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

Marton Balint cus at passwd.hu
Wed Mar 10 21:25:14 EET 2021



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.

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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