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mpv/audio/decode/ad_lavc.c
wm4 f2374f4e4b ad_lavc: use new AVFrame API 
Set refcounted_frames, because in some versions of libavcodec mixing the
new AVFrame API and non-refcounted decoding could cause memory
corruption. Likewise, it's probably still required to unref a frame
before calling the decoder.
2014-03-16 13:19:29 +01:00

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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 <libavutil/common.h>
#include "talloc.h"
#include "config.h"
#include "common/av_common.h"
#include "common/codecs.h"
#include "common/msg.h"
#include "options/options.h"
#include "common/av_opts.h"
#include "ad.h"
#include "audio/fmt-conversion.h"
#include "compat/libav.h"
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
struct mp_audio frame;
bool force_channel_map;
struct demux_packet *packet;
};
static void uninit(struct dec_audio *da);
static int decode_new_packet(struct dec_audio *da);
#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_INTRANGE("threads", ad_lavc_param.threads, 0, 1, 16),
OPT_STRING("o", ad_lavc_param.avopt, 0),
{0}
};
struct pcm_map
{
int tag;
const char *codecs[6]; // {any, 1byte, 2bytes, 3bytes, 4bytes, 8bytes}
};
// NOTE: these are needed to make rawaudio with demux_mkv work.
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", [5] = "pcm_f64le"}},
// 'raw '
{0x20776172, {"pcm_s16be", [1] = "pcm_u8"}},
// 'twos', used by demux_mkv.c internally
{MKTAG('t', 'w', 'o', 's'),
{NULL, "pcm_s8", "pcm_s16be", "pcm_s24be", "pcm_s32be"}},
{-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"}},
{AF_FORMAT_DOUBLE_LE, {"pcm_f64le"}},
{AF_FORMAT_DOUBLE_BE, {"pcm_f64be"}},
{-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;
if (bytes == 8)
bytes = 5; // 64 bit entry
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 <= 5)
dec = entry->codecs[bytes];
if (!dec)
dec = entry->codecs[0];
if (dec)
return dec;
}
}
return NULL;
}
static int setup_format(struct dec_audio *da)
{
struct priv *priv = da->priv;
AVCodecContext *lavc_context = priv->avctx;
struct sh_audio *sh_audio = da->header->audio;
// Note: invalid parameters are rejected by dec_audio.c
int fmt = lavc_context->sample_fmt;
mp_audio_set_format(&da->decoded, af_from_avformat(fmt));
if (!da->decoded.format)
MP_FATAL(da, "unsupported lavc format %s", av_get_sample_fmt_name(fmt));
da->decoded.rate = lavc_context->sample_rate;
if (!da->decoded.rate && sh_audio->wf) {
// If not set, try container samplerate.
// (Maybe this can't happen, and it's an artifact from the past.)
da->decoded.rate = sh_audio->wf->nSamplesPerSec;
MP_WARN(da, "using container rate.\n");
}
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;
}
mp_audio_set_channels(&da->decoded, &lavc_chmap);
return 0;
}
static void set_from_wf(AVCodecContext *avctx, MP_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)
mp_lavc_set_extradata(avctx, wf + 1, wf->cbSize);
}
static int init(struct dec_audio *da, const char *decoder)
{
struct MPOpts *mpopts = da->opts;
struct ad_lavc_param *opts = &mpopts->ad_lavc_param;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
struct sh_stream *sh = da->header;
struct sh_audio *sh_audio = sh->audio;
struct priv *ctx = talloc_zero(NULL, struct priv);
da->priv = ctx;
if (sh_audio->wf && strcmp(decoder, "pcm") == 0) {
decoder = find_pcm_decoder(tag_map, sh->format,
sh_audio->wf->wBitsPerSample);
} else if (sh_audio->wf && strcmp(decoder, "mp-pcm") == 0) {
decoder = find_pcm_decoder(af_map, sh->format, 0);
ctx->force_channel_map = true;
}
lavc_codec = avcodec_find_decoder_by_name(decoder);
if (!lavc_codec) {
MP_ERR(da, "Cannot find codec '%s' in libavcodec...\n", decoder);
uninit(da);
return 0;
}
lavc_context = avcodec_alloc_context3(lavc_codec);
ctx->avctx = lavc_context;
ctx->avframe = av_frame_alloc();
lavc_context->refcounted_frames = 1;
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id;
if (opts->downmix) {
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_ERR(da, "setting AVOptions '%s' failed.\n", opts->avopt);
uninit(da);
return 0;
}
}
lavc_context->codec_tag = sh->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) {
mp_lavc_set_extradata(lavc_context, sh_audio->codecdata,
sh_audio->codecdata_len);
}
if (sh->lav_headers)
mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
mp_set_avcodec_threads(lavc_context, opts->threads);
/* open it */
if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
MP_ERR(da, "Could not open codec.\n");
uninit(da);
return 0;
}
MP_VERBOSE(da, "INFO: libavcodec \"%s\" init OK!\n",
lavc_codec->name);
// Decode at least 1 sample: (to get header filled)
for (int tries = 1; ; tries++) {
int x = decode_new_packet(da);
if (x >= 0 && ctx->frame.samples > 0) {
MP_VERBOSE(da, "Initial decode succeeded after %d packets.\n", tries);
break;
}
if (tries >= 50) {
MP_ERR(da, "initial decode failed\n");
uninit(da);
return 0;
}
}
da->i_bps = lavc_context->bit_rate / 8;
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
da->i_bps = sh_audio->wf->nAvgBytesPerSec;
return 1;
}
static void uninit(struct dec_audio *da)
{
struct priv *ctx = da->priv;
if (!ctx)
return;
AVCodecContext *lavc_context = ctx->avctx;
if (lavc_context) {
if (avcodec_close(lavc_context) < 0)
MP_ERR(da, "Could not close codec.\n");
av_freep(&lavc_context->extradata);
av_freep(&lavc_context);
}
av_frame_free(&ctx->avframe);
}
static int control(struct dec_audio *da, int cmd, void *arg)
{
struct priv *ctx = da->priv;
switch (cmd) {
case ADCTRL_RESET:
avcodec_flush_buffers(ctx->avctx);
ctx->frame.samples = 0;
talloc_free(ctx->packet);
ctx->packet = NULL;
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_new_packet(struct dec_audio *da)
{
struct priv *priv = da->priv;
AVCodecContext *avctx = priv->avctx;
priv->frame.samples = 0;
struct demux_packet *mpkt = priv->packet;
if (!mpkt)
mpkt = demux_read_packet(da->header);
priv->packet = talloc_steal(priv, mpkt);
int in_len = mpkt ? mpkt->len : 0;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, NULL);
// If we don't have a PTS yet, use the first packet PTS we can get.
if (da->pts == MP_NOPTS_VALUE && mpkt && mpkt->pts != MP_NOPTS_VALUE) {
da->pts = mpkt->pts;
da->pts_offset = 0;
}
int got_frame = 0;
av_frame_unref(priv->avframe);
int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
if (mpkt) {
// At least "shorten" decodes sub-frames, instead of the whole packet.
// At least "mpc8" can return 0 and wants the packet again next time.
if (ret >= 0) {
ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads
mpkt->buffer += ret;
mpkt->len -= ret;
mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
}
if (mpkt->len == 0 || ret < 0) {
talloc_free(mpkt);
priv->packet = NULL;
}
// LATM may need many packets to find mux info
if (ret == AVERROR(EAGAIN))
return 0;
}
if (ret < 0) {
MP_VERBOSE(da, "lavc_audio: error\n");
return -1;
}
if (!got_frame)
return mpkt ? 0 : -1; // -1: eof
if (setup_format(da) < 0)
return -1;
priv->frame.samples = priv->avframe->nb_samples;
mp_audio_copy_config(&priv->frame, &da->decoded);
for (int n = 0; n < priv->frame.num_planes; n++)
priv->frame.planes[n] = priv->avframe->data[n];
double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
if (out_pts != MP_NOPTS_VALUE) {
da->pts = out_pts;
da->pts_offset = 0;
}
MP_DBG(da, "Decoded %d -> %d samples\n", in_len,
priv->frame.samples);
return 0;
}
static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxlen)
{
struct priv *priv = da->priv;
if (!priv->frame.samples) {
if (decode_new_packet(da) < 0)
return -1;
}
if (!mp_audio_config_equals(buffer, &priv->frame))
return 0;
buffer->samples = MPMIN(priv->frame.samples, maxlen);
mp_audio_copy(buffer, 0, &priv->frame, 0, buffer->samples);
mp_audio_skip_samples(&priv->frame, buffer->samples);
da->pts_offset += buffer->samples;
return 0;
}
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");
}
const struct ad_functions ad_lavc = {
.name = "lavc",
.add_decoders = add_decoders,
.init = init,
.uninit = uninit,
.control = control,
.decode_audio = decode_audio,
};
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