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authorUoti Urpala <uau@mplayer2.org>2012-04-19 01:26:56 +0300
committerUoti Urpala <uau@mplayer2.org>2012-04-19 01:42:30 +0300
commitdeffd15a056f98bbe1bc34ce87dea0b4a530d61d (patch)
tree1fa363205a2ff7b147bbfd956ace50824642a48e
parentb9fefc87c01bbf464fb74981313efa77e03735f6 (diff)
downloadmpv-deffd15a056f98bbe1bc34ce87dea0b4a530d61d.tar.bz2
mpv-deffd15a056f98bbe1bc34ce87dea0b4a530d61d.tar.xz
ad_ffmpeg: switch to avcodec_decode_audio4()
Switch libavcodec audio decoding from avcodec_decode_audio3() to avcodec_decode_audio4(). Instead of decoding directly to the output buffer, the data is now copied from the libavcodec output packet, adding an extra memory copy (optimizing this would require some interface changes). After libavcodec added avcodec_decode_audio4() earlier, it dropped support for splitting large audio packets into output chunks of size AVCODEC_MAX_AUDIO_FRAME_SIZE or less. This caused a regression with the previous API: audio files with huge packets could fail to decode, as libavcodec refused to write into the AVCODEC_MAX_AUDIO_FRAME_SIZE buffer provided by mplayer2. This occurrend mainly with some lossless audio formats. This commit restores support for those files; there are now no fixed limits on packet size.
-rw-r--r--libmpcodecs/ad_ffmpeg.c181
1 files changed, 112 insertions, 69 deletions
diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
index 4a5062ba00..0bfc5e5f0a 100644
--- a/libmpcodecs/ad_ffmpeg.c
+++ b/libmpcodecs/ad_ffmpeg.c
@@ -49,12 +49,15 @@ LIBAD_EXTERN(ffmpeg)
struct priv {
AVCodecContext *avctx;
- int previous_data_left;
+ AVFrame *avframe;
+ char *output;
+ int output_left;
+ int unitsize;
+ int previous_data_left; // input demuxer packet data
};
static int preinit(sh_audio_t *sh)
{
- sh->audio_out_minsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
return 1;
}
@@ -74,6 +77,7 @@ static int setup_format(sh_audio_t *sh_audio,
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;
@@ -122,6 +126,7 @@ static int init(sh_audio_t *sh_audio)
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,
@@ -223,6 +228,7 @@ static void uninit(sh_audio_t *sh)
av_freep(&lavc_context->extradata);
av_freep(&lavc_context);
}
+ av_free(ctx->avframe);
talloc_free(ctx);
sh->context = NULL;
}
@@ -235,86 +241,123 @@ static int control(sh_audio_t *sh, int cmd, void *arg, ...)
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 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;
+ }
+
+ 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;
+ }
+ if (!sh->parser)
+ priv->previous_data_left += insize - ret;
+ if (!got_frame)
+ return 0;
+ /* An error is reported later from output format checking, but make
+ * sure we don't crash by overreading first plane. */
+ if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1)
+ 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;
+ 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 *ctx = sh_audio->context;
- AVCodecContext *avctx = ctx->avctx;
+ struct priv *priv = sh_audio->context;
+ AVCodecContext *avctx = priv->avctx;
- unsigned char *start = NULL;
- int y, len = -1;
+ int len = -1;
while (len < minlen) {
- AVPacket pkt;
- int len2 = maxlen;
- double pts = MP_NOPTS_VALUE;
- int x;
- bool packet_already_used = ctx->previous_data_left;
- struct demux_packet *mpkt = ds_get_packet2(sh_audio->ds,
- ctx->previous_data_left);
- if (!mpkt) {
- assert(!ctx->previous_data_left);
- start = NULL;
- x = 0;
- ds_parse(sh_audio->ds, &start, &x, pts, 0);
- if (x <= 0)
- break; // error
- } else {
- assert(mpkt->len >= ctx->previous_data_left);
- if (!ctx->previous_data_left) {
- ctx->previous_data_left = mpkt->len;
- pts = mpkt->pts;
- }
- x = ctx->previous_data_left;
- start = mpkt->buffer + mpkt->len - ctx->previous_data_left;
- int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
- ctx->previous_data_left -= consumed;
- }
- av_init_packet(&pkt);
- pkt.data = start;
- pkt.size = x;
- 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_audio->pts = pts;
- sh_audio->pts_bytes = 0;
- }
- y = avcodec_decode_audio3(avctx, (int16_t *)buf, &len2, &pkt);
- // LATM may need many packets to find mux info
- if (y == AVERROR(EAGAIN))
+ if (!priv->output_left) {
+ if (decode_new_packet(sh_audio) < 0)
+ break;
continue;
- if (y < 0) {
- mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
- break;
- }
- if (!sh_audio->parser)
- ctx->previous_data_left += x - y;
- if (len2 > 0) {
- 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,
- len2 / samplesize, samplesize);
- }
- if (len < 0)
- len = len2;
- else
- len += len2;
- buf += len2;
- maxlen -= len2;
- sh_audio->pts_bytes += len2;
}
- mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", y, len2);
-
if (setup_format(sh_audio, avctx))
- break;
+ 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;
}