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mpv/audio/decode/ad_lavc.c

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/*
* 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 "core/mp_msg.h"
#include "core/options.h"
#include "ad_internal.h"
#include "audio/reorder_ch.h"
#include "compat/mpbswap.h"
#include "compat/libav.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
};
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)
{
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;
const char *dll = sh_audio->codec->dll;
if (sh_audio->wf && dll && strcmp(dll, "pcm") == 0) {
if (sh_audio->format == MKTAG('M', 'P', 'a', 'f')) {
// demuxer_rawaudio convenience (abuses wFormatTag)
dll = find_pcm_decoder(af_map, sh_audio->wf->wFormatTag, 0);
} else {
dll = find_pcm_decoder(tag_map, sh_audio->format,
sh_audio->wf->wBitsPerSample);
}
}
if (dll) {
lavc_codec = avcodec_find_decoder_by_name(dll);
if (!lavc_codec) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
"Cannot find codec '%s' in libavcodec...\n", 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->wf && 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->wf && 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;
}
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