/* * 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. */ /* necessary because libavutil/common.h uses UINT64_C */ #define __STDC_CONSTANT_MACROS #include "config.h" #include "FfmpegDecoderPlugin.hxx" #include "lib/ffmpeg/Domain.hxx" #include "../DecoderAPI.hxx" #include "FfmpegMetaData.hxx" #include "tag/TagHandler.hxx" #include "input/InputStream.hxx" #include "CheckAudioFormat.hxx" #include "util/Error.hxx" #include "util/Domain.hxx" #include "LogV.hxx" extern "C" { #include #include #include #include #include #include #if LIBAVUTIL_VERSION_MAJOR >= 53 #include #endif } #include #include /* suppress the ffmpeg compatibility macro */ #ifdef SampleFormat #undef SampleFormat #endif static LogLevel import_ffmpeg_level(int level) { if (level <= AV_LOG_FATAL) return LogLevel::ERROR; if (level <= AV_LOG_WARNING) return LogLevel::WARNING; if (level <= AV_LOG_INFO) return LogLevel::INFO; return LogLevel::DEBUG; } static void mpd_ffmpeg_log_callback(gcc_unused void *ptr, int level, const char *fmt, va_list vl) { const AVClass * cls = nullptr; if (ptr != nullptr) cls = *(const AVClass *const*)ptr; if (cls != nullptr) { char domain[64]; snprintf(domain, sizeof(domain), "%s/%s", ffmpeg_domain.GetName(), cls->item_name(ptr)); const Domain d(domain); LogFormatV(d, import_ffmpeg_level(level), fmt, vl); } } struct AvioStream { Decoder *const decoder; InputStream &input; AVIOContext *io; unsigned char buffer[8192]; AvioStream(Decoder *_decoder, InputStream &_input) :decoder(_decoder), input(_input), io(nullptr) {} ~AvioStream() { if (io != nullptr) av_free(io); } bool Open(); }; static int mpd_ffmpeg_stream_read(void *opaque, uint8_t *buf, int size) { AvioStream *stream = (AvioStream *)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) { AvioStream *stream = (AvioStream *)opaque; switch (whence) { case SEEK_SET: break; case SEEK_CUR: pos += stream->input.GetOffset(); break; case SEEK_END: if (!stream->input.KnownSize()) return -1; pos += stream->input.GetSize(); break; case AVSEEK_SIZE: if (!stream->input.KnownSize()) return -1; return stream->input.GetSize(); default: return -1; } if (!stream->input.LockSeek(pos, IgnoreError())) return -1; return stream->input.GetOffset(); } bool AvioStream::Open() { io = avio_alloc_context(buffer, sizeof(buffer), false, this, mpd_ffmpeg_stream_read, nullptr, input.IsSeekable() ? mpd_ffmpeg_stream_seek : nullptr); return io != nullptr; } /** * API compatibility wrapper for av_open_input_stream() and * avformat_open_input(). */ static int mpd_ffmpeg_open_input(AVFormatContext **ic_ptr, AVIOContext *pb, const char *filename, AVInputFormat *fmt) { AVFormatContext *context = avformat_alloc_context(); if (context == nullptr) return AVERROR(ENOMEM); context->pb = pb; *ic_ptr = context; return avformat_open_input(ic_ptr, filename, fmt, nullptr); } static bool ffmpeg_init(gcc_unused const config_param ¶m) { 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 == AVMEDIA_TYPE_AUDIO) return i; return -1; } gcc_const static double time_from_ffmpeg(int64_t t, const AVRational time_base) { assert(t != (int64_t)AV_NOPTS_VALUE); return (double)av_rescale_q(t, time_base, (AVRational){1, 1024}) / (double)1024; } template static constexpr AVRational RatioToAVRational() { return { Ratio::num, Ratio::den }; } gcc_const static int64_t time_to_ffmpeg(SongTime t, const AVRational time_base) { return av_rescale_q(t.count(), RatioToAVRational(), time_base); } /** * Replace #AV_NOPTS_VALUE with the given fallback. */ static constexpr int64_t timestamp_fallback(int64_t t, int64_t fallback) { return gcc_likely(t != int64_t(AV_NOPTS_VALUE)) ? t : fallback; } /** * Accessor for AVStream::start_time that replaces AV_NOPTS_VALUE with * zero. We can't use AV_NOPTS_VALUE in calculations, and we simply * assume that the stream's start time is zero, which appears to be * the best way out of that situation. */ static int64_t start_time_fallback(const AVStream &stream) { return timestamp_fallback(stream.start_time, 0); } static void copy_interleave_frame2(uint8_t *dest, uint8_t **src, unsigned nframes, unsigned nchannels, unsigned sample_size) { for (unsigned frame = 0; frame < nframes; ++frame) { for (unsigned channel = 0; channel < nchannels; ++channel) { memcpy(dest, src[channel] + frame * sample_size, sample_size); dest += sample_size; } } } /** * Copy PCM data from a AVFrame to an interleaved buffer. */ static int copy_interleave_frame(const AVCodecContext *codec_context, const AVFrame *frame, uint8_t **output_buffer, uint8_t **global_buffer, int *global_buffer_size) { int plane_size; const int data_size = av_samples_get_buffer_size(&plane_size, codec_context->channels, frame->nb_samples, codec_context->sample_fmt, 1); if (data_size <= 0) return data_size; if (av_sample_fmt_is_planar(codec_context->sample_fmt) && codec_context->channels > 1) { if(*global_buffer_size < data_size) { av_freep(global_buffer); *global_buffer = (uint8_t*)av_malloc(data_size); if (!*global_buffer) /* Not enough memory - shouldn't happen */ return AVERROR(ENOMEM); *global_buffer_size = data_size; } *output_buffer = *global_buffer; copy_interleave_frame2(*output_buffer, frame->extended_data, frame->nb_samples, codec_context->channels, av_get_bytes_per_sample(codec_context->sample_fmt)); } else { *output_buffer = frame->extended_data[0]; } return data_size; } static DecoderCommand ffmpeg_send_packet(Decoder &decoder, InputStream &is, const AVPacket *packet, AVCodecContext *codec_context, const AVStream *stream, AVFrame *frame, uint8_t **buffer, int *buffer_size) { if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE) decoder_timestamp(decoder, time_from_ffmpeg(packet->pts - start_time_fallback(*stream), stream->time_base)); AVPacket packet2 = *packet; uint8_t *output_buffer; DecoderCommand cmd = DecoderCommand::NONE; while (packet2.size > 0 && cmd == DecoderCommand::NONE) { int audio_size = 0; int got_frame = 0; int len = avcodec_decode_audio4(codec_context, frame, &got_frame, &packet2); if (len >= 0 && got_frame) { audio_size = copy_interleave_frame(codec_context, frame, &output_buffer, buffer, buffer_size); if (audio_size < 0) len = audio_size; } if (len < 0) { /* if error, we skip the frame */ LogDefault(ffmpeg_domain, "decoding failed, frame skipped"); break; } packet2.data += len; packet2.size -= len; if (audio_size <= 0) continue; cmd = decoder_data(decoder, is, output_buffer, audio_size, codec_context->bit_rate / 1000); } return cmd; } gcc_const static SampleFormat ffmpeg_sample_format(enum AVSampleFormat sample_fmt) { switch (sample_fmt) { case AV_SAMPLE_FMT_S16: case AV_SAMPLE_FMT_S16P: return SampleFormat::S16; case AV_SAMPLE_FMT_S32: case AV_SAMPLE_FMT_S32P: return SampleFormat::S32; case AV_SAMPLE_FMT_FLTP: return SampleFormat::FLOAT; default: break; } char buffer[64]; const char *name = av_get_sample_fmt_string(buffer, sizeof(buffer), sample_fmt); if (name != nullptr) FormatError(ffmpeg_domain, "Unsupported libavcodec SampleFormat value: %s (%d)", name, sample_fmt); else FormatError(ffmpeg_domain, "Unsupported libavcodec SampleFormat value: %d", sample_fmt); return SampleFormat::UNDEFINED; } static AVInputFormat * ffmpeg_probe(Decoder *decoder, InputStream &is) { enum { BUFFER_SIZE = 16384, PADDING = 16, }; unsigned char buffer[BUFFER_SIZE]; size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE); if (nbytes <= PADDING || !is.LockRewind(IgnoreError())) return nullptr; /* 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; avpd.buf = buffer; avpd.buf_size = nbytes; avpd.filename = is.GetURI(); return av_probe_input_format(&avpd, true); } static void ffmpeg_decode(Decoder &decoder, InputStream &input) { AVInputFormat *input_format = ffmpeg_probe(&decoder, input); if (input_format == nullptr) return; FormatDebug(ffmpeg_domain, "detected input format '%s' (%s)", input_format->name, input_format->long_name); AvioStream stream(&decoder, input); if (!stream.Open()) { LogError(ffmpeg_domain, "Failed to open stream"); return; } //ffmpeg works with ours "fileops" helper AVFormatContext *format_context = nullptr; if (mpd_ffmpeg_open_input(&format_context, stream.io, input.GetURI(), input_format) != 0) { LogError(ffmpeg_domain, "Open failed"); return; } const int find_result = avformat_find_stream_info(format_context, nullptr); if (find_result < 0) { LogError(ffmpeg_domain, "Couldn't find stream info"); avformat_close_input(&format_context); return; } int audio_stream = ffmpeg_find_audio_stream(format_context); if (audio_stream == -1) { LogError(ffmpeg_domain, "No audio stream inside"); avformat_close_input(&format_context); return; } AVStream *av_stream = format_context->streams[audio_stream]; AVCodecContext *codec_context = av_stream->codec; #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 25, 0) const AVCodecDescriptor *codec_descriptor = avcodec_descriptor_get(codec_context->codec_id); if (codec_descriptor != nullptr) FormatDebug(ffmpeg_domain, "codec '%s'", codec_descriptor->name); #else if (codec_context->codec_name[0] != 0) FormatDebug(ffmpeg_domain, "codec '%s'", codec_context->codec_name); #endif AVCodec *codec = avcodec_find_decoder(codec_context->codec_id); if (!codec) { LogError(ffmpeg_domain, "Unsupported audio codec"); avformat_close_input(&format_context); return; } const SampleFormat sample_format = ffmpeg_sample_format(codec_context->sample_fmt); if (sample_format == SampleFormat::UNDEFINED) { // (error message already done by ffmpeg_sample_format()) avformat_close_input(&format_context); return; } Error error; AudioFormat audio_format; if (!audio_format_init_checked(audio_format, codec_context->sample_rate, sample_format, codec_context->channels, error)) { LogError(error); avformat_close_input(&format_context); return; } /* the audio format must be read from AVCodecContext by now, because avcodec_open() has been demonstrated to fill bogus values into AVCodecContext.channels - a change that will be reverted later by avcodec_decode_audio3() */ const int open_result = avcodec_open2(codec_context, codec, nullptr); if (open_result < 0) { LogError(ffmpeg_domain, "Could not open codec"); avformat_close_input(&format_context); return; } const SignedSongTime total_time = format_context->duration != (int64_t)AV_NOPTS_VALUE ? SignedSongTime::FromScale(format_context->duration, AV_TIME_BASE) : SignedSongTime::Negative(); decoder_initialized(decoder, audio_format, input.IsSeekable(), total_time); #if LIBAVUTIL_VERSION_MAJOR >= 53 AVFrame *frame = av_frame_alloc(); #else AVFrame *frame = avcodec_alloc_frame(); #endif if (!frame) { LogError(ffmpeg_domain, "Could not allocate frame"); avformat_close_input(&format_context); return; } uint8_t *interleaved_buffer = nullptr; int interleaved_buffer_size = 0; DecoderCommand 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, av_stream, frame, &interleaved_buffer, &interleaved_buffer_size); else cmd = decoder_get_command(decoder); av_free_packet(&packet); if (cmd == DecoderCommand::SEEK) { int64_t where = time_to_ffmpeg(decoder_seek_time(decoder), av_stream->time_base) + start_time_fallback(*av_stream); if (av_seek_frame(format_context, audio_stream, where, AVSEEK_FLAG_ANY) < 0) decoder_seek_error(decoder); else { avcodec_flush_buffers(codec_context); decoder_command_finished(decoder); } } } while (cmd != DecoderCommand::STOP); #if LIBAVUTIL_VERSION_MAJOR >= 53 av_frame_free(&frame); #elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) avcodec_free_frame(&frame); #else av_freep(&frame); #endif av_freep(&interleaved_buffer); avcodec_close(codec_context); avformat_close_input(&format_context); } //no tag reading in ffmpeg, check if playable static bool ffmpeg_scan_stream(InputStream &is, const struct tag_handler *handler, void *handler_ctx) { AVInputFormat *input_format = ffmpeg_probe(nullptr, is); if (input_format == nullptr) return false; AvioStream stream(nullptr, is); if (!stream.Open()) return false; AVFormatContext *f = nullptr; if (mpd_ffmpeg_open_input(&f, stream.io, is.GetURI(), input_format) != 0) return false; const int find_result = avformat_find_stream_info(f, nullptr); if (find_result < 0) { avformat_close_input(&f); return false; } if (f->duration != (int64_t)AV_NOPTS_VALUE) tag_handler_invoke_duration(handler, handler_ctx, f->duration / AV_TIME_BASE); ffmpeg_scan_dictionary(f->metadata, handler, handler_ctx); int idx = ffmpeg_find_audio_stream(f); if (idx >= 0) ffmpeg_scan_dictionary(f->streams[idx]->metadata, handler, handler_ctx); avformat_close_input(&f); return true; } /** * 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", "webm", "wma", "wmv", "wsaud", "wsvga", "wv", "wve", nullptr }; static const char *const ffmpeg_mime_types[] = { "application/flv", "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/webm", "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", nullptr }; const struct DecoderPlugin ffmpeg_decoder_plugin = { "ffmpeg", ffmpeg_init, nullptr, ffmpeg_decode, nullptr, nullptr, ffmpeg_scan_stream, nullptr, ffmpeg_suffixes, ffmpeg_mime_types };