decoder/*: move to decoder/plugins/
This commit is contained in:
Max Kellermann
260
src/decoder/plugins/SndfileDecoderPlugin.cxx
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260
src/decoder/plugins/SndfileDecoderPlugin.cxx
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/*
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* Copyright (C) 2003-2014 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include "SndfileDecoderPlugin.hxx"
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#include "../DecoderAPI.hxx"
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#include "InputStream.hxx"
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#include "CheckAudioFormat.hxx"
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#include "tag/TagHandler.hxx"
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#include "util/Error.hxx"
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#include "util/Domain.hxx"
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#include "Log.hxx"
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#include <sndfile.h>
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static constexpr Domain sndfile_domain("sndfile");
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static sf_count_t
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sndfile_vio_get_filelen(void *user_data)
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{
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const InputStream &is = *(const InputStream *)user_data;
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return is.GetSize();
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}
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static sf_count_t
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sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
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{
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InputStream &is = *(InputStream *)user_data;
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if (!is.LockSeek(offset, whence, IgnoreError()))
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return -1;
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return is.GetOffset();
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}
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static sf_count_t
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sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
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{
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InputStream &is = *(InputStream *)user_data;
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Error error;
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size_t nbytes = is.LockRead(ptr, count, error);
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if (nbytes == 0 && error.IsDefined()) {
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LogError(error);
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return -1;
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}
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return nbytes;
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}
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static sf_count_t
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sndfile_vio_write(gcc_unused const void *ptr,
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gcc_unused sf_count_t count,
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gcc_unused void *user_data)
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{
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/* no writing! */
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return -1;
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}
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static sf_count_t
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sndfile_vio_tell(void *user_data)
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{
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const InputStream &is = *(const InputStream *)user_data;
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return is.GetOffset();
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}
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/**
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* This SF_VIRTUAL_IO implementation wraps MPD's #input_stream to a
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* libsndfile stream.
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*/
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static SF_VIRTUAL_IO vio = {
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sndfile_vio_get_filelen,
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sndfile_vio_seek,
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sndfile_vio_read,
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sndfile_vio_write,
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sndfile_vio_tell,
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};
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/**
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* Converts a frame number to a timestamp (in seconds).
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*/
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static float
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frame_to_time(sf_count_t frame, const AudioFormat *audio_format)
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{
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return (float)frame / (float)audio_format->sample_rate;
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}
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/**
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* Converts a timestamp (in seconds) to a frame number.
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*/
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static sf_count_t
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time_to_frame(float t, const AudioFormat *audio_format)
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{
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return (sf_count_t)(t * audio_format->sample_rate);
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}
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static void
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sndfile_stream_decode(Decoder &decoder, InputStream &is)
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{
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SNDFILE *sf;
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SF_INFO info;
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size_t frame_size;
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sf_count_t read_frames, num_frames;
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int buffer[4096];
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info.format = 0;
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sf = sf_open_virtual(&vio, SFM_READ, &info, &is);
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if (sf == nullptr) {
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LogWarning(sndfile_domain, "sf_open_virtual() failed");
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return;
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}
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/* for now, always read 32 bit samples. Later, we could lower
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MPD's CPU usage by reading 16 bit samples with
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sf_readf_short() on low-quality source files. */
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Error error;
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AudioFormat audio_format;
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if (!audio_format_init_checked(audio_format, info.samplerate,
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SampleFormat::S32,
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info.channels, error)) {
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LogError(error);
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return;
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}
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decoder_initialized(decoder, audio_format, info.seekable,
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frame_to_time(info.frames, &audio_format));
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frame_size = audio_format.GetFrameSize();
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read_frames = sizeof(buffer) / frame_size;
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DecoderCommand cmd;
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do {
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num_frames = sf_readf_int(sf, buffer, read_frames);
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if (num_frames <= 0)
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break;
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cmd = decoder_data(decoder, is,
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buffer, num_frames * frame_size,
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0);
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if (cmd == DecoderCommand::SEEK) {
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sf_count_t c =
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time_to_frame(decoder_seek_where(decoder),
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&audio_format);
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c = sf_seek(sf, c, SEEK_SET);
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if (c < 0)
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decoder_seek_error(decoder);
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else
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decoder_command_finished(decoder);
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cmd = DecoderCommand::NONE;
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}
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} while (cmd == DecoderCommand::NONE);
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sf_close(sf);
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}
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static bool
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sndfile_scan_file(const char *path_fs,
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const struct tag_handler *handler, void *handler_ctx)
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{
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SNDFILE *sf;
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SF_INFO info;
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const char *p;
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info.format = 0;
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sf = sf_open(path_fs, SFM_READ, &info);
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if (sf == nullptr)
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return false;
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if (!audio_valid_sample_rate(info.samplerate)) {
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sf_close(sf);
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FormatWarning(sndfile_domain,
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"Invalid sample rate in %s", path_fs);
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return false;
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}
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tag_handler_invoke_duration(handler, handler_ctx,
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info.frames / info.samplerate);
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p = sf_get_string(sf, SF_STR_TITLE);
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if (p != nullptr)
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tag_handler_invoke_tag(handler, handler_ctx,
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TAG_TITLE, p);
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p = sf_get_string(sf, SF_STR_ARTIST);
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if (p != nullptr)
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tag_handler_invoke_tag(handler, handler_ctx,
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TAG_ARTIST, p);
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p = sf_get_string(sf, SF_STR_DATE);
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if (p != nullptr)
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tag_handler_invoke_tag(handler, handler_ctx,
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TAG_DATE, p);
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sf_close(sf);
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return true;
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}
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static const char *const sndfile_suffixes[] = {
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"wav", "aiff", "aif", /* Microsoft / SGI / Apple */
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"au", "snd", /* Sun / DEC / NeXT */
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"paf", /* Paris Audio File */
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"iff", "svx", /* Commodore Amiga IFF / SVX */
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"sf", /* IRCAM */
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"voc", /* Creative */
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"w64", /* Soundforge */
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"pvf", /* Portable Voice Format */
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"xi", /* Fasttracker */
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"htk", /* HMM Tool Kit */
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"caf", /* Apple */
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"sd2", /* Sound Designer II */
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/* libsndfile also supports FLAC and Ogg Vorbis, but only by
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linking with libFLAC and libvorbis - we can do better, we
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have native plugins for these libraries */
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nullptr
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};
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static const char *const sndfile_mime_types[] = {
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"audio/x-wav",
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"audio/x-aiff",
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/* what are the MIME types of the other supported formats? */
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nullptr
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};
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const struct DecoderPlugin sndfile_decoder_plugin = {
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"sndfile",
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nullptr,
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nullptr,
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sndfile_stream_decode,
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nullptr,
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sndfile_scan_file,
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nullptr,
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nullptr,
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sndfile_suffixes,
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sndfile_mime_types,
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};
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