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avdevice/alsa_dec: make sure we have enough data in non-blocking mode
Otherwise we might return 1-2 samples per packet if av_read_frame() call rate is only sligthly less than the stream sample rate. Signed-off-by: Marton Balint <cus@passwd.hu>
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6e7e3a3820
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@ -286,6 +286,10 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
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}
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}
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s->pkt = av_packet_alloc();
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if (!s->pkt)
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goto fail1;
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s->h = h;
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return 0;
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@ -308,6 +312,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1)
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if (CONFIG_ALSA_INDEV)
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ff_timefilter_destroy(s->timefilter);
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snd_pcm_close(s->h);
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av_packet_free(&s->pkt);
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return 0;
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}
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@ -58,6 +58,7 @@ typedef struct AlsaData {
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void *reorder_buf;
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int reorder_buf_size; ///< in frames
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int64_t timestamp; ///< current timestamp, without latency applied.
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AVPacket *pkt;
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} AlsaData;
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/**
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@ -105,34 +105,36 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
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int64_t dts;
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snd_pcm_sframes_t delay = 0;
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if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
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return AVERROR(EIO);
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if (!s->pkt->data) {
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int ret = av_new_packet(s->pkt, s->period_size * s->frame_size);
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if (ret < 0)
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return ret;
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s->pkt->size = 0;
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}
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while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
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do {
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while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, s->period_size - s->pkt->size / s->frame_size)) < 0) {
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if (res == -EAGAIN) {
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av_packet_unref(pkt);
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return AVERROR(EAGAIN);
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}
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s->pkt->size = 0;
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if (ff_alsa_xrun_recover(s1, res) < 0) {
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av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
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snd_strerror(res));
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av_packet_unref(pkt);
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return AVERROR(EIO);
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}
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ff_timefilter_reset(s->timefilter);
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}
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}
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s->pkt->size += res * s->frame_size;
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} while (s->pkt->size < s->period_size * s->frame_size);
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av_packet_move_ref(pkt, s->pkt);
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dts = av_gettime();
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snd_pcm_delay(s->h, &delay);
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dts -= av_rescale(delay + res, 1000000, s->sample_rate);
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pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
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s->last_period = res;
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pkt->size = res * s->frame_size;
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return 0;
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}
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