[FFmpeg-devel] [PATCH v2 2/5] avdevice/alsa_dec: make sure we have enough data in non-blocking mode

[Marton Balint]

On Wed, 10 Mar 2021, Marton Balint wrote:

>
>
> On Sat, 6 Mar 2021, Marton Balint wrote:
>
>>
>>
>> On Mon, 1 Mar 2021, Marton Balint wrote:
>>
>>> Otherwise we might return 1-2 samples per packet if av_read_frame() call 
>> rate is
>>> only sligthly less than the stream sample rate.
>>
>> Ping for this and the rest of the series. Note that the approach in 
>> this patch makes 1/5 unneeded because constant frame size is now indeed 
>> guaranteed.
>
> Will apply the series soon.

Applied.

Regards,
Marton

>
> Regards,
> Marton
>
>>>
>>> Signed-off-by: Marton Balint <cus at passwd.hu>
>>> ---
>>> libavdevice/alsa.c     |  5 +++++
>>> libavdevice/alsa.h     |  1 +
>>> libavdevice/alsa_dec.c | 22 ++++++++++++----------
>>> 3 files changed, 18 insertions(+), 10 deletions(-)
>>>
>>> diff --git a/libavdevice/alsa.c b/libavdevice/alsa.c
>>> index 117b2ea144..ee282fac16 100644
>>> --- a/libavdevice/alsa.c
>>> +++ b/libavdevice/alsa.c
>>> @@ -286,6 +286,10 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, 
>> snd_pcm_stream_t mode,
>>>         }
>>>     }
>>> 
>>> +    s->pkt = av_packet_alloc();
>>> +    if (!s->pkt)
>>> +        goto fail1;
>>> +
>>>     s->h = h;
>>>     return 0;
>>> 
>>> @@ -308,6 +312,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1)
>>>     if (CONFIG_ALSA_INDEV)
>>>         ff_timefilter_destroy(s->timefilter);
>>>     snd_pcm_close(s->h);
>>> +    av_packet_free(&s->pkt);
>>>     return 0;
>>> }
>>> 
>>> diff --git a/libavdevice/alsa.h b/libavdevice/alsa.h
>>> index 1ed8c82199..07783c983a 100644
>>> --- a/libavdevice/alsa.h
>>> +++ b/libavdevice/alsa.h
>>> @@ -58,6 +58,7 @@ typedef struct AlsaData {
>>>     void *reorder_buf;
>>>     int reorder_buf_size; ///< in frames
>>>     int64_t timestamp; ///< current timestamp, without latency applied.
>>> +    AVPacket *pkt;
>>> } AlsaData;
>>> 
>>> /**
>>> diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c
>>> index 6d568737b3..88bf32d25f 100644
>>> --- a/libavdevice/alsa_dec.c
>>> +++ b/libavdevice/alsa_dec.c
>>> @@ -104,34 +104,36 @@ static int audio_read_packet(AVFormatContext *s1, 
>> AVPacket *pkt)
>>>     int64_t dts;
>>>     snd_pcm_sframes_t delay = 0;
>>> 
>>> -    if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
>>> -        return AVERROR(EIO);
>>> +    if (!s->pkt->data) {
>>> +        int ret = av_new_packet(s->pkt, s->period_size * s->frame_size);
>>> +        if (ret < 0)
>>> +            return ret;
>>> +        s->pkt->size = 0;
>>>     }
>>> 
>>> -    while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
>>> +    do {
>>> +        while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, 
>> s->period_size - s->pkt->size / s->frame_size)) < 0) {
>>>         if (res == -EAGAIN) {
>>> -            av_packet_unref(pkt);
>>> -
>>>             return AVERROR(EAGAIN);
>>>         }
>>> +        s->pkt->size = 0;
>>>         if (ff_alsa_xrun_recover(s1, res) < 0) {
>>>             av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
>>>                    snd_strerror(res));
>>> -            av_packet_unref(pkt);
>>> -
>>>             return AVERROR(EIO);
>>>         }
>>>         ff_timefilter_reset(s->timefilter);
>>> -    }
>>> +        }
>>> +        s->pkt->size += res * s->frame_size;
>>> +    } while (s->pkt->size < s->period_size * s->frame_size);
>>> 
>>> +    av_packet_move_ref(pkt, s->pkt);
>>>     dts = av_gettime();
>>>     snd_pcm_delay(s->h, &delay);
>>>     dts -= av_rescale(delay + res, 1000000, s->sample_rate);
>>>     pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
>>>     s->last_period = res;
>>> 
>>> -    pkt->size = res * s->frame_size;
>>> -
>>>     return 0;
>>> }
>>> 
>>> -- 
>>> 2.26.2
>>>
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