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-rw-r--r--audio/decode/ad_lavc.c34
-rw-r--r--audio/out/ao_lavc.c77
2 files changed, 22 insertions, 89 deletions
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index c42c430850..2e0cade8c5 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -63,12 +63,10 @@ const m_option_t ad_lavc_decode_opts_conf[] = {
struct pcm_map
{
int tag;
- const char *codecs[5]; // {any, 1byte, 2bytes, 3bytes, 4bytes}
+ const char *codecs[6]; // {any, 1byte, 2bytes, 3bytes, 4bytes, 8bytes}
};
-// 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.
+// 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"}},
@@ -76,26 +74,12 @@ static const struct pcm_map tag_map[] = {
// MS PCM, Extended
{0xfffe, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
// IEEE float
- {0x3, {"pcm_f32le"}},
+ {0x3, {"pcm_f32le", [5] = "pcm_f64le"}},
// '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"}},
+ // 'twos', used by demux_mkv.c internally
+ {MKTAG('t', 'w', 'o', 's'),
+ {NULL, "pcm_s8", "pcm_s16be", "pcm_s24be", "pcm_s32be"}},
{-1},
};
@@ -118,6 +102,8 @@ static const struct pcm_map af_map[] = {
{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},
};
@@ -125,11 +111,13 @@ 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 <= 4)
+ if (bytes >= 1 && bytes <= 5)
dec = entry->codecs[bytes];
if (!dec)
dec = entry->codecs[0];
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index 9be3a7b6ae..c1fce6a715 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -50,8 +50,6 @@ struct priv {
int pcmhack;
int aframesize;
int aframecount;
- int offset;
- int offset_left;
int64_t savepts;
int framecount;
int64_t lastpts;
@@ -271,9 +269,6 @@ out_takefirst:
ac->savepts = MP_NOPTS_VALUE;
ac->lastpts = MP_NOPTS_VALUE;
- ac->offset = ac->stream->codec->sample_rate *
- encode_lavc_getoffset(ao->encode_lavc_ctx, ac->stream);
- ac->offset_left = ac->offset;
ao->untimed = true;
ao->priv = ac;
@@ -450,7 +445,6 @@ static int play(struct ao *ao, void **ni_data, int samples, int flags)
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
int bufpos = 0;
- int64_t ptsoffset;
void *paddingbuf = NULL;
double nextpts;
double pts = ao->pts;
@@ -542,63 +536,10 @@ static int play(struct ao *ao, void **ni_data, int samples, int flags)
// Absurd, as this range MUST contain at least one multiple of B.
}
- ptsoffset = ac->offset;
- // this basically just edits ao->apts for syncing purposes
-
- if (ectx->options->copyts || ectx->options->rawts) {
- // we do not send time sync data to the video side,
- // but we always need the exact pts, even if zero
- } else {
- // here we must "simulate" the pts editing
- // 1. if we have to skip stuff, we skip it
- // 2. if we have to add samples, we add them
- // 3. we must still adjust ptsoffset appropriately for AV sync!
- // invariant:
- // if no partial skipping is done, the first frame gets ao->apts passed as pts!
-
- if (ac->offset_left < 0) {
- if (ac->offset_left <= -len) {
- // skip whole frame
- ac->offset_left += len;
- return len;
- } else {
- // skip part of this frame, buffer/encode the rest
- bufpos -= ac->offset_left;
- ptsoffset += ac->offset_left;
- ac->offset_left = 0;
- }
- } else if (ac->offset_left > 0) {
- // make a temporary buffer, filled with zeroes at the start
- // (don't worry, only happens once)
-
- paddingbuf = talloc_size(ac, ac->sample_size * ao->channels.num *
- (ac->offset_left + len));
- fill_with_padding(paddingbuf, ac->offset_left, ac->sample_size,
- ac->sample_padding);
- data = (char *) paddingbuf + ac->sample_size * ao->channels.num *
- ac->offset_left;
- bufpos -= ac->offset_left; // yes, negative!
- ptsoffset += ac->offset_left;
- ac->offset_left = 0;
-
- // now adjust the bufpos so the final value of bufpos is positive!
- /*
- int cnt = (len - bufpos) / ac->aframesize;
- int finalbufpos = bufpos + cnt * ac->aframesize;
- */
- int finalbufpos = len - (len - bufpos) % ac->aframesize;
- if (finalbufpos < 0) {
- MP_WARN(ao, "cannot attain the "
- "exact requested audio sync; shifting by %d frames\n",
- -finalbufpos);
- bufpos -= finalbufpos;
- }
- }
- }
-
+ // Fix and apply the discontinuity pts offset.
if (!ectx->options->rawts && ectx->options->copyts) {
// fix the discontinuity pts offset
- nextpts = pts + ptsoffset / (double) ao->samplerate;
+ nextpts = pts;
if (ectx->discontinuity_pts_offset == MP_NOPTS_VALUE) {
ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
}
@@ -611,23 +552,27 @@ static int play(struct ao *ao, void **ni_data, int samples, int flags)
outpts = pts + ectx->discontinuity_pts_offset;
}
- else
+ else {
outpts = pts;
+ }
+
+ // Shift pts by the pts offset first.
+ outpts += encode_lavc_getoffset(ectx, ac->stream);
while (len - bufpos >= ac->aframesize) {
encode(ao,
- outpts + (bufpos + ptsoffset) / (double) ao->samplerate + encode_lavc_getoffset(ectx, ac->stream),
+ outpts + bufpos / (double) ao->samplerate,
(char *) data + ac->sample_size * bufpos * ao->channels.num);
bufpos += ac->aframesize;
}
talloc_free(paddingbuf);
- // calculate expected pts of next audio frame
- ac->expected_next_pts = pts + (bufpos + ptsoffset) / (double) ao->samplerate;
+ // Calculate expected pts of next audio frame (input side).
+ ac->expected_next_pts = pts + bufpos / (double) ao->samplerate;
+ // Set next allowed input pts value (input side).
if (!ectx->options->rawts && ectx->options->copyts) {
- // set next allowed output pts value
nextpts = ac->expected_next_pts + ectx->discontinuity_pts_offset;
if (nextpts > ectx->next_in_pts)
ectx->next_in_pts = nextpts;
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-06 16:20:45 +0000