[FFmpeg-devel] [PATCH] ffplay: use 48 kHz for ALSA output

[Reimar Döffinger]

On 20.04.2016, at 23:59, Marton Balint <cus at passwd.hu> wrote:

> Signed-off-by: Marton Balint <cus at passwd.hu>
> ---
> ffplay.c | 5 +++++
> 1 file changed, 5 insertions(+)
> 
> diff --git a/ffplay.c b/ffplay.c
> index 804bcbc..89a34d2 100644
> --- a/ffplay.c
> +++ b/ffplay.c
> @@ -2578,12 +2578,17 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
>     static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};
>     static const int next_sample_rates[] = {0, 44100, 48000, 96000, 192000};
>     int next_sample_rate_idx = FF_ARRAY_ELEMS(next_sample_rates) - 1;
> +    char driver_name[32];
> 
>     env = SDL_getenv("SDL_AUDIO_CHANNELS");
>     if (env) {
>         wanted_nb_channels = atoi(env);
>         wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
>     }
> +    /* By using the default dmix sample rate we should be able to avoid the
> +     * ALSA resampler because using it causes small buffers and underruns. */

That only makes sense if the alsa device used actually uses dmix.
Plus I am generally sceptical about such a hackish workaround (in particular making assumptions on internals like default sample rate) for what sounds like an ALSA issue (assuming we can't/shouldn't just request a larger buffer or query the "native" sample rate).