/* * This file is part of MPlayer. * * MPlayer 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. * * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include #include #include #include #include #include #include #include "talloc.h" #include "config.h" #include "core/av_common.h" #include "core/codecs.h" #include "core/mp_msg.h" #include "core/options.h" #include "core/av_opts.h" #include "ad_internal.h" #include "audio/reorder_ch.h" #include "audio/fmt-conversion.h" #include "compat/mpbswap.h" #include "compat/libav.h" LIBAD_EXTERN(lavc) struct priv { AVCodecContext *avctx; AVFrame *avframe; uint8_t *output; uint8_t *output_packed; // used by deplanarize to store packed audio samples int output_left; int unitsize; bool force_channel_map; }; #define OPT_BASE_STRUCT struct MPOpts const m_option_t ad_lavc_decode_opts_conf[] = { OPT_FLOATRANGE("ac3drc", ad_lavc_param.ac3drc, 0, 0, 2), OPT_FLAG("downmix", ad_lavc_param.downmix, 0), OPT_STRING("o", ad_lavc_param.avopt, 0), {0} }; struct pcm_map { int tag; const char *codecs[5]; // {any, 1byte, 2bytes, 3bytes, 4bytes} }; // NOTE: some of these are needed to make rawaudio with demux_mkv and others // work. ffmpeg does similar mapping internally, not part of the public // API. Some of these might be dead leftovers for demux_mov support. static const struct pcm_map tag_map[] = { // Microsoft PCM {0x0, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}}, {0x1, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}}, // MS PCM, Extended {0xfffe, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}}, // IEEE float {0x3, {"pcm_f32le"}}, // 'raw ' {0x20776172, {"pcm_s16be", [1] = "pcm_u8"}}, // 'twos'/'sowt' {0x736F7774, {"pcm_s16be", [1] = "pcm_s8"}}, {0x74776F73, {"pcm_s16be", [1] = "pcm_s8"}}, // 'fl32'/'FL32' {0x32336c66, {"pcm_f32be"}}, {0x32334C46, {"pcm_f32be"}}, // '23lf'/'lpcm' {0x666c3332, {"pcm_f32le"}}, {0x6D63706C, {"pcm_f32le"}}, // 'in24', bigendian int24 {0x34326e69, {"pcm_s24be"}}, // '42ni', little endian int24, MPlayer internal fourCC {0x696e3234, {"pcm_s24le"}}, // 'in32', bigendian int32 {0x32336e69, {"pcm_s32be"}}, // '23ni', little endian int32, MPlayer internal fourCC {0x696e3332, {"pcm_s32le"}}, {-1}, }; // For demux_rawaudio.c; needed because ffmpeg doesn't have these sample // formats natively. static const struct pcm_map af_map[] = { {AF_FORMAT_U8, {"pcm_u8"}}, {AF_FORMAT_S8, {"pcm_u8"}}, {AF_FORMAT_U16_LE, {"pcm_u16le"}}, {AF_FORMAT_U16_BE, {"pcm_u16be"}}, {AF_FORMAT_S16_LE, {"pcm_s16le"}}, {AF_FORMAT_S16_BE, {"pcm_s16be"}}, {AF_FORMAT_U24_LE, {"pcm_u24le"}}, {AF_FORMAT_U24_BE, {"pcm_u24be"}}, {AF_FORMAT_S24_LE, {"pcm_s24le"}}, {AF_FORMAT_S24_BE, {"pcm_s24be"}}, {AF_FORMAT_U32_LE, {"pcm_u32le"}}, {AF_FORMAT_U32_BE, {"pcm_u32be"}}, {AF_FORMAT_S32_LE, {"pcm_s32le"}}, {AF_FORMAT_S32_BE, {"pcm_s32be"}}, {AF_FORMAT_FLOAT_LE, {"pcm_f32le"}}, {AF_FORMAT_FLOAT_BE, {"pcm_f32be"}}, {-1}, }; static const char *find_pcm_decoder(const struct pcm_map *map, int format, int bits_per_sample) { int bytes = (bits_per_sample + 7) / 8; for (int n = 0; map[n].tag != -1; n++) { const struct pcm_map *entry = &map[n]; if (entry->tag == format) { const char *dec = NULL; if (bytes >= 1 && bytes <= 4) dec = entry->codecs[bytes]; if (!dec) dec = entry->codecs[0]; if (dec) return dec; } } return NULL; } static int preinit(sh_audio_t *sh) { return 1; } /* Prefer playing audio with the samplerate given in container data * if available, but take number the number of channels and sample format * from the codec, since if the codec isn't using the correct values for * those everything breaks anyway. */ static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context) { struct priv *priv = sh_audio->context; int sample_format = af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt)); bool broken_srate = false; int samplerate = lavc_context->sample_rate; int container_samplerate = sh_audio->container_out_samplerate; if (!container_samplerate && sh_audio->wf) container_samplerate = sh_audio->wf->nSamplesPerSec; if (lavc_context->codec_id == AV_CODEC_ID_AAC && samplerate == 2 * container_samplerate) broken_srate = true; else if (container_samplerate) samplerate = container_samplerate; struct mp_chmap lavc_chmap; mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout); // No channel layout or layout disagrees with channel count if (lavc_chmap.num != lavc_context->channels) mp_chmap_from_channels(&lavc_chmap, lavc_context->channels); if (priv->force_channel_map) { if (lavc_chmap.num == sh_audio->channels.num) lavc_chmap = sh_audio->channels; } if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) || samplerate != sh_audio->samplerate || sample_format != sh_audio->sample_format) { sh_audio->channels = lavc_chmap; sh_audio->samplerate = samplerate; sh_audio->sample_format = sample_format; sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8; if (broken_srate) mp_msg(MSGT_DECAUDIO, MSGL_WARN, "Ignoring broken container sample rate for AAC with SBR\n"); return 1; } return 0; } static void set_from_wf(AVCodecContext *avctx, WAVEFORMATEX *wf) { avctx->channels = wf->nChannels; avctx->sample_rate = wf->nSamplesPerSec; avctx->bit_rate = wf->nAvgBytesPerSec * 8; avctx->block_align = wf->nBlockAlign; avctx->bits_per_coded_sample = wf->wBitsPerSample; if (wf->cbSize > 0) { avctx->extradata = av_mallocz(wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); avctx->extradata_size = wf->cbSize; memcpy(avctx->extradata, wf + 1, avctx->extradata_size); } } static int init(sh_audio_t *sh_audio, const char *decoder) { struct MPOpts *mpopts = sh_audio->opts; struct ad_lavc_param *opts = &mpopts->ad_lavc_param; AVCodecContext *lavc_context; AVCodec *lavc_codec; struct priv *ctx = talloc_zero(NULL, struct priv); sh_audio->context = ctx; if (sh_audio->wf && strcmp(decoder, "pcm") == 0) { decoder = find_pcm_decoder(tag_map, sh_audio->format, sh_audio->wf->wBitsPerSample); } else if (sh_audio->wf && strcmp(decoder, "mp-pcm") == 0) { decoder = find_pcm_decoder(af_map, sh_audio->format, 0); ctx->force_channel_map = true; } lavc_codec = avcodec_find_decoder_by_name(decoder); if (!lavc_codec) { mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Cannot find codec '%s' in libavcodec...\n", decoder); uninit(sh_audio); return 0; } lavc_context = avcodec_alloc_context3(lavc_codec); ctx->avctx = lavc_context; ctx->avframe = avcodec_alloc_frame(); lavc_context->codec_type = AVMEDIA_TYPE_AUDIO; lavc_context->codec_id = lavc_codec->id; if (opts->downmix) { lavc_context->request_channels = mpopts->audio_output_channels.num; lavc_context->request_channel_layout = mp_chmap_to_lavc(&mpopts->audio_output_channels); } // Always try to set - option only exists for AC3 at the moment av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc, AV_OPT_SEARCH_CHILDREN); if (opts->avopt) { if (parse_avopts(lavc_context, opts->avopt) < 0) { mp_msg(MSGT_DECVIDEO, MSGL_ERR, "ad_lavc: setting AVOptions '%s' failed.\n", opts->avopt); uninit(sh_audio); return 0; } } lavc_context->codec_tag = sh_audio->format; lavc_context->sample_rate = sh_audio->samplerate; lavc_context->bit_rate = sh_audio->i_bps * 8; lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels); if (sh_audio->wf) set_from_wf(lavc_context, sh_audio->wf); // demux_mkv, demux_mpg if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) { lavc_context->extradata = av_malloc(sh_audio->codecdata_len + FF_INPUT_BUFFER_PADDING_SIZE); lavc_context->extradata_size = sh_audio->codecdata_len; memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, lavc_context->extradata_size); } if (sh_audio->gsh->lav_headers) mp_copy_lav_codec_headers(lavc_context, sh_audio->gsh->lav_headers); /* open it */ if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) { mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n"); uninit(sh_audio); return 0; } mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); // Decode at least 1 byte: (to get header filled) for (int tries = 0;;) { int x = decode_audio(sh_audio, sh_audio->a_buffer, 1, sh_audio->a_buffer_size); if (x > 0) { sh_audio->a_buffer_len = x; break; } if (++tries >= 5) { mp_msg(MSGT_DECAUDIO, MSGL_ERR, "ad_lavc: initial decode failed\n"); uninit(sh_audio); return 0; } } sh_audio->i_bps = lavc_context->bit_rate / 8; if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec; int af_sample_fmt = af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt)); if (af_sample_fmt == AF_FORMAT_UNKNOWN) { uninit(sh_audio); return 0; } return 1; } static void uninit(sh_audio_t *sh) { struct priv *ctx = sh->context; if (!ctx) return; AVCodecContext *lavc_context = ctx->avctx; if (lavc_context) { if (avcodec_close(lavc_context) < 0) mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n"); av_freep(&lavc_context->extradata); av_freep(&lavc_context); } avcodec_free_frame(&ctx->avframe); talloc_free(ctx); sh->context = NULL; } static int control(sh_audio_t *sh, int cmd, void *arg) { struct priv *ctx = sh->context; switch (cmd) { case ADCTRL_RESYNC_STREAM: avcodec_flush_buffers(ctx->avctx); ctx->output_left = 0; return CONTROL_TRUE; } return CONTROL_UNKNOWN; } static av_always_inline void deplanarize(struct sh_audio *sh) { struct priv *priv = sh->context; uint8_t **planes = priv->avframe->extended_data; size_t bps = av_get_bytes_per_sample(priv->avctx->sample_fmt); size_t nb_samples = priv->avframe->nb_samples; size_t channels = priv->avctx->channels; size_t size = bps * nb_samples * channels; if (talloc_get_size(priv->output_packed) != size) priv->output_packed = talloc_realloc_size(priv, priv->output_packed, size); reorder_to_packed(priv->output_packed, planes, bps, channels, nb_samples); priv->output = priv->output_packed; } static int decode_new_packet(struct sh_audio *sh) { struct priv *priv = sh->context; AVCodecContext *avctx = priv->avctx; struct demux_packet *mpkt = ds_get_packet2(sh->ds, false); if (!mpkt) return -1; // error or EOF AVPacket pkt; mp_set_av_packet(&pkt, mpkt); if (mpkt->pts != MP_NOPTS_VALUE) { sh->pts = mpkt->pts; sh->pts_bytes = 0; } int got_frame = 0; int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt); // LATM may need many packets to find mux info if (ret == AVERROR(EAGAIN)) return 0; if (ret < 0) { mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n"); return -1; } if (!got_frame) return 0; uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) * avctx->channels; if (unitsize > 100000) abort(); priv->unitsize = unitsize; uint64_t output_left = unitsize * priv->avframe->nb_samples; if (output_left > 500000000) abort(); priv->output_left = output_left; if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) { deplanarize(sh); } else { priv->output = priv->avframe->data[0]; } mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", mpkt->len, priv->output_left); return 0; } static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen, int maxlen) { struct priv *priv = sh_audio->context; AVCodecContext *avctx = priv->avctx; int len = -1; while (len < minlen) { if (!priv->output_left) { if (decode_new_packet(sh_audio) < 0) break; continue; } if (setup_format(sh_audio, avctx)) return len; int size = (minlen - len + priv->unitsize - 1); size -= size % priv->unitsize; size = FFMIN(size, priv->output_left); if (size > maxlen) abort(); memcpy(buf, priv->output, size); priv->output += size; priv->output_left -= size; if (len < 0) len = size; else len += size; buf += size; maxlen -= size; sh_audio->pts_bytes += size; } return len; } static void add_decoders(struct mp_decoder_list *list) { mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO); mp_add_decoder(list, "lavc", "pcm", "pcm", "Raw PCM"); mp_add_decoder(list, "lavc", "mp-pcm", "mp-pcm", "Raw PCM"); }