On Thu, 21 Apr 2016, Reimar Döffinger wrote:

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

Signed-off-by: Marton Balint <c...@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).

You're right, this is a hack, and probably alsa is buggy, or SDL could create the player differently. I tried to reproduce the issue with pure alsa, unfotunately without success.

In the meantime I also found another workaround, setting the SDL_AUDIO_ALSA_SET_BUFFER_SIZE environment variable, as this causes the initialized alsa buffer to have 4 periods instead of 2, which also makes the issue disappear. Setting this (if the variable is not already defined) is an acceptable workaround?

I see little chance of me digging into the deep of ALSA, I have no better idea which benefits everybody other than setting the SDL_AUDIO_ALSA_SET_BUFFER_SIZE environment variable.

Regards,
Marton
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