From d4bdd0473d6f43132257c9fb3848d829755167a3 Mon Sep 17 00:00:00 2001 From: wm4 Date: Mon, 5 Nov 2012 17:02:04 +0100 Subject: Rename directories, move files (step 1 of 2) (does not compile) Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code. --- libmpcodecs/ad_ffmpeg.c | 413 ------------------------------------------------ 1 file changed, 413 deletions(-) delete mode 100644 libmpcodecs/ad_ffmpeg.c (limited to 'libmpcodecs/ad_ffmpeg.c') diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c deleted file mode 100644 index 2eacfadb8f..0000000000 --- a/libmpcodecs/ad_ffmpeg.c +++ /dev/null @@ -1,413 +0,0 @@ -/* - * 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 "mp_msg.h" -#include "options.h" - -#include "ad_internal.h" -#include "libaf/reorder_ch.h" - -#include "mpbswap.h" - -static const ad_info_t info = -{ - "libavcodec audio decoders", - "ffmpeg", - "", - "", - "", - .print_name = "libavcodec", -}; - -LIBAD_EXTERN(ffmpeg) - -struct priv { - AVCodecContext *avctx; - AVFrame *avframe; - char *output; - char *output_packed; // used by deplanarize to store packed audio samples - int output_left; - int unitsize; - int previous_data_left; // input demuxer packet data -}; - -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) -{ - int sample_format = sh_audio->sample_format; - switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) { - case AV_SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break; - case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break; - case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break; - case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break; - default: - mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n"); - sample_format = AF_FORMAT_UNKNOWN; - } - - 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 == CODEC_ID_AAC - && samplerate == 2 * container_samplerate) - broken_srate = true; - else if (container_samplerate) - samplerate = container_samplerate; - - if (lavc_context->channels != sh_audio->channels || - samplerate != sh_audio->samplerate || - sample_format != sh_audio->sample_format) { - sh_audio->channels = lavc_context->channels; - 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 int init(sh_audio_t *sh_audio) -{ - struct MPOpts *opts = sh_audio->opts; - AVCodecContext *lavc_context; - AVCodec *lavc_codec; - - if (sh_audio->codec->dll) { - lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll); - if (!lavc_codec) { - mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, - "Cannot find codec '%s' in libavcodec...\n", - sh_audio->codec->dll); - return 0; - } - } else if (!sh_audio->libav_codec_id) { - mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "No Libav codec ID known. " - "Generic lavc decoder is not applicable.\n"); - return 0; - } else { - lavc_codec = avcodec_find_decoder(sh_audio->libav_codec_id); - if (!lavc_codec) { - mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Libavcodec has no decoder " - "for this codec\n"); - return 0; - } - } - - sh_audio->codecname = lavc_codec->long_name; - if (!sh_audio->codecname) - sh_audio->codecname = lavc_codec->name; - - struct priv *ctx = talloc_zero(NULL, struct priv); - sh_audio->context = ctx; - lavc_context = avcodec_alloc_context3(lavc_codec); - ctx->avctx = lavc_context; - ctx->avframe = avcodec_alloc_frame(); - - // Always try to set - option only exists for AC3 at the moment - av_opt_set_double(lavc_context, "drc_scale", opts->drc_level, - AV_OPT_SEARCH_CHILDREN); - lavc_context->sample_rate = sh_audio->samplerate; - lavc_context->bit_rate = sh_audio->i_bps * 8; - if (sh_audio->wf) { - lavc_context->channels = sh_audio->wf->nChannels; - lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; - lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; - lavc_context->block_align = sh_audio->wf->nBlockAlign; - lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; - } - lavc_context->request_channels = opts->audio_output_channels; - lavc_context->codec_tag = sh_audio->format; //FOURCC - if (sh_audio->gsh->lavf_codec_tag) - lavc_context->codec_tag = sh_audio->gsh->lavf_codec_tag; - lavc_context->codec_type = AVMEDIA_TYPE_AUDIO; - lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi - - /* alloc extra data */ - if (sh_audio->wf && sh_audio->wf->cbSize > 0) { - lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); - lavc_context->extradata_size = sh_audio->wf->cbSize; - memcpy(lavc_context->extradata, sh_audio->wf + 1, - lavc_context->extradata_size); - } - - // for QDM2 - 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); - } - - /* 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); - - if (sh_audio->format == 0x3343414D) { - // MACE 3:1 - sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet - sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet - } else if (sh_audio->format == 0x3643414D) { - // MACE 6:1 - sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet - sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet - } - - // 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_ffmpeg: 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; - - switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) { - case AV_SAMPLE_FMT_U8: - case AV_SAMPLE_FMT_S16: - case AV_SAMPLE_FMT_S32: - case AV_SAMPLE_FMT_FLT: - break; - default: - uninit(sh_audio); - return 0; - } - return 1; -} - -static void uninit(sh_audio_t *sh) -{ - sh->codecname = NULL; - 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); - ds_clear_parser(sh->ds); - ctx->previous_data_left = 0; - 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; - - 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); - - size_t offset = 0; - unsigned char *output_ptr = priv->output_packed; - unsigned char **src = priv->avframe->data; - - for (size_t s = 0; s < nb_samples; s++) { - for (size_t c = 0; c < channels; c++) { - memcpy(output_ptr, src[c] + offset, bps); - output_ptr += bps; - } - offset += bps; - } - - priv->output = priv->output_packed; -} - -static int decode_new_packet(struct sh_audio *sh) -{ - struct priv *priv = sh->context; - AVCodecContext *avctx = priv->avctx; - double pts = MP_NOPTS_VALUE; - int insize; - bool packet_already_used = priv->previous_data_left; - struct demux_packet *mpkt = ds_get_packet2(sh->ds, - priv->previous_data_left); - unsigned char *start; - if (!mpkt) { - assert(!priv->previous_data_left); - start = NULL; - insize = 0; - ds_parse(sh->ds, &start, &insize, pts, 0); - if (insize <= 0) - return -1; // error or EOF - } else { - assert(mpkt->len >= priv->previous_data_left); - if (!priv->previous_data_left) { - priv->previous_data_left = mpkt->len; - pts = mpkt->pts; - } - insize = priv->previous_data_left; - start = mpkt->buffer + mpkt->len - priv->previous_data_left; - int consumed = ds_parse(sh->ds, &start, &insize, pts, 0); - priv->previous_data_left -= consumed; - priv->previous_data_left = FFMAX(priv->previous_data_left, 0); - } - - AVPacket pkt; - av_init_packet(&pkt); - pkt.data = start; - pkt.size = insize; - if (mpkt && mpkt->avpacket) { - pkt.side_data = mpkt->avpacket->side_data; - pkt.side_data_elems = mpkt->avpacket->side_data_elems; - } - if (pts != MP_NOPTS_VALUE && !packet_already_used) { - sh->pts = 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; - } - // The "insize >= ret" test is sanity check against decoder overreads - if (!sh->parser && insize >= ret) - priv->previous_data_left = insize - ret; - 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", insize, - 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 (avctx->channels >= 5) { - int samplesize = av_get_bytes_per_sample(avctx->sample_fmt); - reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, - AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, - avctx->channels, - size / samplesize, samplesize); - } - if (len < 0) - len = size; - else - len += size; - buf += size; - maxlen -= size; - sh_audio->pts_bytes += size; - } - return len; -} -- cgit v1.2.3