in + * PcmDecoderPlugin.cxx:pcm_stream_decode(). + * We use it to calculate how many audio frames to buffer in the alsa driver + * before reading from the device. snd_pcm_readi() blocks until that many + * frames are ready. + */ +static constexpr size_t read_buffer_size = 4096; + +class AlsaInputStream final : MultiSocketMonitor, DeferredMonitor { + InputStream base; + snd_pcm_t *capture_handle; + size_t frame_size; + int frames_to_read; + bool eof; + + /** + * Is somebody waiting for data? This is set by method + * Available(). + */ + std::atomic_bool waiting; + + ReusableArray pfd_buffer; + +public: + AlsaInputStream(EventLoop &loop, + const char *uri, Mutex &mutex, Cond &cond, + snd_pcm_t *_handle, int _frame_size) + :MultiSocketMonitor(loop), + DeferredMonitor(loop), + base(input_plugin_alsa, uri, mutex, cond), + capture_handle(_handle), + frame_size(_frame_size), + eof(false) + { + assert(uri != nullptr); + assert(_handle != nullptr); + + /* this mime type forces use of the PcmDecoderPlugin. + Needs to be generalised when/if that decoder is + updated to support other audio formats */ + base.mime = "audio/x-mpd-cdda-pcm"; + base.seekable = false; + base.size = -1; + base.ready = true; + frames_to_read = read_buffer_size / frame_size; + + snd_pcm_start(capture_handle); + + DeferredMonitor::Schedule(); + } + + ~AlsaInputStream() { + snd_pcm_close(capture_handle); + } + + using DeferredMonitor::GetEventLoop; + + static InputStream *Create(const char *uri, Mutex &mutex, Cond &cond, + Error &error); + +#if GCC_CHECK_VERSION(4,6) || defined(__clang__) +#pragma GCC diagnostic push +#pragma GCC diagnostic ignored "-Winvalid-offsetof" +#endif + + static constexpr AlsaInputStream *Cast(InputStream *is) { + return ContainerCast(is, AlsaInputStream, base); + } + +#if GCC_CHECK_VERSION(4,6) || defined(__clang__) +#pragma GCC diagnostic pop +#endif + + bool Available() { + if (snd_pcm_avail(capture_handle) > frames_to_read) + return true; + + if (!waiting.exchange(true)) + SafeInvalidateSockets(); + + return false; + } + + size_t Read(void *ptr, size_t size, Error &error); + + bool IsEOF() { + return eof; + } + +private: + static snd_pcm_t *OpenDevice(const char *device, int rate, + snd_pcm_format_t format, int channels, + Error &error); + + int Recover(int err); + + void SafeInvalidateSockets() { + DeferredMonitor::Schedule(); + } + + virtual void RunDeferred() override { + InvalidateSockets(); + } + + virtual int PrepareSockets() override; + virtual void DispatchSockets() override; +}; + +inline InputStream * +AlsaInputStream::Create(const char *uri, Mutex &mutex, Cond &cond, + Error &error) +{ + const char *const scheme = "alsa://"; + if (!StringStartsWith(uri, scheme)) + return nullptr; + + const char *device = uri + strlen(scheme); + if (strlen(device) == 0) + device = default_device; + + /* placeholders - eventually user-requested audio format will + be passed via the URI. For now we just force the + defaults */ + int rate = default_rate; + snd_pcm_format_t format = default_format; + int channels = default_channels; + + snd_pcm_t *handle = OpenDevice(device, rate, format, channels, + error); + if (handle == nullptr) + return nullptr; + + int frame_size = snd_pcm_format_width(format) / 8 * channels; + AlsaInputStream *stream = new AlsaInputStream(io_thread_get(), + uri, mutex, cond, + handle, frame_size); + return &stream->base; +} + +inline size_t +AlsaInputStream::Read(void *ptr, size_t size, Error &error) +{ + assert(ptr != nullptr); + + int num_frames = size / frame_size; + int ret; + while ((ret = snd_pcm_readi(capture_handle, ptr, num_frames)) < 0) { + if (Recover(ret) < 0) { + eof = true; + error.Format(alsa_input_domain, + "PCM error - stream aborted"); + return 0; + } + } + + size_t nbytes = ret * frame_size; + base.offset += nbytes; + return nbytes; +} + +int +AlsaInputStream::PrepareSockets() +{ + if (!waiting) { + ClearSocketList(); + return -1; + } + + int count = snd_pcm_poll_descriptors_count(capture_handle); + if (count < 0) { + ClearSocketList(); + return -1; + } + + struct pollfd *pfds = pfd_buffer.Get(count); + + count = snd_pcm_poll_descriptors(capture_handle, pfds, count); + if (count < 0) + count = 0; + + ReplaceSocketList(pfds, count); + return -1; +} + +void +AlsaInputStream::DispatchSockets() +{ + waiting = false; + + const ScopeLock protect(base.mutex); + /* wake up the thread that is waiting for more data */ + base.cond.broadcast(); +} + +inline int +AlsaInputStream::Recover(int err) +{ + switch(err) { + case -EPIPE: + LogDebug(alsa_input_domain, "Buffer Overrun"); + // drop through + case -ESTRPIPE: + case -EINTR: + err = snd_pcm_recover(capture_handle, err, 1); + break; + default: + // something broken somewhere, give up + err = -1; + } + return err; +} + +inline snd_pcm_t * +AlsaInputStream::OpenDevice(const char *device, + int rate, snd_pcm_format_t format, int channels, + Error &error) +{ + snd_pcm_t *capture_handle; + int err; + if ((err = snd_pcm_open(&capture_handle, device, + SND_PCM_STREAM_CAPTURE, 0)) < 0) { + error.Format(alsa_input_domain, "Failed to open device: %s (%s)", device, snd_strerror(err)); + return nullptr; + } + + snd_pcm_hw_params_t *hw_params; + if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) { + error.Format(alsa_input_domain, "Cannot allocate hardware parameter structure (%s)", snd_strerror(err)); + snd_pcm_close(capture_handle); + return nullptr; + } + + if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) { + error.Format(alsa_input_domain, "Cannot initialize hardware parameter structure (%s)", snd_strerror(err)); + snd_pcm_hw_params_free(hw_params); + snd_pcm_close(capture_handle); + return nullptr; + } + + if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { + error.Format(alsa_input_domain, "Cannot set access type (%s)", snd_strerror (err)); + snd_pcm_hw_params_free(hw_params); + snd_pcm_close(capture_handle); + return nullptr; + } + + if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params, format)) < 0) { + snd_pcm_hw_params_free(hw_params); + snd_pcm_close(capture_handle); + error.Format(alsa_input_domain, "Cannot set sample format (%s)", snd_strerror (err)); + return nullptr; + } + + if ((err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, channels)) < 0) { + snd_pcm_hw_params_free(hw_params); + snd_pcm_close(capture_handle); + error.Format(alsa_input_domain, "Cannot set channels (%s)", snd_strerror (err)); + return nullptr; + } + + if ((err = snd_pcm_hw_params_set_rate(capture_handle, hw_params, rate, 0)) < 0) { + snd_pcm_hw_params_free(hw_params); + snd_pcm_close(capture_handle); + error.Format(alsa_input_domain, "Cannot set sample rate (%s)", snd_strerror (err)); + return nullptr; + } + + /* period needs to be big enough so that poll() doesn't fire too often, + * but small enough that buffer overruns don't occur if Read() is not + * invoked often enough. + * the calculation here is empirical; however all measurements were + * done using 44100:16:2. When we extend this plugin to support + * other audio formats then this may need to be revisited */ + snd_pcm_uframes_t period = read_buffer_size * 2; + int direction = -1; + if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params, + &period, &direction)) < 0) { + error.Format(alsa_input_domain, "Cannot set period size (%s)", + snd_strerror(err)); + snd_pcm_hw_params_free(hw_params); + snd_pcm_close(capture_handle); + return nullptr; + } + + if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) { + error.Format(alsa_input_domain, "Cannot set parameters (%s)", + snd_strerror(err)); + snd_pcm_hw_params_free(hw_params); + snd_pcm_close(capture_handle); + return nullptr; + } + + snd_pcm_hw_params_free (hw_params); + + snd_pcm_sw_params_t *sw_params; + + snd_pcm_sw_params_malloc(&sw_params); + snd_pcm_sw_params_current(capture_handle, sw_params); + + if ((err = snd_pcm_sw_params_set_start_threshold(capture_handle, sw_params, + period)) < 0) { + error.Format(alsa_input_domain, + "unable to set start threshold (%s)", snd_strerror(err)); + snd_pcm_sw_params_free(sw_params); + snd_pcm_close(capture_handle); + return nullptr; + } + + if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0) { + error.Format(alsa_input_domain, + "unable to install sw params (%s)", snd_strerror(err)); + snd_pcm_sw_params_free(sw_params); + snd_pcm_close(capture_handle); + return nullptr; + } + + snd_pcm_sw_params_free(sw_params); + + snd_pcm_prepare(capture_handle); + + return capture_handle; +} + +/*######################### Plugin Functions ##############################*/ + +static InputStream * +alsa_input_open(const char *uri, Mutex &mutex, Cond &cond, Error &error) +{ + return AlsaInputStream::Create(uri, mutex, cond, error); +} + +static void +alsa_input_close(InputStream *is) +{ + AlsaInputStream *ais = AlsaInputStream::Cast(is); + delete ais; +} + +static bool +alsa_input_available(InputStream *is) +{ + AlsaInputStream *ais = AlsaInputStream::Cast(is); + return ais->Available(); +} + +static size_t +alsa_input_read(InputStream *is, void *ptr, size_t size, Error &error) +{ + AlsaInputStream *ais = AlsaInputStream::Cast(is); + return ais->Read(ptr, size, error); +} + +static bool +alsa_input_eof(gcc_unused InputStream *is) +{ + AlsaInputStream *ais = AlsaInputStream::Cast(is); + return ais->IsEOF(); +} + +const struct InputPlugin input_plugin_alsa = { + "alsa", + nullptr, + nullptr, + alsa_input_open, + alsa_input_close, + nullptr, + nullptr, + nullptr, + alsa_input_available, + alsa_input_read, + alsa_input_eof, + nullptr, +}; -- cgit v1.2.3