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-rw-r--r--src/inputPlugins/ffmpeg_plugin.c419
1 files changed, 0 insertions, 419 deletions
diff --git a/src/inputPlugins/ffmpeg_plugin.c b/src/inputPlugins/ffmpeg_plugin.c
deleted file mode 100644
index 6455cd1ce..000000000
--- a/src/inputPlugins/ffmpeg_plugin.c
+++ /dev/null
@@ -1,419 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2008 Viliam Mateicka <viliam.mateicka@gmail.com>
- * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- */
-
-#include "../decoder_api.h"
-#include "../log.h"
-#include "../utils.h"
-#include "../log.h"
-
-#include <stdio.h>
-#include <unistd.h>
-#include <stdlib.h>
-#include <string.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <unistd.h>
-
-#ifdef OLD_FFMPEG_INCLUDES
-#include <avcodec.h>
-#include <avformat.h>
-#include <avio.h>
-#else
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavformat/avio.h>
-#endif
-
-typedef struct {
- int audioStream;
- AVFormatContext *pFormatCtx;
- AVCodecContext *aCodecCtx;
- AVCodec *aCodec;
- struct decoder *decoder;
- InputStream *input;
- struct tag *tag;
-} BasePtrs;
-
-typedef struct {
- /** hack - see url_to_base() */
- char url[8];
-
- struct decoder *decoder;
- InputStream *input;
-} FopsHelper;
-
-/**
- * Convert a faked mpd:// URL to a FopsHelper structure. This is a
- * hack because ffmpeg does not provide a nice API for passing a
- * user-defined pointer to mpdurl_open().
- */
-static FopsHelper *url_to_base(const char *url)
-{
- union {
- const char *in;
- FopsHelper *out;
- } u = { .in = url };
- return u.out;
-}
-
-static int mpdurl_open(URLContext *h, const char *filename,
- mpd_unused int flags)
-{
- FopsHelper *base = url_to_base(filename);
- h->priv_data = base;
- h->is_streamed = (base->input->seekable ? 0 : 1);
- return 0;
-}
-
-static int mpdurl_read(URLContext *h, unsigned char *buf, int size)
-{
- int ret;
- FopsHelper *base = (FopsHelper *) h->priv_data;
- while (1) {
- ret = readFromInputStream(base->input, (void *)buf, size);
- if (ret == 0) {
- DEBUG("ret 0\n");
- if (inputStreamAtEOF(base->input) ||
- (base->decoder &&
- decoder_get_command(base->decoder) != DECODE_COMMAND_NONE)) {
- DEBUG("eof stream\n");
- return ret;
- } else {
- my_usleep(10000);
- }
- } else {
- break;
- }
- }
- return ret;
-}
-
-static int64_t mpdurl_seek(URLContext *h, int64_t pos, int whence)
-{
- FopsHelper *base = (FopsHelper *) h->priv_data;
- if (whence != AVSEEK_SIZE) { //only ftell
- (void) seekInputStream(base->input, pos, whence);
- }
- return base->input->offset;
-}
-
-static int mpdurl_close(URLContext *h)
-{
- FopsHelper *base = (FopsHelper *) h->priv_data;
- if (base && base->input->seekable) {
- (void) seekInputStream(base->input, 0, SEEK_SET);
- }
- h->priv_data = 0;
- return 0;
-}
-
-static URLProtocol mpdurl_fileops = {
- .name = "mpd",
- .url_open = mpdurl_open,
- .url_read = mpdurl_read,
- .url_seek = mpdurl_seek,
- .url_close = mpdurl_close,
-};
-
-static int ffmpeg_init(void)
-{
- av_register_all();
- register_protocol(&mpdurl_fileops);
- return 0;
-}
-
-static int ffmpeg_helper(InputStream *input, int (*callback)(BasePtrs *ptrs),
- BasePtrs *ptrs)
-{
- AVFormatContext *pFormatCtx;
- AVCodecContext *aCodecCtx;
- AVCodec *aCodec;
- int ret, audioStream;
- unsigned i;
- FopsHelper fopshelp = {
- .url = "mpd://X", /* only the mpd:// prefix matters */
- };
-
- fopshelp.input = input;
- if (ptrs && ptrs->decoder) {
- fopshelp.decoder = ptrs->decoder; //are we in decoding loop ?
- } else {
- fopshelp.decoder = NULL;
- }
-
- //ffmpeg works with ours "fileops" helper
- if (av_open_input_file(&pFormatCtx, fopshelp.url, NULL, 0, NULL)!=0) {
- ERROR("Open failed!\n");
- return -1;
- }
-
- if (av_find_stream_info(pFormatCtx)<0) {
- ERROR("Couldn't find stream info!\n");
- return -1;
- }
-
- audioStream = -1;
- for(i=0; i<pFormatCtx->nb_streams; i++) {
- if (pFormatCtx->streams[i]->codec->codec_type==CODEC_TYPE_AUDIO &&
- audioStream < 0) {
- audioStream=i;
- }
- }
-
- if(audioStream==-1) {
- ERROR("No audio stream inside!\n");
- return -1;
- }
-
- aCodecCtx = pFormatCtx->streams[audioStream]->codec;
- aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
-
- if (!aCodec) {
- ERROR("Unsupported audio codec!\n");
- return -1;
- }
-
- if (avcodec_open(aCodecCtx, aCodec)<0) {
- ERROR("Could not open codec!\n");
- return -1;
- }
-
- if (callback) {
- ptrs->audioStream = audioStream;
- ptrs->pFormatCtx = pFormatCtx;
- ptrs->aCodecCtx = aCodecCtx;
- ptrs->aCodec = aCodec;
-
- ret = (*callback)( ptrs );
- } else {
- ret = 0;
- DEBUG("playable\n");
- }
-
- avcodec_close(aCodecCtx);
- av_close_input_file(pFormatCtx);
-
- return ret;
-}
-
-static bool ffmpeg_try_decode(InputStream *input)
-{
- int ret;
- if (input->seekable) {
- ret = ffmpeg_helper(input, NULL, NULL);
- } else {
- ret = 0;
- }
- return (ret == -1 ? 0 : 1);
-}
-
-static int ffmpeg_decode_internal(BasePtrs *base)
-{
- struct decoder *decoder = base->decoder;
- AVCodecContext *aCodecCtx = base->aCodecCtx;
- AVFormatContext *pFormatCtx = base->pFormatCtx;
- AVPacket packet;
- int len, audio_size;
- int position;
- struct audio_format audio_format;
- int current, total_time;
- uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
-
- total_time = 0;
-
- DEBUG("decoder_start\n");
-
- if (aCodecCtx->channels > 2) {
- aCodecCtx->channels = 2;
- }
-
- audio_format.bits = (uint8_t)16;
- audio_format.sample_rate = (unsigned int)aCodecCtx->sample_rate;
- audio_format.channels = aCodecCtx->channels;
-
- // frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
- // total_time = ((float)frame_count / (float)audio_format.sample_rate);
-
- //there is some problem with this on some demux (mp3 at least)
- if (pFormatCtx->duration != (int)AV_NOPTS_VALUE) {
- total_time = pFormatCtx->duration / AV_TIME_BASE;
- }
-
- DEBUG("ffmpeg sample rate: %dHz %d channels\n",
- aCodecCtx->sample_rate, aCodecCtx->channels);
-
- decoder_initialized(decoder, &audio_format, total_time);
-
- position = 0;
-
- DEBUG("duration:%d (%d secs)\n", (int) pFormatCtx->duration,
- (int) total_time);
-
- do {
-
- if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
-
- DEBUG("seek\n");
- decoder_clear(decoder);
- current = decoder_seek_where(decoder) * AV_TIME_BASE;
-
- if (av_seek_frame(pFormatCtx, -1, current , 0) < 0) {
- WARNING("seek to %d failed\n", current);
- }
-
- decoder_command_finished(decoder);
- }
-
- if (av_read_frame(pFormatCtx, &packet) >= 0) {
- if(packet.stream_index == base->audioStream) {
-
- position = av_rescale_q(packet.pts, pFormatCtx->streams[base->audioStream]->time_base,
- (AVRational){1, 1});
-
- audio_size = sizeof(audio_buf);
- len = avcodec_decode_audio2(aCodecCtx,
- (int16_t *)audio_buf,
- &audio_size,
- packet.data,
- packet.size);
-
- if(len >= 0) {
- if(audio_size >= 0) {
- // DEBUG("sending data %d/%d\n", audio_size, len);
-
- decoder_data(decoder, NULL, 1,
- audio_buf, audio_size,
- position, //(float)current / (float)audio_format.sample_rate,
- aCodecCtx->bit_rate / 1000, NULL);
-
- }
- } else {
- WARNING("skiping frame!\n");
- }
- }
- av_free_packet(&packet);
- } else {
- //end of file
- break;
- }
- } while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
-
- decoder_flush(decoder);
-
- DEBUG("decoder finish\n");
-
- return 0;
-}
-
-static int ffmpeg_decode(struct decoder *decoder, InputStream *input)
-{
- BasePtrs base;
- int ret;
-
- DEBUG("decode start\n");
-
- base.input = input;
- base.decoder = decoder;
-
- ret = ffmpeg_helper(input, ffmpeg_decode_internal, &base);
-
- DEBUG("decode finish\n");
-
- return ret;
-}
-
-static int ffmpeg_tag_internal(BasePtrs *base)
-{
- struct tag *tag = (struct tag *) base->tag;
-
- if (base->pFormatCtx->duration != (int)AV_NOPTS_VALUE) {
- tag->time = base->pFormatCtx->duration / AV_TIME_BASE;
- } else {
- tag->time = 0;
- }
- return 0;
-}
-
-//no tag reading in ffmpeg, check if playable
-static struct tag *ffmpeg_tag(char *file)
-{
- InputStream input;
- BasePtrs base;
- int ret;
- struct tag *tag = NULL;
-
- if (openInputStream(&input, file) < 0) {
- ERROR("failed to open %s\n", file);
- return NULL;
- }
-
- tag = tag_new();
-
- base.tag = tag;
- ret = ffmpeg_helper(&input, ffmpeg_tag_internal, &base);
-
- if (ret != 0) {
- free(tag);
- tag = NULL;
- }
-
- closeInputStream(&input);
-
- return tag;
-}
-
-/**
- * ffmpeg can decode almost everything from open codecs
- * and also some of propietary codecs
- * its hard to tell what can ffmpeg decode
- * we can later put this into configure script
- * to be sure ffmpeg is used to handle
- * only that files
- */
-
-static const char *ffmpeg_Suffixes[] = {
- "wma", "asf", "wmv", "mpeg", "mpg", "avi", "vob", "mov", "qt", "swf", "rm", "swf",
- "mp1", "mp2", "mp3", "mp4", "m4a", "flac", "ogg", "wav", "au", "aiff", "aif", "ac3", "aac", "mpc",
- NULL
-};
-
-//not sure if this is correct...
-static const char *ffmpeg_Mimetypes[] = {
- "video/x-ms-asf",
- "audio/x-ms-wma",
- "audio/x-ms-wax",
- "video/x-ms-wmv",
- "video/x-ms-wvx",
- "video/x-ms-wm",
- "video/x-ms-wmx",
- "application/x-ms-wmz",
- "application/x-ms-wmd",
- "audio/mpeg",
- NULL
-};
-
-struct decoder_plugin ffmpegPlugin = {
- .name = "ffmpeg",
- .init = ffmpeg_init,
- .try_decode = ffmpeg_try_decode,
- .stream_decode = ffmpeg_decode,
- .tag_dup = ffmpeg_tag,
- .stream_types = INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
- .suffixes = ffmpeg_Suffixes,
- .mime_types = ffmpeg_Mimetypes
-};