/* * Copyright (C) 2003-2011 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. */ #include "config.h" #include "decoder_api.h" #include "audio_check.h" #include #include #include #include #include #include #include #include #include #include #include #include #include #undef G_LOG_DOMAIN #define G_LOG_DOMAIN "ffmpeg" static GLogLevelFlags level_ffmpeg_to_glib(int level) { if (level <= AV_LOG_FATAL) return G_LOG_LEVEL_CRITICAL; if (level <= AV_LOG_ERROR) return G_LOG_LEVEL_WARNING; if (level <= AV_LOG_INFO) return G_LOG_LEVEL_MESSAGE; return G_LOG_LEVEL_DEBUG; } static void mpd_ffmpeg_log_callback(G_GNUC_UNUSED void *ptr, int level, const char *fmt, va_list vl) { const AVClass * cls = NULL; if (ptr != NULL) cls = *(const AVClass *const*)ptr; if (cls != NULL) { char *domain = g_strconcat(G_LOG_DOMAIN, "/", cls->item_name(ptr), NULL); g_logv(domain, level_ffmpeg_to_glib(level), fmt, vl); g_free(domain); } } struct mpd_ffmpeg_stream { struct decoder *decoder; struct input_stream *input; #if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0) AVIOContext *io; #else ByteIOContext *io; #endif unsigned char buffer[8192]; }; static int mpd_ffmpeg_stream_read(void *opaque, uint8_t *buf, int size) { struct mpd_ffmpeg_stream *stream = opaque; return decoder_read(stream->decoder, stream->input, (void *)buf, size); } static int64_t mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence) { struct mpd_ffmpeg_stream *stream = opaque; if (whence == AVSEEK_SIZE) return stream->input->size; if (!input_stream_seek(stream->input, pos, whence, NULL)) return -1; return stream->input->offset; } static struct mpd_ffmpeg_stream * mpd_ffmpeg_stream_open(struct decoder *decoder, struct input_stream *input) { struct mpd_ffmpeg_stream *stream = g_new(struct mpd_ffmpeg_stream, 1); stream->decoder = decoder; stream->input = input; #if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0) stream->io = avio_alloc_context(stream->buffer, sizeof(stream->buffer), false, stream, mpd_ffmpeg_stream_read, NULL, input->seekable ? mpd_ffmpeg_stream_seek : NULL); #else stream->io = av_alloc_put_byte(stream->buffer, sizeof(stream->buffer), false, stream, mpd_ffmpeg_stream_read, NULL, input->seekable ? mpd_ffmpeg_stream_seek : NULL); #endif if (stream->io == NULL) { g_free(stream); return NULL; } return stream; } static void mpd_ffmpeg_stream_close(struct mpd_ffmpeg_stream *stream) { av_free(stream->io); g_free(stream); } static bool ffmpeg_init(G_GNUC_UNUSED const struct config_param *param) { av_log_set_callback(mpd_ffmpeg_log_callback); av_register_all(); return true; } static int ffmpeg_find_audio_stream(const AVFormatContext *format_context) { for (unsigned i = 0; i < format_context->nb_streams; ++i) if (format_context->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO) return i; return -1; } /** * On some platforms, libavcodec wants the output buffer aligned to 16 * bytes (because it uses SSE/Altivec internally). This function * returns the aligned version of the specified buffer, and corrects * the buffer size. */ static void * align16(void *p, size_t *length_p) { unsigned add = 16 - (size_t)p % 16; *length_p -= add; return (char *)p + add; } static enum decoder_command ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is, const AVPacket *packet, AVCodecContext *codec_context, const AVRational *time_base) { if (packet->pts != (int64_t)AV_NOPTS_VALUE) decoder_timestamp(decoder, av_rescale_q(packet->pts, *time_base, (AVRational){1, 1})); #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0) AVPacket packet2 = *packet; #else const uint8_t *packet_data = packet->data; int packet_size = packet->size; #endif uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16]; size_t buffer_size = sizeof(audio_buf); int16_t *aligned_buffer = align16(audio_buf, &buffer_size); enum decoder_command cmd = DECODE_COMMAND_NONE; while ( #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0) packet2.size > 0 && #else packet_size > 0 && #endif cmd == DECODE_COMMAND_NONE) { int audio_size = buffer_size; #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0) int len = avcodec_decode_audio3(codec_context, aligned_buffer, &audio_size, &packet2); #else int len = avcodec_decode_audio2(codec_context, aligned_buffer, &audio_size, packet_data, packet_size); #endif if (len < 0) { /* if error, we skip the frame */ g_message("decoding failed\n"); break; } #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0) packet2.data += len; packet2.size -= len; #else packet_data += len; packet_size -= len; #endif if (audio_size <= 0) continue; cmd = decoder_data(decoder, is, aligned_buffer, audio_size, codec_context->bit_rate / 1000); } return cmd; } static enum sample_format ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context) { switch (codec_context->sample_fmt) { case SAMPLE_FMT_S16: return SAMPLE_FORMAT_S16; case SAMPLE_FMT_S32: return SAMPLE_FORMAT_S32; default: g_warning("Unsupported libavcodec SampleFormat value: %d", codec_context->sample_fmt); return SAMPLE_FORMAT_UNDEFINED; } } static AVInputFormat * ffmpeg_probe(struct decoder *decoder, struct input_stream *is) { enum { BUFFER_SIZE = 16384, PADDING = 16, }; unsigned char *buffer = g_malloc(BUFFER_SIZE); size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE); if (nbytes <= PADDING || !input_stream_seek(is, 0, SEEK_SET, NULL)) { g_free(buffer); return NULL; } /* some ffmpeg parsers (e.g. ac3_parser.c) read a few bytes beyond the declared buffer limit, which makes valgrind angry; this workaround removes some padding from the buffer size */ nbytes -= PADDING; AVProbeData avpd = { .buf = buffer, .buf_size = nbytes, .filename = is->uri, }; AVInputFormat *format = av_probe_input_format(&avpd, true); g_free(buffer); return format; } static void ffmpeg_decode(struct decoder *decoder, struct input_stream *input) { AVInputFormat *input_format = ffmpeg_probe(decoder, input); if (input_format == NULL) return; g_debug("detected input format '%s' (%s)", input_format->name, input_format->long_name); struct mpd_ffmpeg_stream *stream = mpd_ffmpeg_stream_open(decoder, input); if (stream == NULL) { g_warning("Failed to open stream"); return; } //ffmpeg works with ours "fileops" helper AVFormatContext *format_context; if (av_open_input_stream(&format_context, stream->io, input->uri, input_format, NULL) != 0) { g_warning("Open failed\n"); mpd_ffmpeg_stream_close(stream); return; } if (av_find_stream_info(format_context)<0) { g_warning("Couldn't find stream info\n"); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); return; } int audio_stream = ffmpeg_find_audio_stream(format_context); if (audio_stream == -1) { g_warning("No audio stream inside\n"); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); return; } AVCodecContext *codec_context = format_context->streams[audio_stream]->codec; if (codec_context->codec_name[0] != 0) g_debug("codec '%s'", codec_context->codec_name); AVCodec *codec = avcodec_find_decoder(codec_context->codec_id); if (!codec) { g_warning("Unsupported audio codec\n"); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); return; } if (avcodec_open(codec_context, codec)<0) { g_warning("Could not open codec\n"); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); return; } GError *error = NULL; struct audio_format audio_format; if (!audio_format_init_checked(&audio_format, codec_context->sample_rate, ffmpeg_sample_format(codec_context), codec_context->channels, &error)) { g_warning("%s", error->message); g_error_free(error); avcodec_close(codec_context); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); return; } int total_time = format_context->duration != (int64_t)AV_NOPTS_VALUE ? format_context->duration / AV_TIME_BASE : 0; decoder_initialized(decoder, &audio_format, input->seekable, total_time); enum decoder_command cmd; do { AVPacket packet; if (av_read_frame(format_context, &packet) < 0) /* end of file */ break; if (packet.stream_index == audio_stream) cmd = ffmpeg_send_packet(decoder, input, &packet, codec_context, &format_context->streams[audio_stream]->time_base); else cmd = decoder_get_command(decoder); av_free_packet(&packet); if (cmd == DECODE_COMMAND_SEEK) { int64_t where = decoder_seek_where(decoder) * AV_TIME_BASE; if (av_seek_frame(format_context, -1, where, 0) < 0) decoder_seek_error(decoder); else decoder_command_finished(decoder); } } while (cmd != DECODE_COMMAND_STOP); avcodec_close(codec_context); av_close_input_stream(format_context); mpd_ffmpeg_stream_close(stream); } typedef struct ffmpeg_tag_map { enum tag_type type; const char *name; } ffmpeg_tag_map; static const ffmpeg_tag_map ffmpeg_tag_maps[] = { { TAG_TITLE, "title" }, #if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(50<<8)) { TAG_ARTIST, "artist" }, { TAG_DATE, "date" }, #else { TAG_ARTIST, "author" }, { TAG_DATE, "year" }, #endif { TAG_ALBUM, "album" }, { TAG_COMMENT, "comment" }, { TAG_GENRE, "genre" }, { TAG_TRACK, "track" }, { TAG_ARTIST_SORT, "author-sort" }, { TAG_ALBUM_ARTIST, "album_artist" }, { TAG_ALBUM_ARTIST_SORT, "album_artist-sort" }, { TAG_COMPOSER, "composer" }, { TAG_PERFORMER, "performer" }, { TAG_DISC, "disc" }, }; static bool ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m, const ffmpeg_tag_map tag_map) { AVMetadataTag *mt = NULL; while ((mt = av_metadata_get(m, tag_map.name, mt, 0)) != NULL) tag_add_item(tag, tag_map.type, mt->value); return mt != NULL; } //no tag reading in ffmpeg, check if playable static struct tag * ffmpeg_stream_tag(struct input_stream *is) { AVInputFormat *input_format = ffmpeg_probe(NULL, is); if (input_format == NULL) return NULL; struct mpd_ffmpeg_stream *stream = mpd_ffmpeg_stream_open(NULL, is); if (stream == NULL) return NULL; AVFormatContext *f; if (av_open_input_stream(&f, stream->io, is->uri, input_format, NULL) != 0) { mpd_ffmpeg_stream_close(stream); return NULL; } if (av_find_stream_info(f) < 0) { av_close_input_stream(f); mpd_ffmpeg_stream_close(stream); return NULL; } struct tag *tag = tag_new(); tag->time = f->duration != (int64_t)AV_NOPTS_VALUE ? f->duration / AV_TIME_BASE : 0; av_metadata_conv(f, NULL, f->iformat->metadata_conv); for (unsigned i = 0; i < sizeof(ffmpeg_tag_maps)/sizeof(ffmpeg_tag_map); i++) { int idx = ffmpeg_find_audio_stream(f); ffmpeg_copy_metadata(tag, f->metadata, ffmpeg_tag_maps[i]); if (idx >= 0) ffmpeg_copy_metadata(tag, f->streams[idx]->metadata, ffmpeg_tag_maps[i]); } av_close_input_stream(f); mpd_ffmpeg_stream_close(stream); return tag; } /** * A list of extensions found for the formats supported by ffmpeg. * This list is current as of 02-23-09; To find out if there are more * supported formats, check the ffmpeg changelog since this date for * more formats. */ static const char *const ffmpeg_suffixes[] = { "16sv", "3g2", "3gp", "4xm", "8svx", "aa3", "aac", "ac3", "afc", "aif", "aifc", "aiff", "al", "alaw", "amr", "anim", "apc", "ape", "asf", "atrac", "au", "aud", "avi", "avm2", "avs", "bap", "bfi", "c93", "cak", "cin", "cmv", "cpk", "daud", "dct", "divx", "dts", "dv", "dvd", "dxa", "eac3", "film", "flac", "flc", "fli", "fll", "flx", "flv", "g726", "gsm", "gxf", "iss", "m1v", "m2v", "m2t", "m2ts", "m4a", "m4b", "m4v", "mad", "mj2", "mjpeg", "mjpg", "mka", "mkv", "mlp", "mm", "mmf", "mov", "mp+", "mp1", "mp2", "mp3", "mp4", "mpc", "mpeg", "mpg", "mpga", "mpp", "mpu", "mve", "mvi", "mxf", "nc", "nsv", "nut", "nuv", "oga", "ogm", "ogv", "ogx", "oma", "ogg", "omg", "psp", "pva", "qcp", "qt", "r3d", "ra", "ram", "rl2", "rm", "rmvb", "roq", "rpl", "rvc", "shn", "smk", "snd", "sol", "son", "spx", "str", "swf", "tgi", "tgq", "tgv", "thp", "ts", "tsp", "tta", "xa", "xvid", "uv", "uv2", "vb", "vid", "vob", "voc", "vp6", "vmd", "wav", "wma", "wmv", "wsaud", "wsvga", "wv", "wve", NULL }; static const char *const ffmpeg_mime_types[] = { "application/m4a", "application/mp4", "application/octet-stream", "application/ogg", "application/x-ms-wmz", "application/x-ms-wmd", "application/x-ogg", "application/x-shockwave-flash", "application/x-shorten", "audio/8svx", "audio/16sv", "audio/aac", "audio/ac3", "audio/aiff" "audio/amr", "audio/basic", "audio/flac", "audio/m4a", "audio/mp4", "audio/mpeg", "audio/musepack", "audio/ogg", "audio/qcelp", "audio/vorbis", "audio/vorbis+ogg", "audio/x-8svx", "audio/x-16sv", "audio/x-aac", "audio/x-ac3", "audio/x-aiff" "audio/x-alaw", "audio/x-au", "audio/x-dca", "audio/x-eac3", "audio/x-flac", "audio/x-gsm", "audio/x-mace", "audio/x-matroska", "audio/x-monkeys-audio", "audio/x-mpeg", "audio/x-ms-wma", "audio/x-ms-wax", "audio/x-musepack", "audio/x-ogg", "audio/x-vorbis", "audio/x-vorbis+ogg", "audio/x-pn-realaudio", "audio/x-pn-multirate-realaudio", "audio/x-speex", "audio/x-tta" "audio/x-voc", "audio/x-wav", "audio/x-wma", "audio/x-wv", "video/anim", "video/quicktime", "video/msvideo", "video/ogg", "video/theora", "video/x-dv", "video/x-flv", "video/x-matroska", "video/x-mjpeg", "video/x-mpeg", "video/x-ms-asf", "video/x-msvideo", "video/x-ms-wmv", "video/x-ms-wvx", "video/x-ms-wm", "video/x-ms-wmx", "video/x-nut", "video/x-pva", "video/x-theora", "video/x-vid", "video/x-wmv", "video/x-xvid", /* special value for the "ffmpeg" input plugin: all streams by the "ffmpeg" input plugin shall be decoded by this plugin */ "audio/x-mpd-ffmpeg", NULL }; const struct decoder_plugin ffmpeg_decoder_plugin = { .name = "ffmpeg", .init = ffmpeg_init, .stream_decode = ffmpeg_decode, .stream_tag = ffmpeg_stream_tag, .suffixes = ffmpeg_suffixes, .mime_types = ffmpeg_mime_types };