diff options
Diffstat (limited to 'audio/decode')
| -rw-r--r-- | audio/decode/ad.c | 50 | ||||
| -rw-r--r-- | audio/decode/ad.h | 54 | ||||
| -rw-r--r-- | audio/decode/ad_dvdpcm.c | 162 | ||||
| -rw-r--r-- | audio/decode/ad_internal.h | 46 | ||||
| -rw-r--r-- | audio/decode/ad_lavc.c | 413 | ||||
| -rw-r--r-- | audio/decode/ad_mpg123.c | 489 | ||||
| -rw-r--r-- | audio/decode/ad_pcm.c | 220 | ||||
| -rw-r--r-- | audio/decode/ad_spdif.c | 310 | ||||
| -rw-r--r-- | audio/decode/dec_audio.c | 462 | ||||
| -rw-r--r-- | audio/decode/dec_audio.h | 38 |
10 files changed, 2244 insertions, 0 deletions
diff --git a/audio/decode/ad.c b/audio/decode/ad.c new file mode 100644 index 0000000000..93cebed86d --- /dev/null +++ b/audio/decode/ad.c @@ -0,0 +1,50 @@ +/* + * audio decoder interface + * + * 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 <stdio.h> +#include <stdlib.h> +#include <string.h> + +#include "config.h" + +#include "stream/stream.h" +#include "libmpdemux/demuxer.h" +#include "libmpdemux/stheader.h" +#include "ad.h" + +/* Missed vorbis, mad, dshow */ + +extern const ad_functions_t mpcodecs_ad_mpg123; +extern const ad_functions_t mpcodecs_ad_ffmpeg; +extern const ad_functions_t mpcodecs_ad_pcm; +extern const ad_functions_t mpcodecs_ad_dvdpcm; +extern const ad_functions_t mpcodecs_ad_spdif; + +const ad_functions_t * const mpcodecs_ad_drivers[] = +{ +#ifdef CONFIG_MPG123 + &mpcodecs_ad_mpg123, +#endif + &mpcodecs_ad_ffmpeg, + &mpcodecs_ad_pcm, + &mpcodecs_ad_dvdpcm, + &mpcodecs_ad_spdif, + NULL +}; diff --git a/audio/decode/ad.h b/audio/decode/ad.h new file mode 100644 index 0000000000..5396085d04 --- /dev/null +++ b/audio/decode/ad.h @@ -0,0 +1,54 @@ +/* + * 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. + */ + +#ifndef MPLAYER_AD_H +#define MPLAYER_AD_H + +#include "mpc_info.h" +#include "libmpdemux/stheader.h" + +typedef struct mp_codec_info ad_info_t; + +/* interface of video decoder drivers */ +typedef struct ad_functions +{ + const ad_info_t *info; + int (*preinit)(sh_audio_t *sh); + int (*init)(sh_audio_t *sh); + void (*uninit)(sh_audio_t *sh); + int (*control)(sh_audio_t *sh,int cmd,void* arg, ...); + int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen, + int maxlen); +} ad_functions_t; + +// NULL terminated array of all drivers +extern const ad_functions_t * const mpcodecs_ad_drivers[]; + +// fallback if ADCTRL_RESYNC not implemented: sh_audio->a_in_buffer_len=0; +#define ADCTRL_RESYNC_STREAM 1 // resync, called after seeking + +// fallback if ADCTRL_SKIP not implemented: ds_fill_buffer(sh_audio->ds); +#define ADCTRL_SKIP_FRAME 2 // skip block/frame, called while seeking + +// fallback if ADCTRL_QUERY_FORMAT not implemented: sh_audio->sample_format +#define ADCTRL_QUERY_FORMAT 3 // test for availabilty of a format + +// fallback: use hw mixer in libao +#define ADCTRL_SET_VOLUME 4 // not used at the moment + +#endif /* MPLAYER_AD_H */ diff --git a/audio/decode/ad_dvdpcm.c b/audio/decode/ad_dvdpcm.c new file mode 100644 index 0000000000..41f6a1426d --- /dev/null +++ b/audio/decode/ad_dvdpcm.c @@ -0,0 +1,162 @@ +/* + * 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 <stdio.h> +#include <stdlib.h> +#include <unistd.h> + +#include "config.h" +#include "mp_msg.h" +#include "ad_internal.h" + +static const ad_info_t info = +{ + "Uncompressed DVD/VOB LPCM audio decoder", + "dvdpcm", + "Nick Kurshev", + "A'rpi", + "" +}; + +LIBAD_EXTERN(dvdpcm) + +static int init(sh_audio_t *sh) +{ +/* DVD PCM Audio:*/ + sh->i_bps = 0; + if(sh->codecdata_len==3){ + // we have LPCM header: + unsigned char h=sh->codecdata[1]; + sh->channels=1+(h&7); + switch((h>>4)&3){ + case 0: sh->samplerate=48000;break; + case 1: sh->samplerate=96000;break; + case 2: sh->samplerate=44100;break; + case 3: sh->samplerate=32000;break; + } + switch ((h >> 6) & 3) { + case 0: + sh->sample_format = AF_FORMAT_S16_BE; + sh->samplesize = 2; + break; + case 1: + mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n"); + sh->i_bps = sh->channels * sh->samplerate * 5 / 2; + case 2: + sh->sample_format = AF_FORMAT_S24_BE; + sh->samplesize = 3; + break; + default: + sh->sample_format = AF_FORMAT_S16_BE; + sh->samplesize = 2; + } + } else { + // use defaults: + sh->channels=2; + sh->samplerate=48000; + sh->sample_format = AF_FORMAT_S16_BE; + sh->samplesize = 2; + } + if (!sh->i_bps) + sh->i_bps = sh->samplesize * sh->channels * sh->samplerate; + return 1; +} + +static int preinit(sh_audio_t *sh) +{ + sh->audio_out_minsize=2048; + return 1; +} + +static void uninit(sh_audio_t *sh) +{ +} + +static int control(sh_audio_t *sh,int cmd,void* arg, ...) +{ + int skip; + switch(cmd) + { + case ADCTRL_SKIP_FRAME: + skip=sh->i_bps/16; + skip=skip&(~3); + demux_read_data(sh->ds,NULL,skip); + return CONTROL_TRUE; + } + return CONTROL_UNKNOWN; +} + +static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) +{ + int j,len; + if (sh_audio->samplesize == 3) { + if (((sh_audio->codecdata[1] >> 6) & 3) == 1) { + // 20 bit + // not sure if the "& 0xf0" and "<< 4" are the right way around + // can somebody clarify? + for (j = 0; j < minlen; j += 12) { + char tmp[10]; + len = demux_read_data(sh_audio->ds, tmp, 10); + if (len < 10) break; + // first sample + buf[j + 0] = tmp[0]; + buf[j + 1] = tmp[1]; + buf[j + 2] = tmp[8] & 0xf0; + // second sample + buf[j + 3] = tmp[2]; + buf[j + 4] = tmp[3]; + buf[j + 5] = tmp[8] << 4; + // third sample + buf[j + 6] = tmp[4]; + buf[j + 7] = tmp[5]; + buf[j + 8] = tmp[9] & 0xf0; + // fourth sample + buf[j + 9] = tmp[6]; + buf[j + 10] = tmp[7]; + buf[j + 11] = tmp[9] << 4; + } + len = j; + } else { + // 24 bit + for (j = 0; j < minlen; j += 12) { + char tmp[12]; + len = demux_read_data(sh_audio->ds, tmp, 12); + if (len < 12) break; + // first sample + buf[j + 0] = tmp[0]; + buf[j + 1] = tmp[1]; + buf[j + 2] = tmp[8]; + // second sample + buf[j + 3] = tmp[2]; + buf[j + 4] = tmp[3]; + buf[j + 5] = tmp[9]; + // third sample + buf[j + 6] = tmp[4]; + buf[j + 7] = tmp[5]; + buf[j + 8] = tmp[10]; + // fourth sample + buf[j + 9] = tmp[6]; + buf[j + 10] = tmp[7]; + buf[j + 11] = tmp[11]; + } + len = j; + } + } else + len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3)); + return len; +} diff --git a/audio/decode/ad_internal.h b/audio/decode/ad_internal.h new file mode 100644 index 0000000000..4cffc95126 --- /dev/null +++ b/audio/decode/ad_internal.h @@ -0,0 +1,46 @@ +/* + * 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. + */ + +#ifndef MPLAYER_AD_INTERNAL_H +#define MPLAYER_AD_INTERNAL_H + +#include "codec-cfg.h" +#include "libaf/format.h" + +#include "stream/stream.h" +#include "libmpdemux/demuxer.h" +#include "libmpdemux/stheader.h" + +#include "ad.h" + +static int init(sh_audio_t *sh); +static int preinit(sh_audio_t *sh); +static void uninit(sh_audio_t *sh); +static int control(sh_audio_t *sh,int cmd,void* arg, ...); +static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen); + +#define LIBAD_EXTERN(x) const ad_functions_t mpcodecs_ad_##x = {\ + &info,\ + preinit,\ + init,\ + uninit,\ + control,\ + decode_audio\ +}; + +#endif /* MPLAYER_AD_INTERNAL_H */ diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c new file mode 100644 index 0000000000..2eacfadb8f --- /dev/null +++ b/audio/decode/ad_lavc.c @@ -0,0 +1,413 @@ +/* + * 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 <stdio.h> +#include <stdlib.h> +#include <unistd.h> +#include <stdbool.h> +#include <assert.h> + +#include <libavcodec/avcodec.h> +#include <libavutil/opt.h> + +#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; +} diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c new file mode 100644 index 0000000000..a3ce2cdcf6 --- /dev/null +++ b/audio/decode/ad_mpg123.c @@ -0,0 +1,489 @@ +/* + * MPEG 1.0/2.0/2.5 audio layer I, II, III decoding with libmpg123 + * + * Copyright (C) 2010-2012 Thomas Orgis <thomas@orgis.org> + * + * 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 <stdio.h> +#include <stdlib.h> +#include <unistd.h> + +#include "config.h" + +#include "ad_internal.h" + +static const ad_info_t info = { + "MPEG 1.0/2.0/2.5 layers I, II, III", + "mpg123", + "Thomas Orgis", + "mpg123.org", + "High-performance decoder using libmpg123." +}; + +LIBAD_EXTERN(mpg123) + +/* Reducing the ifdeffery to two main variants: + * 1. most compatible to any libmpg123 version + * 2. fastest variant with recent libmpg123 (>=1.14) + * Running variant 2 on older libmpg123 versions may work in + * principle, but is not supported. + * So, please leave the check for MPG123_API_VERSION there, m-kay? + */ +#include <mpg123.h> + +/* Enable faster mode of operation with newer libmpg123, avoiding + * unnecessary memcpy() calls. */ +#if (defined MPG123_API_VERSION) && (MPG123_API_VERSION >= 33) +#define AD_MPG123_FRAMEWISE +#endif + +/* Switch for updating bitrate info of VBR files. Not essential. */ +#define AD_MPG123_MEAN_BITRATE + +/* Funny thing, that. I assume I shall use it for selecting mpg123 channels. + * Please correct me if I guessed wrong. */ +extern int fakemono; + +struct ad_mpg123_context { + mpg123_handle *handle; +#ifdef AD_MPG123_MEAN_BITRATE + /* Running mean for bit rate, stream length estimation. */ + float mean_rate; + unsigned int mean_count; + /* Time delay for updates. */ + short delay; +#endif + /* If the stream is actually VBR. */ + char vbr; +}; + +/* This initializes libmpg123 and prepares the handle, including funky + * parameters. */ +static int preinit(sh_audio_t *sh) +{ + int err, flag; + struct ad_mpg123_context *con; + /* Assumption: You always call preinit + init + uninit, on every file. + * But you stop at preinit in case it fails. + * If that is not true, one must ensure not to call mpg123_init / exit + * twice in a row. */ + if (mpg123_init() != MPG123_OK) + return 0; + + sh->context = malloc(sizeof(struct ad_mpg123_context)); + con = sh->context; + /* Auto-choice of optimized decoder (first argument NULL). */ + con->handle = mpg123_new(NULL, &err); + if (!con->handle) + goto bad_end; + + /* Guessing here: Default value triggers forced upmix of mono to stereo. */ + flag = fakemono == 0 ? MPG123_FORCE_STEREO : + fakemono == 1 ? MPG123_MONO_LEFT : + fakemono == 2 ? MPG123_MONO_RIGHT : 0; + if (mpg123_param(con->handle, MPG123_ADD_FLAGS, flag, 0.0) != MPG123_OK) + goto bad_end; + + /* Basic settings. + * Don't spill messages, enable better resync with non-seekable streams. + * Give both flags individually without error checking to keep going with + * old libmpg123. Generally, it is not fatal if the flags are not + * honored */ + mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0.0); + /* Do not bail out on malformed streams at all. + * MPlayer does not handle a decoder throwing the towel on crappy input. */ + mpg123_param(con->handle, MPG123_RESYNC_LIMIT, -1, 0.0); + + /* Open decisions: Configure libmpg123 to force encoding (or stay open about + * library builds that support only float or int32 output), (de)configure + * gapless decoding (won't work with seeking in MPlayer, though). + * Don't forget to eventually enable ReplayGain/RVA support, too. + * Let's try to run with the default for now. */ + + /* That would produce floating point output. + * You can get 32 and 24 bit ints, even 8 bit via format matrix. */ + /* mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_FORCE_FLOAT, 0.); */ + + /* Example for RVA choice (available since libmpg123 1.0.0): + mpg123_param(con->handle, MPG123_RVA, MPG123_RVA_MIX, 0.0) */ + +#ifdef AD_MPG123_FRAMEWISE + /* Prevent funky automatic resampling. + * This way, we can be sure that one frame will never produce + * more than 1152 stereo samples. */ + mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.); +#else + /* Older mpg123 is vulnerable to concatenated streams when gapless cutting + * is enabled (will only play the jingle of a badly constructed radio + * stream). The versions using framewise decoding are fine with that. */ + mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0.); +#endif + + return 1; + + bad_end: + if (!con->handle) + mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n", + mpg123_plain_strerror(err)); + else + mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n", + mpg123_strerror(con->handle)); + + if (con->handle) + mpg123_delete(con->handle); + mpg123_exit(); + free(sh->context); + sh->context = NULL; + return 0; +} + +/* Compute bitrate from frame size. */ +static int compute_bitrate(struct mpg123_frameinfo *i) +{ + static const int samples_per_frame[4][4] = { + {-1, 384, 1152, 1152}, /* MPEG 1 */ + {-1, 384, 1152, 576}, /* MPEG 2 */ + {-1, 384, 1152, 576}, /* MPEG 2.5 */ + {-1, -1, -1, -1}, /* Unknown */ + }; + return (int) ((i->framesize + 4) * 8 * i->rate * 0.001 / + samples_per_frame[i->version][i->layer] + 0.5); +} + +/* Opted against the header printout from old mp3lib, too much + * irrelevant info. This is modelled after the mpg123 app's + * standard output line. + * If more verbosity is demanded, one can add more detail and + * also throw in ID3v2 info which libmpg123 collects anyway. */ +static void print_header_compact(struct mpg123_frameinfo *i) +{ + static const char *smodes[5] = { + "stereo", "joint-stereo", "dual-channel", "mono", "invalid" + }; + static const char *layers[4] = { + "Unknown", "I", "II", "III" + }; + static const char *versions[4] = { + "1.0", "2.0", "2.5", "x.x" + }; + + mp_msg(MSGT_DECAUDIO, MSGL_V, "MPEG %s layer %s, ", + versions[i->version], layers[i->layer]); + switch (i->vbr) { + case MPG123_CBR: + if (i->bitrate) + mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s", i->bitrate); + else + mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s (free format)", + compute_bitrate(i)); + break; + case MPG123_VBR: + mp_msg(MSGT_DECAUDIO, MSGL_V, "VBR"); + break; + case MPG123_ABR: + mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s ABR", i->abr_rate); + break; + default: + mp_msg(MSGT_DECAUDIO, MSGL_V, "???"); + } + mp_msg(MSGT_DECAUDIO, MSGL_V, ", %ld Hz %s\n", i->rate, + smodes[i->mode]); |