On 2024-01-25 06:47 pm, Andreas Rheinhardt wrote:
Gyan Doshi:

On 2024-01-25 10:29 am, Andreas Rheinhardt wrote:
Gyan Doshi:
Set up framework for non-PCM decoding in-place and
add support for Dolby-E decoding.

Useful for direct transcoding of non-PCM audio in live inputs.
---
   configure          |   1 +
   doc/decoders.texi  |  40 +++
   libavcodec/s302m.c | 609 +++++++++++++++++++++++++++++++++++++--------
   3 files changed, 543 insertions(+), 107 deletions(-)

diff --git a/configure b/configure
index c8ae0a061d..8db3fa3f4b 100755
--- a/configure
+++ b/configure
@@ -2979,6 +2979,7 @@ rv20_decoder_select="h263_decoder"
   rv20_encoder_select="h263_encoder"
   rv30_decoder_select="golomb h264pred h264qpel mpegvideodec rv34dsp"
   rv40_decoder_select="golomb h264pred h264qpel mpegvideodec rv34dsp"
+s302m_decoder_select="dolby_e_decoder"
   screenpresso_decoder_deps="zlib"
   shorten_decoder_select="bswapdsp"
   sipr_decoder_select="lsp"
diff --git a/doc/decoders.texi b/doc/decoders.texi
index 293c82c2ba..9f85c876bf 100644
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -347,6 +347,46 @@ configuration. You need to explicitly configure
the build with
   An FFmpeg native decoder for Opus exists, so users can decode Opus
   without this library.
   +@section s302m
+
+SMPTE ST 302 decoder.
+
+SMPTE ST 302 is a method for storing AES3 data format within an MPEG
Transport
+Stream. AES3 streams can contain LPCM streams of 2, 4, 6 or 8
channels with a
+bit depth of 16, 20 or 24-bits at a sample rate of 48 kHz.
+They can also contain non-PCM codec streams such as AC-3 or Dolby-E.
+
This sounds like we should add bitstream filters to extract the proper
underlying streams instead.
(I see only two problems with this approach: The BSF API needs to set
the CodecID of the output during init, but at this point no packet has
reached the BSF to determine it. And changing codec IDs mid-stream is
also not supported.)
In theory, this decoder shouldn't exist, as it is just a carrier,
whether of LPCM or non-PCM.
FFmpeg architecture also imposes a fundamental limitation in that one
s302m stream may
carry multiple payload streams and we support only one decoding context
per input stream
Then why does the demuxer not separate the data into multiple streams?

I didn't add demuxing support for this codec in MPEGTS, but I can venture

a) it would mean essentially inlining this decoder in the demuxer.
b) it would lead to a mapping of 1 actual to N virtual streams. How would stream_ids be assigned? c) this codec specifies for multiple payloads but I haven't seen an actual sample


+
+    ret = init_get_bits8(&gb, buf, buf_size);
+    if (ret < 0)
+        return ret;
+
+    aes_frm_size = (s->bits + 4) * 2 / 8;
+    if (buf_size < aes_frm_size * 2)  // not enough to contain
data_type & length_code
+        return AVERROR_INVALIDDATA;
+
+    state = get_bits64(&gb, aes_frm_size * 8);
+
+    while (!IS_NONPCMSYNC(s->bits,state) && (get_bits_left(&gb) >= 8))
+        state = (state << 8) | get_bits(&gb, 8);
Reading byte-aligned data with a GetBit context is very suboptimal.
What is the performance difference vs. uint8 pointers?
Typically worse by a constant factor.

But in this scenario (of a few dozen bytes per packet if dolby_e), how significant is it?

Regards,
Gyan
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