/* * Copyright (C) 2009-2011 Julien BLACHE * * Adapted from mt-daapd: * Copyright (C) 2006-2007 Ron Pedde * * 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 */ #ifdef HAVE_CONFIG_H # include #endif #include #include #include #include #include #include #include #if defined(__linux__) || defined(__GLIBC__) # include # include #elif defined(__FreeBSD__) || defined(__FreeBSD_kernel__) # include #endif #include #include "evhttp/evhttp.h" #include #include #include "logger.h" #include "conffile.h" #include "db.h" #include "transcode.h" #define XCODE_BUFFER_SIZE ((AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2) struct transcode_ctx { AVFormatContext *fmtctx; /* Audio stream */ int astream; AVCodecContext *acodec; /* pCodecCtx */ AVCodec *adecoder; /* pCodec */ AVPacket apacket; AVPacket apacket2; int16_t *abuffer; /* Resampling */ int need_resample; int input_size; ReSampleContext *resample_ctx; int16_t *re_abuffer; off_t offset; uint32_t duration; uint64_t samples; /* WAV header */ int wavhdr; uint8_t header[44]; }; static char *default_codecs = "mpeg,wav"; static char *roku_codecs = "mpeg,mp4a,wma,wav"; static char *itunes_codecs = "mpeg,mp4a,mp4v,alac,wav"; static inline void add_le16(uint8_t *dst, uint16_t val) { dst[0] = val & 0xff; dst[1] = (val >> 8) & 0xff; } static inline void add_le32(uint8_t *dst, uint32_t val) { dst[0] = val & 0xff; dst[1] = (val >> 8) & 0xff; dst[2] = (val >> 16) & 0xff; dst[3] = (val >> 24) & 0xff; } static void make_wav_header(struct transcode_ctx *ctx, off_t *est_size) { uint32_t wav_len; int duration; if (ctx->duration) duration = ctx->duration; else duration = 3 * 60 * 1000; /* 3 minutes, in ms */ if (ctx->samples && !ctx->need_resample) wav_len = 2 * 2 * ctx->samples; else wav_len = 2 * 2 * 44100 * (duration / 1000); *est_size = wav_len + sizeof(ctx->header); memcpy(ctx->header, "RIFF", 4); add_le32(ctx->header + 4, 36 + wav_len); memcpy(ctx->header + 8, "WAVEfmt ", 8); add_le32(ctx->header + 16, 16); add_le16(ctx->header + 20, 1); add_le16(ctx->header + 22, 2); /* channels */ add_le32(ctx->header + 24, 44100); /* samplerate */ add_le32(ctx->header + 28, 44100 * 2 * 2); /* byte rate */ add_le16(ctx->header + 32, 2 * 2); /* block align */ add_le16(ctx->header + 34, 16); /* bits per sample */ memcpy(ctx->header + 36, "data", 4); add_le32(ctx->header + 40, wav_len); } int transcode(struct transcode_ctx *ctx, struct evbuffer *evbuf, int wanted) { int16_t *buf; int buflen; int processed; int used; int stop; int ret; #if BYTE_ORDER == BIG_ENDIAN int i; #endif processed = 0; if (ctx->wavhdr && (ctx->offset == 0)) { evbuffer_add(evbuf, ctx->header, sizeof(ctx->header)); processed += sizeof(ctx->header); ctx->offset += sizeof(ctx->header); } stop = 0; while ((processed < wanted) && !stop) { /* Decode data */ while (ctx->apacket2.size > 0) { buflen = XCODE_BUFFER_SIZE; #if LIBAVCODEC_VERSION_MAJOR >= 53 || (LIBAVCODEC_VERSION_MAJOR == 52 && LIBAVCODEC_VERSION_MINOR >= 32) /* FFmpeg 0.6 */ used = avcodec_decode_audio3(ctx->acodec, ctx->abuffer, &buflen, &ctx->apacket2); #else used = avcodec_decode_audio2(ctx->acodec, ctx->abuffer, &buflen, ctx->apacket2.data, ctx->apacket2.size); #endif if (used < 0) { /* Something happened, skip this packet */ ctx->apacket2.size = 0; break; } ctx->apacket2.data += used; ctx->apacket2.size -= used; /* No frame decoded this time around */ if (buflen == 0) continue; if (ctx->need_resample) { buflen = audio_resample(ctx->resample_ctx, ctx->re_abuffer, ctx->abuffer, buflen / ctx->input_size); if (buflen == 0) { DPRINTF(E_WARN, L_XCODE, "Resample returned no samples!\n"); continue; } buflen = buflen * 2 * 2; /* 16bit samples, 2 channels */ buf = ctx->re_abuffer; } else buf = ctx->abuffer; #if BYTE_ORDER == BIG_ENDIAN /* swap buffer, LE16 */ for (i = 0; i < (buflen / 2); i++) { buf[i] = htole16(buf[i]); } #endif ret = evbuffer_add(evbuf, buf, buflen); if (ret != 0) { DPRINTF(E_WARN, L_XCODE, "Could not copy WAV data to buffer\n"); return -1; } processed += buflen; } /* Read more data */ do { if (ctx->apacket.data) av_free_packet(&ctx->apacket); ret = av_read_frame(ctx->fmtctx, &ctx->apacket); if (ret < 0) { DPRINTF(E_WARN, L_XCODE, "Could not read more data\n"); stop = 1; break; } } while (ctx->apacket.stream_index != ctx->astream); /* Copy apacket and do not mess with it */ ctx->apacket2 = ctx->apacket; } ctx->offset += processed; return processed; } int transcode_seek(struct transcode_ctx *ctx, int ms) { int64_t start_time; int64_t target_pts; int64_t got_pts; int got_ms; int flags; int ret; start_time = ctx->fmtctx->streams[ctx->astream]->start_time; target_pts = ms; target_pts = target_pts * AV_TIME_BASE / 1000; target_pts = av_rescale_q(target_pts, AV_TIME_BASE_Q, ctx->fmtctx->streams[ctx->astream]->time_base); if ((start_time != AV_NOPTS_VALUE) && (start_time > 0)) target_pts += start_time; ret = av_seek_frame(ctx->fmtctx, ctx->astream, target_pts, AVSEEK_FLAG_BACKWARD); if (ret < 0) { DPRINTF(E_WARN, L_XCODE, "Could not seek into stream: %s\n", strerror(AVUNERROR(ret))); return -1; } avcodec_flush_buffers(ctx->acodec); #if LIBAVCODEC_VERSION_MAJOR >= 53 ctx->acodec->skip_frame = AVDISCARD_NONREF; #else ctx->acodec->hurry_up = 1; #endif flags = 0; while (1) { if (ctx->apacket.data) av_free_packet(&ctx->apacket); ret = av_read_frame(ctx->fmtctx, &ctx->apacket); if (ret < 0) { DPRINTF(E_WARN, L_XCODE, "Could not read more data while seeking\n"); flags = 1; break; } if (ctx->apacket.stream_index != ctx->astream) continue; /* Need a pts to return the real position */ if (ctx->apacket.pts == AV_NOPTS_VALUE) continue; break; } #if LIBAVCODEC_VERSION_MAJOR >= 53 ctx->acodec->skip_frame = AVDISCARD_DEFAULT; #else ctx->acodec->hurry_up = 0; #endif /* Error while reading frame above */ if (flags) return -1; /* Copy apacket and do not mess with it */ ctx->apacket2 = ctx->apacket; /* Compute position in ms from pts */ got_pts = ctx->apacket.pts; if ((start_time != AV_NOPTS_VALUE) && (start_time > 0)) got_pts -= start_time; got_pts = av_rescale_q(got_pts, ctx->fmtctx->streams[ctx->astream]->time_base, AV_TIME_BASE_Q); got_ms = got_pts / (AV_TIME_BASE / 1000); DPRINTF(E_DBG, L_XCODE, "Seek wanted %d ms, got %d ms\n", ms, got_ms); return got_ms; } struct transcode_ctx * transcode_setup(struct media_file_info *mfi, off_t *est_size, int wavhdr) { struct transcode_ctx *ctx; int i; int ret; ctx = (struct transcode_ctx *)malloc(sizeof(struct transcode_ctx)); if (!ctx) { DPRINTF(E_WARN, L_XCODE, "Could not allocate transcode context\n"); return NULL; } memset(ctx, 0, sizeof(struct transcode_ctx)); #if LIBAVFORMAT_VERSION_MAJOR >= 53 || (LIBAVFORMAT_VERSION_MAJOR == 53 && LIBAVCODEC_VERSION_MINOR >= 3) ret = avformat_open_input(&ctx->fmtctx, mfi->path, NULL, NULL); #else ret = av_open_input_file(&ctx->fmtctx, mfi->path, NULL, 0, NULL); #endif if (ret != 0) { DPRINTF(E_WARN, L_XCODE, "Could not open file %s: %s\n", mfi->fname, strerror(AVUNERROR(ret))); free(ctx); return NULL; } ret = avformat_find_stream_info(ctx->fmtctx,NULL); if (ret < 0) { DPRINTF(E_WARN, L_XCODE, "Could not find stream info: %s\n", strerror(AVUNERROR(ret))); goto setup_fail; } ctx->astream = -1; for (i = 0; i < ctx->fmtctx->nb_streams; i++) { #if LIBAVCODEC_VERSION_MAJOR >= 53 || (LIBAVCODEC_VERSION_MAJOR == 52 && LIBAVCODEC_VERSION_MINOR >= 64) if (ctx->fmtctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) #else if (ctx->fmtctx->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO) #endif { ctx->astream = i; break; } } if (ctx->astream < 0) { DPRINTF(E_WARN, L_XCODE, "No audio stream found in file %s\n", mfi->fname); goto setup_fail; } ctx->acodec = ctx->fmtctx->streams[ctx->astream]->codec; ctx->adecoder = avcodec_find_decoder(ctx->acodec->codec_id); if (!ctx->adecoder) { DPRINTF(E_WARN, L_XCODE, "No suitable decoder found for codec\n"); goto setup_fail; } if (ctx->adecoder->capabilities & CODEC_CAP_TRUNCATED) ctx->acodec->flags |= CODEC_FLAG_TRUNCATED; ret = avcodec_open2(ctx->acodec, ctx->adecoder,NULL); if (ret != 0) { DPRINTF(E_WARN, L_XCODE, "Could not open codec: %s\n", strerror(AVUNERROR(ret))); goto setup_fail; } ctx->abuffer = (int16_t *)av_malloc(XCODE_BUFFER_SIZE); if (!ctx->abuffer) { DPRINTF(E_WARN, L_XCODE, "Could not allocate transcode buffer\n"); goto setup_fail_codec; } if ((ctx->acodec->sample_fmt != AV_SAMPLE_FMT_S16) // if ((ctx->acodec->sample_fmt != SAMPLE_FMT_S16) || (ctx->acodec->channels != 2) || (ctx->acodec->sample_rate != 44100)) { DPRINTF(E_DBG, L_XCODE, "Setting up resampling (%d@%d)\n", ctx->acodec->channels, ctx->acodec->sample_rate); ctx->resample_ctx = av_audio_resample_init(2, ctx->acodec->channels, 44100, ctx->acodec->sample_rate, AV_SAMPLE_FMT_S16, ctx->acodec->sample_fmt, // SAMPLE_FMT_S16, ctx->acodec->sample_fmt, 16, 10, 0, 0.8); if (!ctx->resample_ctx) { DPRINTF(E_WARN, L_XCODE, "Could not init resample from %d@%d to 2@44100\n", ctx->acodec->channels, ctx->acodec->sample_rate); goto setup_fail_codec; } ctx->re_abuffer = (int16_t *)av_malloc(XCODE_BUFFER_SIZE * 2); if (!ctx->re_abuffer) { DPRINTF(E_WARN, L_XCODE, "Could not allocate resample buffer\n"); audio_resample_close(ctx->resample_ctx); goto setup_fail_codec; } ctx->need_resample = 1; #if LIBAVUTIL_VERSION_MAJOR >= 51 || (LIBAVUTIL_VERSION_MAJOR == 51 && LIBAVUTIL_VERSION_MINOR >= 4) ctx->input_size = ctx->acodec->channels * av_get_bytes_per_sample(ctx->acodec->sample_fmt); #elif LIBAVCODEC_VERSION_MAJOR >= 53 ctx->input_size = ctx->acodec->channels * av_get_bits_per_sample_fmt(ctx->acodec->sample_fmt) / 8; #else ctx->input_size = ctx->acodec->channels * av_get_bits_per_sample_format(ctx->acodec->sample_fmt) / 8; #endif } ctx->duration = mfi->song_length; ctx->samples = mfi->sample_count; ctx->wavhdr = wavhdr; if (wavhdr) make_wav_header(ctx, est_size); return ctx; setup_fail_codec: avcodec_close(ctx->acodec); setup_fail: av_close_input_file(ctx->fmtctx); free(ctx); return NULL; } void transcode_cleanup(struct transcode_ctx *ctx) { if (ctx->apacket.data) av_free_packet(&ctx->apacket); avcodec_close(ctx->acodec); av_close_input_file(ctx->fmtctx); av_free(ctx->abuffer); if (ctx->need_resample) { audio_resample_close(ctx->resample_ctx); av_free(ctx->re_abuffer); } free(ctx); } int transcode_needed(struct evkeyvalq *headers, char *file_codectype) { const char *client_codecs; const char *user_agent; char *codectype; cfg_t *lib; int size; int i; DPRINTF(E_DBG, L_XCODE, "Determining transcoding status for codectype %s\n", file_codectype); lib = cfg_getsec(cfg, "library"); size = cfg_size(lib, "no_transcode"); if (size > 0) { for (i = 0; i < size; i++) { codectype = cfg_getnstr(lib, "no_transcode", i); if (strcmp(file_codectype, codectype) == 0) { DPRINTF(E_DBG, L_XCODE, "Codectype is in no_transcode\n"); return 0; } } } size = cfg_size(lib, "force_transcode"); if (size > 0) { for (i = 0; i < size; i++) { codectype = cfg_getnstr(lib, "force_transcode", i); if (strcmp(file_codectype, codectype) == 0) { DPRINTF(E_DBG, L_XCODE, "Codectype is in force_transcode\n"); return 1; } } } client_codecs = evhttp_find_header(headers, "Accept-Codecs"); if (!client_codecs) { user_agent = evhttp_find_header(headers, "User-Agent"); if (user_agent) { DPRINTF(E_DBG, L_XCODE, "User-Agent: %s\n", user_agent); if (strncmp(user_agent, "iTunes", strlen("iTunes")) == 0) { DPRINTF(E_DBG, L_XCODE, "Client is iTunes\n"); client_codecs = itunes_codecs; } else if (strncmp(user_agent, "QuickTime", strlen("QuickTime")) == 0) { DPRINTF(E_DBG, L_XCODE, "Client is QuickTime, using iTunes codecs\n"); client_codecs = itunes_codecs; } else if (strncmp(user_agent, "Front%20Row", strlen("Front%20Row")) == 0) { DPRINTF(E_DBG, L_XCODE, "Client is Front Row, using iTunes codecs\n"); client_codecs = itunes_codecs; } else if (strncmp(user_agent, "Remote", strlen("Remote")) == 0) { DPRINTF(E_DBG, L_XCODE, "Client is Remote, using iTunes codecs\n"); client_codecs = itunes_codecs; } else if (strncmp(user_agent, "Roku", strlen("Roku")) == 0) { DPRINTF(E_DBG, L_XCODE, "Client is a Roku device\n"); client_codecs = roku_codecs; } else if (strncmp(user_agent, "AppleCoreMedia", strlen("AppleCoreMedia")) == 0) { DPRINTF(E_DBG, L_XCODE, "Client is a AppleCoreMedia, using iTunes codecs\n"); client_codecs = itunes_codecs; } else if (strncmp(user_agent, "Hifidelio", strlen("Hifidelio")) == 0) { DPRINTF(E_DBG, L_XCODE, "Client is a Hifidelio device, allegedly cannot transcode\n"); /* Allegedly can't transcode for Hifidelio because their * HTTP implementation doesn't honour Connection: close. * At least, that's why mt-daapd didn't do it. */ return 0; } } } else DPRINTF(E_DBG, L_XCODE, "Client advertises codecs: %s\n", client_codecs); if (!client_codecs) { DPRINTF(E_DBG, L_XCODE, "Could not identify client, using default codectype set\n"); client_codecs = default_codecs; } if (strstr(client_codecs, file_codectype)) { DPRINTF(E_DBG, L_XCODE, "Codectype supported by client, no transcoding needed\n"); return 0; } DPRINTF(E_DBG, L_XCODE, "Will transcode\n"); return 1; }