view src/flacng/plugin.c @ 2284:d19b53359b24

cleaned up the sndfile wav plugin, currently limiting it ONLY TO WAV PLAYBACK. if somebody is more experienced with it and wants to restore the other formats, go ahead (maybe change the name of the plugin too?).

/*
 *  A FLAC decoder plugin for the Audacious Media Player
 *  Copyright (C) 2005 Ralf Ertzinger
 *
 *  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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
 */

#include "flacng.h"
#include <audacious/util.h>
#include <audacious/output.h>
#include <audacious/i18n.h>
#include "tools.h"
#include "plugin.h"
#include "seekable_stream_callbacks.h"
#include "version.h"
#include "debug.h"

static gchar *flac_fmts[] = { "flac", "fla", NULL };

InputPlugin flac_ip = {
    .description = "FLACng Audio Plugin",
    .init = flac_init,
    .cleanup = flac_cleanup,
    .about = flac_aboutbox,
    .is_our_file = flac_is_our_file,
    .play_file = flac_play_file,
    .stop = flac_stop,
    .pause = flac_pause,
    .seek = flac_seek,
    .get_song_info = flac_get_song_info,
    .get_song_tuple = flac_get_song_tuple,	// get a tuple
    .is_our_file_from_vfs = flac_is_our_fd,	// version of is_our_file which is handed an FD
    .vfs_extensions = flac_fmts			// vector of fileextensions allowed by the plugin
};

InputPlugin *flac_iplist[] = { &flac_ip, NULL };

DECLARE_PLUGIN(flacng, NULL, NULL, flac_iplist, NULL, NULL, NULL, NULL, NULL);

FLAC__StreamDecoder* test_decoder;
FLAC__StreamDecoder* main_decoder;
callback_info* test_info;
callback_info* main_info;
gboolean plugin_initialized = FALSE;
gint seek_to = -1;
static GThread* thread = NULL;

/* === */

void flac_init(void) {

    FLAC__StreamDecoderInitStatus ret;

    _ENTER;

    /*
     * Callback structure and decoder for file test
     * purposes
     */
    if (NULL == (test_info = init_callback_info("test"))) {
        _ERROR("Could not initialize the test callback structure!");
        _LEAVE;
    }
    _DEBUG("Test callback structure at %p", test_info);

    if (NULL == (test_decoder = FLAC__stream_decoder_new())) {
        _ERROR("Could not create the test FLAC decoder instance!");
        _LEAVE;
    }


    /*
     * We want the VORBISCOMMENT metadata, for the file tags
     * and SEEKTABLE
     */
    FLAC__stream_decoder_set_metadata_respond(test_decoder, FLAC__METADATA_TYPE_VORBIS_COMMENT);
    FLAC__stream_decoder_set_metadata_respond(test_decoder, FLAC__METADATA_TYPE_SEEKTABLE);

    /*
     * Callback structure and decoder for main decoding
     * loop
     */
    if (NULL == (main_info = init_callback_info("main"))) {
        _ERROR("Could not initialize the main callback structure!");
        _LEAVE;
    }
    _DEBUG("Main callback structure at %p", main_info);

    if (NULL == (main_decoder = FLAC__stream_decoder_new())) {
        _ERROR("Could not create the main FLAC decoder instance!");
        _LEAVE;
    }


    /*
     * We want the VORBISCOMMENT metadata, for the file tags
     * and SEEKTABLE
     */
    FLAC__stream_decoder_set_metadata_respond(main_decoder, FLAC__METADATA_TYPE_VORBIS_COMMENT);
    FLAC__stream_decoder_set_metadata_respond(main_decoder, FLAC__METADATA_TYPE_SEEKTABLE);

    /*
     * Initialize decoders
     */
    if (FLAC__STREAM_DECODER_INIT_STATUS_OK != (ret = FLAC__stream_decoder_init_stream(
            test_decoder,
            read_callback,
            seek_callback,
            tell_callback,
            length_callback,
            eof_callback,
            write_callback,
            metadata_callback,
            error_callback,
            test_info))) {
        _ERROR("Could not initialize test FLAC decoder: %s(%d)",
                FLAC__StreamDecoderInitStatusString[ret], ret);
        _LEAVE;
     }

    if (FLAC__STREAM_DECODER_INIT_STATUS_OK != (ret = FLAC__stream_decoder_init_stream(
            main_decoder,
            read_callback,
            seek_callback,
            tell_callback,
            length_callback,
            eof_callback,
            write_callback,
            metadata_callback,
            error_callback,
            main_info))) {
        _ERROR("Could not initialize main FLAC decoder: %s(%d)",
                FLAC__StreamDecoderInitStatusString[ret], ret);
        _LEAVE;
     }

     _DEBUG("plugin initialized OK!");
     plugin_initialized = TRUE;
    _LEAVE;
}

/* --- */

void flac_cleanup(void)
{
    _ENTER;

    FLAC__stream_decoder_delete(main_decoder);
    clean_callback_info(main_info);

    FLAC__stream_decoder_delete(test_decoder);
    clean_callback_info(test_info);

    plugin_initialized = FALSE;

    _LEAVE;
}

/* --- */

gboolean flac_is_our_fd(gchar* filename, VFSFile* fd) {

    _ENTER;

    if (!plugin_initialized) {
        _ERROR("Plugin not initialized!");
        _LEAVE FALSE;
    }

    _DEBUG("Testing fd for file: %s", filename);

    INFO_LOCK(test_info);

    if (FALSE == read_metadata(fd, test_decoder, test_info)) {
        _DEBUG("File not handled by this plugin!");
        INFO_UNLOCK(test_info);
        _LEAVE FALSE;
    }

    /*
     * See if the metadata has changed
     */
    if (FALSE == test_info->metadata_changed) {
        _DEBUG("No metadata found in stream");
        INFO_UNLOCK(test_info);
        _LEAVE FALSE;
    }

    /*
     * If we get here, the file is supported by FLAC.
     * The stream characteristics have been filled in by
     * the metadata callback.
     */

    _DEBUG("Stream encoded at %d Hz, %d bps, %d channels",
        test_info->stream.samplerate,
        test_info->stream.bits_per_sample,
        test_info->stream.channels);

    if (MAX_SUPPORTED_CHANNELS < test_info->stream.channels) {
        _ERROR("This number of channels (%d) is currently not supported, stream not handled by this plugin",
            test_info->stream.channels);
        INFO_UNLOCK(test_info);
        _LEAVE FALSE;
    }

    if ((16 != test_info->stream.bits_per_sample) &&
        (24 != test_info->stream.bits_per_sample) &&
        (8 != test_info->stream.bits_per_sample)) {
        _ERROR("This number of bits (%d) is currently not supported, stream not handled by this plugin",
            test_info->stream.bits_per_sample);
        INFO_UNLOCK(test_info);
        _LEAVE FALSE;
    }

    /*
     * Looks good.
     */

    _DEBUG("Accepting file %s", filename);

    reset_info(test_info, FALSE);
    INFO_UNLOCK(test_info);

    _LEAVE TRUE;
}

/* --- */

gboolean flac_is_our_file(gchar* filename) {

    VFSFile* fd;
    gboolean ret;

    _ENTER;

    _DEBUG("Testing file: %s", filename);
    /*
     * Open the file
     */
    if (NULL == (fd = aud_vfs_fopen(filename, "rb"))) {
        _ERROR("Could not open file for reading! (%s)", filename);
        _LEAVE FALSE;
    }

    ret = flac_is_our_fd(filename, fd);

    aud_vfs_fclose(fd);

    _LEAVE ret;
}

/* --- */

void squeeze_audio(gint32* src, void* dst, guint count, guint src_res, guint dst_res) {

    /*
     * Changes the sample width of count samples in src from
     * src_res to dst_res and places the result in dst
     */

    gint i;
    gint32* rp = src;
    gint8* wp = dst;
    gint16* wp2 = dst;
    gint32* wp4 = dst;

    _ENTER;

    _DEBUG("Converting %d samples %d->%d", count, src_res, dst_res);

    if ((src_res % 8 != 0) || (dst_res % 8 != 0)) {
        _ERROR("Can not convert from %d bps to %d bps: not a multiple of 8",
                src_res, dst_res);
        _LEAVE;
    }

    if (16 == dst_res) {
        if (8 == src_res) {
            for (i=0; i<count; i++, wp2++, rp++) {
                *wp2 = (*rp << 8) & 0xffff;
            }
        } else if (16 == src_res) {
            for (i=0; i<count; i++, wp2++, rp++) {
                *wp2 = *rp & 0xffff;
            }
        } else if (24 == src_res) {
            for (i=0; i<count; i++, wp2++, rp++) {
                *wp2 = (*rp >> 8) & 0xffff;
            }
        } else if (32 == src_res) {
            for (i=0; i<count; i++, wp2++, rp++) {
                *wp2 = (*rp >> 16) & 0xffff;
            }
        }
    } else if (8 == dst_res) {
        if (8 == src_res) {
            for (i=0; i<count; i++, wp++, rp++) {
                *wp = *rp & 0xff;
            }
        } else if (16 == src_res) {
            for (i=0; i<count; i++, wp++, rp++) {
                *wp = (*rp >> 8) & 0xff;
            }
        } else if (24 == src_res) {
            for (i=0; i<count; i++, wp++, rp++) {
                *wp = (*rp >> 16) & 0xff;
            }
        } else if (32 == src_res) {
            for (i=0; i<count; i++, wp++, rp++) {
                *wp = (*rp >> 24) & 0xff;
            }
        }
    } else if (32 == dst_res) {
        if (8 == src_res) {
            for (i=0; i<count; i++, wp4++, rp++) {
                *wp4 = (*rp << 24) & 0xffffffff;
            }
        } else if (16 == src_res) {
            for (i=0; i<count; i++, wp4++, rp++) {
                *wp4 = (*rp << 16) & 0xffffffff;
            }
        } else if (24 == src_res) {
            for (i=0; i<count; i++, wp4++, rp++) {
                *wp4 = (*rp << 8) & 0xffffffff;
            }
        } else if (32 == src_res) {
            for (i=0; i<count; i++, wp4++, rp++) {
                *wp4 = *rp;
            }
        }
    }

    _LEAVE;
}

/* --- */

static gpointer flac_play_loop(gpointer arg) {

    /*
     * The main play loop.
     * Decode a frame, push the decoded data to the output plugin
     * chunkwise. Repeat until finished.
     */

    gint32* read_pointer;
    gint elements_left;
    gint seek_sample;
    FLAC__StreamDecoderState state;
    struct stream_info stream_info;
    guint sample_count;
    void* play_buffer;
    InputPlayback* playback = (InputPlayback *) arg;

    _ENTER;

    if (NULL == (play_buffer = malloc(BUFFER_SIZE_BYTE))) {
        _ERROR("Could not allocate conversion buffer");
        playback->playing = FALSE;
        _LEAVE NULL;
    }

    stream_info.samplerate = main_info->stream.samplerate;
    stream_info.channels = main_info->stream.channels;
    main_info->metadata_changed = FALSE;

    if (!playback->output->open_audio(FMT_S16_NE,
        main_info->stream.samplerate,
        main_info->stream.channels)) {
        playback->playing = FALSE;
        _ERROR("Could not open output plugin!");
        _LEAVE NULL;
    }

    while (TRUE == playback->playing) {

        /*
         * Try to decode a single frame of audio
         */
        if (FALSE == FLAC__stream_decoder_process_single(main_decoder)) {
            /*
             * Something went wrong
             */
            _ERROR("Error while decoding!");
            break;
        }

        /*
         * Maybe the metadata changed midstream. Check.
         */

        if (main_info->metadata_changed) {
            /*
             * Samplerate and channels are important
             */
            if (stream_info.samplerate != main_info->stream.samplerate) {
                _ERROR("Samplerate changed midstream (now: %d, was: %d). This is not supported yet.",
                    main_info->stream.samplerate, stream_info.samplerate);
                break;
            }

            if (stream_info.channels != main_info->stream.channels) {
                _ERROR("Number of channels changed midstream (now: %d, was: %d). This is not supported yet.",
                    main_info->stream.channels, stream_info.channels);
                break;
            }

            main_info->metadata_changed = FALSE;
        }

        /*
         * Compare the frame metadata to the current stream metadata
         */
        if (main_info->stream.samplerate != main_info->frame.samplerate) {
            _ERROR("Frame samplerate mismatch (stream: %d, frame: %d)!",
                main_info->stream.samplerate, main_info->frame.samplerate);
            break;
        }

        if (main_info->stream.channels != main_info->frame.channels) {
            _ERROR("Frame channel mismatch (stream: %d, frame: %d)!",
                main_info->stream.channels, main_info->frame.channels);
            break;
        }

        /*
         * If the frame decoded was an audio frame we now
         * have data in info->output_buffer
         *
         * The data in this buffer is in 32 bit wide samples, even if the
         * real sample width is smaller. It has to be converted before
         * feeding it to he output plugin.
         *
         * Feed the data in chunks of OUTPUT_BLOCK_SIZE elements to the output
         * plugin
         */
        read_pointer = main_info->output_buffer;
        elements_left = main_info->buffer_used;

        while ((TRUE == playback->playing) && (0 != elements_left)) {

            sample_count = MIN(OUTPUT_BLOCK_SIZE, elements_left);

            /*
             * Convert the audio.
             * Currently this is hardcoded to 16 bit.
             * 8 and 24 bit are sampled up/down linear.
             */
            _DEBUG("Converting %d samples to FMT_S16_NE", sample_count);
            squeeze_audio(read_pointer, play_buffer, sample_count, main_info->frame.bits_per_sample, 16);

            _DEBUG("Copying %d samples to output plugin", sample_count);

            playback->pass_audio(playback, FMT_S16_NE, main_info->frame.channels, sample_count * sizeof(gint16), play_buffer, NULL);

            read_pointer += sample_count;
            elements_left -= sample_count;

            _DEBUG("%d elements left to be output", elements_left);
        }

        /*
         * Clear the buffer.
         */
        main_info->write_pointer = main_info->output_buffer;
        main_info->buffer_free = BUFFER_SIZE_SAMP;
        main_info->buffer_used = 0;

        /*
         * Do we have to seek to somewhere?
         */
        if (-1 != seek_to) {
            _DEBUG("Seek requested to %d seconds", seek_to);

            seek_sample = seek_to * main_info->stream.samplerate;
            _DEBUG("Seek requested to sample %d", seek_sample);
            if (FALSE == FLAC__stream_decoder_seek_absolute(main_decoder, seek_sample)) {
                _ERROR("Could not seek to sample %d!", seek_sample);
            } else {
                /*
                 * Flush the buffers
                 */
                flac_ip.output->flush(seek_to * 1000);
            }
            seek_to = -1;
        }

        /*
         * Have we reached the end of the stream?
         */
        state = FLAC__stream_decoder_get_state(main_decoder);
        if (0 == elements_left && (FLAC__STREAM_DECODER_END_OF_STREAM == state)) {
            /*
             * Yes. Drain the output buffer and stop playing.
             */

            _DEBUG("End of stream reached, draining output buffer");

            while((-1 == seek_to) && flac_ip.output->buffer_playing() && playback->playing == TRUE) {
                g_usleep(40000);
            }

            if (-1 == seek_to) {
                _DEBUG("Output buffer empty.");
                playback->playing = FALSE;
            }
        }
    }

    /*
     * Clean up a bit
     */
    playback->playing = FALSE;
    _DEBUG("Closing audio device");
    flac_ip.output->close_audio();
    _DEBUG("Audio device closed");

    free(play_buffer);

    if (FALSE == FLAC__stream_decoder_flush(main_decoder)) {
        _ERROR("Could not flush decoder state!");
    }

    _LEAVE NULL;
}

/* --- */

void flac_play_file (InputPlayback* input) {

    VFSFile* fd;
    gint l;

    _ENTER;

    if (!plugin_initialized) {
        _ERROR("plugin not initialized!");
        _LEAVE;
    }

    /*
     * Open the file
     */
    if (NULL == (fd = aud_vfs_fopen(input->filename, "rb"))) {
        _ERROR("Could not open file for reading! (%s)", input->filename);
        _LEAVE;
    }

    if (FALSE == read_metadata(fd, main_decoder, main_info)) {
        _ERROR("Could not prepare file for playing!");
        _LEAVE;
    }

    /*
     * Calculate the length
     */
    if (0 == main_info->stream.samplerate) {
        _ERROR("Invalid sample rate for stream!");
        l = -1;
    } else {
        l = (main_info->stream.samples / main_info->stream.samplerate) * 1000;
    }

    input->playing = TRUE;

    input->set_params(input, get_title(input->filename, main_info), l, -1, main_info->stream.samplerate, main_info->stream.channels);

    thread = g_thread_self();
    input->set_pb_ready(input);
    flac_play_loop(input);

    _LEAVE;
}

/* --- */

void flac_stop(InputPlayback* input) {

    _ENTER;

    input->playing = FALSE;

    if (NULL != thread) {
        /*
         * Wait for the decoder thread to finish
         */
        _DEBUG("Waiting for decoder thread to die...");
        g_thread_join(thread);
        thread = NULL;
        _DEBUG("Decoder thread has finished");
    }

    reset_info(main_info, TRUE);

    _LEAVE;
}

/* --- */

void flac_pause(InputPlayback* input, gshort p) {

    _ENTER;

    input->output->pause(p);

    _LEAVE;
}

/* --- */

void flac_seek(InputPlayback* input, gint time) {

    _ENTER;

    if (!input->playing) {
        _DEBUG("Can not seek while not playing");
        _LEAVE;
    }

    _DEBUG("Requesting seek to %d", time);
    seek_to = time;

    while (-1 != seek_to) {
        g_usleep(10000);
    }

    _LEAVE;
}

/* --- */

void flac_get_song_info(gchar* filename, gchar** title, gint* length) {

    gint l;
    VFSFile* fd;

    _ENTER;

    _DEBUG("Testing file: %s", filename);
    /*
     * Open the file
     */
    if (NULL == (fd = aud_vfs_fopen(filename, "rb"))) {
        _ERROR("Could not open file for reading! (%s)", filename);
        _LEAVE;
    }

    INFO_LOCK(test_info);

    if (FALSE == read_metadata(fd, test_decoder, test_info)) {
        _ERROR("Could not read file info!");
        *length = -1;
        *title = g_strdup("");
        reset_info(test_info, TRUE);
        INFO_UNLOCK(test_info);
        _LEAVE;
    }

    /*
     * Calculate the stream length (milliseconds)
     */
    if (0 == test_info->stream.samplerate) {
        _ERROR("Invalid sample rate for stream!");
        l = -1;
    } else {
        l = (test_info->stream.samples / test_info->stream.samplerate) * 1000;
        _DEBUG("Stream length: %d seconds", l/1000);
    }

    *length = l;
    *title = get_title(filename, test_info);

    reset_info(test_info, TRUE);
    INFO_UNLOCK(test_info);

    _LEAVE;
}

/* --- */

Tuple *flac_get_song_tuple(gchar* filename) {

    VFSFile *fd;
    Tuple *tuple;

    _ENTER;

    _DEBUG("Testing file: %s", filename);
    /*
     * Open the file
     */
    if (NULL == (fd = aud_vfs_fopen(filename, "rb"))) {
        _ERROR("Could not open file for reading! (%s)", filename);
        _LEAVE NULL;
    }

    INFO_LOCK(test_info);

    if (FALSE == read_metadata(fd, test_decoder, test_info)) {
        _ERROR("Could not read metadata tuple for file <%s>", filename);
        reset_info(test_info, TRUE);
        INFO_UNLOCK(test_info);
        _LEAVE NULL;
    }

    tuple = get_tuple(filename, test_info);

    reset_info(test_info, TRUE);
    INFO_UNLOCK(test_info);

    _LEAVE tuple;
}

/* --- */

void flac_aboutbox(void) {

    static GtkWidget* about_window;
    gchar *about_text;

    if (about_window) {
        gtk_window_present(GTK_WINDOW(about_window));
        return;
    }

    about_text = g_strjoin("", _("FLAC Audio Plugin "), _VERSION,
			       _("\n\nOriginal code by\n"
                               "Ralf Ertzinger <[e-mail]>\n"
                               "\n"
                               "http://www.skytale.net/projects/bmp-flac2/"), NULL);

    about_window = audacious_info_dialog(_("About FLAC Audio Plugin"),
                                     about_text,
                                     _("OK"), FALSE, NULL, NULL);

    g_signal_connect(G_OBJECT(about_window), "destroy",
                     G_CALLBACK(gtk_widget_destroyed), &about_window);
    g_free(about_text);
}