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-rw-r--r--audio/decode/ad_lavc.c413
1 files changed, 413 insertions, 0 deletions
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;
+}