[vlc-devel] [PATCH 4/5] wasapi: add 24bits support

[Thomas Guillem]

On Fri, Oct 25, 2019, at 12:17, Rémi Denis-Courmont wrote:
> I disagree and this contradicts earlier decisions here.

Yes that is what you said in the first reply and I asked you:

Who ? when ? Do you have a trace of this decision ?

> 
> Le 25 octobre 2019 10:16:28 GMT+03:00, Thomas Guillem <thomas at gllm.fr> a écrit :
>> 
>> On Thu, Oct 24, 2019, at 18:29, Rémi Denis-Courmont wrote:
>>> So what? Some cards don't support PCM at all, or only some really weird formats. That's why we use the OS-provided HAL.
>> 
>> In that case, we can also add support for such format.
>> 
>> Yes, we must use the OS-provided HAL by default. But we can still add an advanced option to enable exclusive mode. I know a lot of audiophile would like to play their 24bit 96khz directly. Even if I know that 16bit 48khz cover fully the hearing range.
>> 
>>> 
>>> I don't think this belongs in VLC. We already decided to drop 24-bits PCM earlier.
>> 
>> Who ? when ? Do you have a trace ?
>> 
>>> 
>>> Le 24 octobre 2019 15:37:12 GMT+03:00, Thomas Guillem <thomas at gllm.fr> a écrit :
>>>> But don't use VLC_CODEC_S24N as this fourcc is not handled by any audio
>>>> filters. Instead, request VLC_CODEC_S32N to VLC and do the conversion
>>>> internally.
>>>> 
>>>> This commit is needed by the next commit since some sound cards only support
>>>> 24bits. modules/audio_output/wasapi.c | 30 +++++++++++++++++++++++++++---
>>>>  1 file changed, 27 insertions(+), 3 deletions(-)
>>>> 
>>>> diff --git a/modules/audio_output/wasapi.c b/modules/audio_output/wasapi.c
>>>> index d6105169b8e..daa946f8afc 100644
>>>> --- a/modules/audio_output/wasapi.c
>>>> +++ b/modules/audio_output/wasapi.c
>>>> @@ -106,6 +106,7 @@ typedef struct aout_stream_sys
>>>>      unsigned block_align;
>>>>      UINT64 written; /**< Frames written to the buffer */
>>>>      UINT32 frames; /**< Total buffer size (frames) */
>>>> +    bool s24s32; /**< Output configured as S24N, but input as S32N */
>>>>  } aout_stream_sys_t;
>>>>  
>>>>  static void ResetTimer(aout_stream_t *s)
>>>> @@ -268,7 +269,28 @@ static HRESULT Play(aout_stream_t *s, block_t *block, vlc_tick_t date)
>>>>  
>>>>          const size_t copy = frames * sys->block_align;
>>>>  
>>>> -        memcpy(dst, block->p_buffer, copy);
>>>> +        if (!sys->s24s32)
>>>> +        {
>>>> +            memcpy(dst, block->p_buffer, copy);
>>>> +            block->p_buffer += copy;
>>>> +            block->i_buffer -= copy;
>>>> +        }
>>>> +        else
>>>> +        {
>>>> +            /* Convert back S32L to S24L. The following is doing the opposite
>>>> +             * of S24LDecode() from codec/araw.c */
>>>> +            BYTE *end = dst + copy;
>>>> +            while (dst < end)
>>>> +            {
>>>> +                dst[0] = block->p_buffer[1];
>>>> +                dst[1] = block->p_buffer[2];
>>>> +                dst[2] = block->p_buffer[3];
>>>> +                dst += 3;
>>>> +                block->p_buffer += 4;
>>>> +                block->i_buffer -= 4;
>>>> +            }
>>>> +
>>>> +        }
>>>>          hr = IAudioRenderClient_ReleaseBuffer(render, frames, 0);
>>>>          if (FAILED(hr))
>>>>          {
>>>> @@ -276,8 +298,6 @@ static HRESULT Play(aout_stream_t *s, block_t *block, vlc_tick_t date)
>>>>              break;
>>>>          }
>>>>  
>>>> -        block->p_buffer += copy;
>>>> -        block->i_buffer -= copy;
>>>>          block->i_nb_samples -= frames;
>>>>          sys->written += frames;
>>>>          if (block->i_nb_samples == 0)
>>>> @@ -512,6 +532,7 @@ static int vlc_FromWave(const WAVEFORMATEX *restrict wf,
>>>>              switch (wf->wBitsPerSample)
>>>>              {
>>>>                  case 32:
>>>> +                case 24:
>>>>                      audio->i_format = VLC_CODEC_S32N;
>>>>                      break;
>>>>                  case 16:
>>>> @@ -679,6 +700,9 @@ static HRESULT Start(aout_stream_t *s, audio_sample_format_t *restrict pfmt,
>>>>      sys->format = fmt.i_format;
>>>>      sys->block_align = pwf->nBlockAlign;
>>>>      sys->rate = pwf->nSamplesPerSec;
>>>> +    sys->s24s32 = pwf->wBitsPerSample == 24 && fmt.i_format == VLC_CODEC_S32N;
>>>> +    if (sys->s24s32)
>>>> +        msg_Dbg(s, "audio device configured as s24");
>>>>  
>>>>      hr = IAudioClient_Initialize(sys->client, shared_mode, 0, buffer_duration,
>>>>                                   0, pwf, sid);
>>> 
>>> -- 
>>> Envoyé de mon appareil Android avec Courriel K-9 Mail. Veuillez excuser ma brièveté. 
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>> 
> 
> -- 
> Envoyé de mon appareil Android avec Courriel K-9 Mail. Veuillez excuser ma brièveté. 
> _______________________________________________
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