converting to 24 bit DTS using ffmpeg, but the getting 16 bit

[jbiribi]

I used this command --> "ffmpeg -i Journey1.wav -strict -2 -vn -b:a 1536k -sample_fmt s32 -f dts final.dts" but the converted audio is always 16 bit. Why? With -sample_fmt s24 I get an error saying the command doesn't exist, and with -sample_fmt s32 I'm getting a 16 bit audio.

[Marsia Mariner]

Lossy audio actually does not have bitdepth. And the bitdepth flag in the frame headers may be ignored by the decoders.
If I remember correctly, you can use eac3to for patching the DTS frame headers to 24-bits.

But now, some questions... is your source .WAV file 16-bits, 24-bits, or 32-bits? ffdcaenc supports all of these.
Also, how many channels does your source .WAV file have? If the answer is 1 or 2, then 1536kbps is overkill,
and if the answer is 6, even 1509kbps may be unnecessarily high as well — except if your goal is DVD-authoring or playback in a standalone (hardware) player.

[jbiribi]

My original source was 5.1 DTS 24 bit 1536 kbps. I converted to 5.1 PCM 24 bits 6900 kbps to use in premiere. Rendered in PCM with the same specs. Now, I want to convert this WAV PCM to his orginal format, DTS, with the same specs from the original source.

[Marsia Mariner]

Then you'd better switch to ffdcaenc.
I use ffmpeg only for decoding, demuxing and remuxing, so I have no idea of how good it is at using libdcaenc.
I have just checked (with both LeeAudBi and MediaInfo) some DTS files that I created with ffdcaenc many moons ago, and all of them are (wisely) marked as 24-bits, even when the sources were ordinary 16-bit stereo .WAV files.

[pandy]

DCA encoder capabilities:

Encoder dca [DCA (DTS Coherent Acoustics)]:
    General capabilities: exp 
    Threading capabilities: none
    Supported sample rates: 8000 16000 32000 11025 22050 44100 12000 24000 48000
    Supported sample formats: s32
    Supported channel layouts: mono stereo quad(side) 5.0(side) 5.1(side)
DCA (DTS Coherent Acoustics) AVOptions:
  -dca_adpcm         <boolean>    E...A... Use ADPCM encoding (default false)

Not sure about 24 bit encoding within DTS standard (it may imply to use different profile - check if DTS core is capable for something more than 16 bit audio - perhaps you need to use extension to core).

[jbiribi]

Then you'd better switch to ffdcaenc.
I use ffmpeg only for decoding, demuxing and remuxing, so I have no idea of how good it is at using libdcaenc.
I have just checked (with both LeeAudBi and MediaInfo) some DTS files that I created with ffdcaenc many moons ago, and all of them are (wisely) marked as 24-bits, even when the sources were ordinary 16-bit stereo .WAV files.

Ok, I'll try to use ffdcaenc. Can I use the same code I used in ffmpeg?

[jbiribi]

DCA encoder capabilities:

Encoder dca [DCA (DTS Coherent Acoustics)]:
    General capabilities: exp 
    Threading capabilities: none
    Supported sample rates: 8000 16000 32000 11025 22050 44100 12000 24000 48000
    Supported sample formats: s32
    Supported channel layouts: mono stereo quad(side) 5.0(side) 5.1(side)
DCA (DTS Coherent Acoustics) AVOptions:
  -dca_adpcm         <boolean>    E...A... Use ADPCM encoding (default false)

Not sure about 24 bit encoding within DTS standard (it may imply to use different profile - check if DTS core is capable for something more than 16 bit audio - perhaps you need to use extension to core).

I think so, because my original dts audio was 24 bit. In it information was writen 24 bit

[Marsia Mariner]

Ok, I'll try to use ffdcaenc. Can I use the same code I used in ffmpeg?

Nope, because ffdcaenc is much simpler to use.

[C:\]
=> ffdcaenc -h

FFDCAENC --- experimental 'Coherent Acoustics' compressor.

Usage:
  ffdcaenc -i <input.wav> -o <output.dts> -b <bitrate_kbps>

Optional:
  -l  Ignore input length, can be useful when reading from stdin
  -e  Switch output endianess to Little Endian (default is: Big-Endian)
  -r  Reduced Bit Depth for DTS CD format (default is: Full Bit-Depth)
  -h  Print this help screen
  -c  Overwrite the channel configuration (default is: auto-selection)
  -f  Add an additional LFE channel (default: used for 6-channel input)
  -m  Multiple Mono input files (default: -i for multi-channel input file)
       Use -0 <input.wav> -1 <input.wav> etc. (up to -5) Channels are in ITU order:
       0,1,2,3,4,5 --> LF, RF, C, LFE, LS, RS
       The following mono input file combinations are supported:
                1.0             -2 center.wav
                1.1             -2 center.wav -3 lfe-wav
                2.0             -0 left.wav -1 right.wav
                2.1             -0 left.wav -1 right.wav -3 lfe.wav
                3.0             -0 left.wav -1 right.wav -2 center.wav
                3.1             -0 left.wav -1 right.wav -2 center.wav -3 lfe.wav
                4.0             -0 left.wav -1 right.wav -4 ls.wav -5 rs.wav
                4.1             -0 left.wav -1 right.wav -4 ls.wav -5 rs.wav -3 lfe.wav
                5.0             -0 left.wav -1 right.wav -2 center.wav -4 ls.wav -5 rs.wav
                5.1             -0 left.wav -1 right.wav -2 center.wav -4 ls.wav -5 rs.wav -3 lfe.wav

  -v  Show version info

REMARKS:
The input or output filename can be "-" for stdin/stdout.
The bitrate is specified in kilobits per second and may be rounded up
-- use floating-point values for bitrates that are not a multiple of 1 kbps.
Because the encoder uses a 4-byte granularity, i.e., 32 bits per audio frame
(with 512 samples/frame), the ACTUAL bitrate will always be a multiple of:

3         kbps for     48 kHz
2.75625   kbps for   44.1 kHz
2         kbps for     32 kHz
1.5       kbps for     24 kHz
1.378125  kbps for  22.05 kHz
1         kbps for     16 kHz
0.75      kbps for     12 kHz
0.6890625 kbps for 11.025 kHz
0.5       kbps for      8 kHz

-- NOTICE: the values 377.25, 503.25, 754.5 and 1509.75 AT _48kHz_ are exceptions.

* Available channel-layouts:

  -  1: A
  -  2: A, B
  -  3: L, R
  -  4: (L+R), (L-R)
  -  5: Lt, Rt
  -  6: FC, FL, FR
  -  7: FL, FR, BC
  -  8: FC, FL, FR, BC
  -  9: FL, FR, BL, BR
  - 10: FC, FL, FR, BL, BR
  - 11: CL, CR, FL, FR, BL, BR (not supported)
  - 12: FC, FL, FR, BL, BR, OV (not supported)
  - 13: FC, BC, FL, FR, BL, BR (not supported)
  - 14: CL, FC, CR, FL, FR, BL, BR (not supported)
  - 15: CL, CR, FL, FR, SL1, SL2, SR1, SR2 (not supported)
  - 16: CL, FC, CR, FL, FR, BL, BC, BR (not supported)

[jbiribi]

Ok, I'll try to use ffdcaenc. Can I use the same code I used in ffmpeg?

Nope, because ffdcaenc is much simpler to use.

[C:\]
=> ffdcaenc -h

FFDCAENC --- experimental 'Coherent Acoustics' compressor.

Usage:
  ffdcaenc -i <input.wav> -o <output.dts> -b <bitrate_kbps>

Optional:
  -l  Ignore input length, can be useful when reading from stdin
  -e  Switch output endianess to Little Endian (default is: Big-Endian)
  -r  Reduced Bit Depth for DTS CD format (default is: Full Bit-Depth)
  -h  Print this help screen
  -c  Overwrite the channel configuration (default is: auto-selection)
  -f  Add an additional LFE channel (default: used for 6-channel input)
  -m  Multiple Mono input files (default: -i for multi-channel input file)
       Use -0 <input.wav> -1 <input.wav> etc. (up to -5) Channels are in ITU order:
       0,1,2,3,4,5 --> LF, RF, C, LFE, LS, RS
       The following mono input file combinations are supported:
                1.0             -2 center.wav
                1.1             -2 center.wav -3 lfe-wav
                2.0             -0 left.wav -1 right.wav
                2.1             -0 left.wav -1 right.wav -3 lfe.wav
                3.0             -0 left.wav -1 right.wav -2 center.wav
                3.1             -0 left.wav -1 right.wav -2 center.wav -3 lfe.wav
                4.0             -0 left.wav -1 right.wav -4 ls.wav -5 rs.wav
                4.1             -0 left.wav -1 right.wav -4 ls.wav -5 rs.wav -3 lfe.wav
                5.0             -0 left.wav -1 right.wav -2 center.wav -4 ls.wav -5 rs.wav
                5.1             -0 left.wav -1 right.wav -2 center.wav -4 ls.wav -5 rs.wav -3 lfe.wav

  -v  Show version info

REMARKS:
The input or output filename can be "-" for stdin/stdout.
The bitrate is specified in kilobits per second and may be rounded up
-- use floating-point values for bitrates that are not a multiple of 1 kbps.
Because the encoder uses a 4-byte granularity, i.e., 32 bits per audio frame
(with 512 samples/frame), the ACTUAL bitrate will always be a multiple of:

3         kbps for     48 kHz
2.75625   kbps for   44.1 kHz
2         kbps for     32 kHz
1.5       kbps for     24 kHz
1.378125  kbps for  22.05 kHz
1         kbps for     16 kHz
0.75      kbps for     12 kHz
0.6890625 kbps for 11.025 kHz
0.5       kbps for      8 kHz

-- NOTICE: the values 377.25, 503.25, 754.5 and 1509.75 AT _48kHz_ are exceptions.

* Available channel-layouts:

  -  1: A
  -  2: A, B
  -  3: L, R
  -  4: (L+R), (L-R)
  -  5: Lt, Rt
  -  6: FC, FL, FR
  -  7: FL, FR, BC
  -  8: FC, FL, FR, BC
  -  9: FL, FR, BL, BR
  - 10: FC, FL, FR, BL, BR
  - 11: CL, CR, FL, FR, BL, BR (not supported)
  - 12: FC, FL, FR, BL, BR, OV (not supported)
  - 13: FC, BC, FL, FR, BL, BR (not supported)
  - 14: CL, FC, CR, FL, FR, BL, BR (not supported)
  - 15: CL, CR, FL, FR, SL1, SL2, SR1, SR2 (not supported)
  - 16: CL, FC, CR, FL, FR, BL, BC, BR (not supported)

Ok, I will try

[pandy]

I think so, because my original dts audio was 24 bit. In it information was writen 24 bit

Based on ETSI TS 102 114 V1.4.1 (2012-09) seem that DTS core DCA shall be capable to encode 24 bit samples.

5 Core Audio
The DTS core encoder delivers 5.1 channel audio at 24 bits per sample with a sampling frequency of up to 48 kHz.

So 16 bit outcome may be internal limitation for DCA encoder implementation.

[jbiribi]

Ok, I'll try to use ffdcaenc. Can I use the same code I used in ffmpeg?

Nope, because ffdcaenc is much simpler to use.

[C:\]
=> ffdcaenc -h

FFDCAENC --- experimental 'Coherent Acoustics' compressor.

Usage:
  ffdcaenc -i <input.wav> -o <output.dts> -b <bitrate_kbps>

Optional:
  -l  Ignore input length, can be useful when reading from stdin
  -e  Switch output endianess to Little Endian (default is: Big-Endian)
  -r  Reduced Bit Depth for DTS CD format (default is: Full Bit-Depth)
  -h  Print this help screen
  -c  Overwrite the channel configuration (default is: auto-selection)
  -f  Add an additional LFE channel (default: used for 6-channel input)
  -m  Multiple Mono input files (default: -i for multi-channel input file)
       Use -0 <input.wav> -1 <input.wav> etc. (up to -5) Channels are in ITU order:
       0,1,2,3,4,5 --> LF, RF, C, LFE, LS, RS
       The following mono input file combinations are supported:
                1.0             -2 center.wav
                1.1             -2 center.wav -3 lfe-wav
                2.0             -0 left.wav -1 right.wav
                2.1             -0 left.wav -1 right.wav -3 lfe.wav
                3.0             -0 left.wav -1 right.wav -2 center.wav
                3.1             -0 left.wav -1 right.wav -2 center.wav -3 lfe.wav
                4.0             -0 left.wav -1 right.wav -4 ls.wav -5 rs.wav
                4.1             -0 left.wav -1 right.wav -4 ls.wav -5 rs.wav -3 lfe.wav
                5.0             -0 left.wav -1 right.wav -2 center.wav -4 ls.wav -5 rs.wav
                5.1             -0 left.wav -1 right.wav -2 center.wav -4 ls.wav -5 rs.wav -3 lfe.wav

  -v  Show version info

REMARKS:
The input or output filename can be "-" for stdin/stdout.
The bitrate is specified in kilobits per second and may be rounded up
-- use floating-point values for bitrates that are not a multiple of 1 kbps.
Because the encoder uses a 4-byte granularity, i.e., 32 bits per audio frame
(with 512 samples/frame), the ACTUAL bitrate will always be a multiple of:

3         kbps for     48 kHz
2.75625   kbps for   44.1 kHz
2         kbps for     32 kHz
1.5       kbps for     24 kHz
1.378125  kbps for  22.05 kHz
1         kbps for     16 kHz
0.75      kbps for     12 kHz
0.6890625 kbps for 11.025 kHz
0.5       kbps for      8 kHz

-- NOTICE: the values 377.25, 503.25, 754.5 and 1509.75 AT _48kHz_ are exceptions.

* Available channel-layouts:

  -  1: A
  -  2: A, B
  -  3: L, R
  -  4: (L+R), (L-R)
  -  5: Lt, Rt
  -  6: FC, FL, FR
  -  7: FL, FR, BC
  -  8: FC, FL, FR, BC
  -  9: FL, FR, BL, BR
  - 10: FC, FL, FR, BL, BR
  - 11: CL, CR, FL, FR, BL, BR (not supported)
  - 12: FC, FL, FR, BL, BR, OV (not supported)
  - 13: FC, BC, FL, FR, BL, BR (not supported)
  - 14: CL, FC, CR, FL, FR, BL, BR (not supported)
  - 15: CL, CR, FL, FR, SL1, SL2, SR1, SR2 (not supported)
  - 16: CL, FC, CR, FL, FR, BL, BC, BR (not supported)

Ok, I will try

I used this code --> ffdcaenc -i "D:\Desktop\Downloads\Journey.wav" -o output.dts -b 1536
I got this message --> Could not open or parse "D:\Desktop\Downloads\Journey.wav". Error: Data chunk size is invalid!

[Marsia Mariner]

I used this code --> ffdcaenc -i "D:\Desktop\Downloads\Journey.wav" -o output.dts -b 1536
I got this message --> Could not open or parse "D:\Desktop\Downloads\Journey.wav". Error: Data chunk size is invalid!

Two things:

1) what is the size of "Journey.wav"? If the answer is 'greater than 4 gigbytes', then you'll have to feed ffdcaenc from stdin and use the -l (ignore length) switch.

2) Are you sure "D:\Desktop\Downloads\Journey.wav" is a valid path string?

[jbiribi]

I used this code --> ffdcaenc -i "D:\Desktop\Downloads\Journey.wav" -o output.dts -b 1536
I got this message --> Could not open or parse "D:\Desktop\Downloads\Journey.wav". Error: Data chunk size is invalid!

Two things:

1) what is the size of "Journey.wav"? If the answer is 'greater than 4 gigbytes', then you'll have to feed ffdcaenc from stdin and use the -l (ignore length) switch.

2) Are you sure "D:\Desktop\Downloads\Journey.wav" is a valid path string?

Adding -l worked for me. The wav have 10,3 gb. Thank you very much for your help!!