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avdevice/pulse_audio_dec: do not read undersized frames
Keep on reading fragments until we got fragment_size amount of data, otherwise we might get frames with 1-2 samples only if pa_stream_peek is called slightly less frequently than sample rate. Note that fragments might contain a lot less data than fragment_size, so reading multiple fragments to get fragment_size amount of data is intentional. Signed-off-by: Marton Balint <cus@passwd.hu>
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7f059a250b
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@ -48,6 +48,7 @@ typedef struct PulseData {
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pa_threaded_mainloop *mainloop;
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pa_context *context;
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pa_stream *stream;
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size_t pa_frame_size;
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TimeFilter *timefilter;
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int last_period;
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@ -250,6 +251,7 @@ static av_cold int pulse_read_header(AVFormatContext *s)
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goto unlock_and_fail;
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}
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pd->fragment_size = queried_attr->fragsize;
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pd->pa_frame_size = pa_frame_size(&ss);
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pa_threaded_mainloop_unlock(pd->mainloop);
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@ -261,7 +263,7 @@ static av_cold int pulse_read_header(AVFormatContext *s)
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
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1000, 1.5E-6);
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pd->fragment_size / pd->pa_frame_size, 1.5E-6);
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if (!pd->timefilter) {
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pulse_close(s);
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@ -286,12 +288,13 @@ static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
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int64_t dts;
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pa_usec_t latency;
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int negative;
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ptrdiff_t pos = 0;
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pa_threaded_mainloop_lock(pd->mainloop);
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CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
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while (!read_data) {
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while (pos < pd->fragment_size) {
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int r;
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r = pa_stream_peek(pd->stream, &read_data, &read_length);
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@ -305,43 +308,51 @@ static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
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* silence, but that wouldn't work for compressed streams. */
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r = pa_stream_drop(pd->stream);
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CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
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} else {
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if (!pos) {
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if (av_new_packet(pkt, pd->fragment_size) < 0) {
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ret = AVERROR(ENOMEM);
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goto unlock_and_fail;
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}
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dts = av_gettime();
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pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
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if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
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if (negative) {
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dts += latency;
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} else
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dts -= latency;
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} else {
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av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
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}
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}
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if (pkt->size - pos < read_length) {
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if (pos)
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break;
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pa_stream_drop(pd->stream);
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/* Oversized fragment??? */
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ret = AVERROR_EXTERNAL;
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goto unlock_and_fail;
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}
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memcpy(pkt->data + pos, read_data, read_length);
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pos += read_length;
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pa_stream_drop(pd->stream);
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}
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}
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if (av_new_packet(pkt, read_length) < 0) {
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ret = AVERROR(ENOMEM);
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goto unlock_and_fail;
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}
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dts = av_gettime();
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pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
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if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
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enum AVCodecID codec_id =
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s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
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int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels);
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int frame_duration = read_length / frame_size;
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if (negative) {
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dts += latency;
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} else
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dts -= latency;
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if (pd->wallclock)
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pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);
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pd->last_period = frame_duration;
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} else {
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av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
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}
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memcpy(pkt->data, read_data, read_length);
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pa_stream_drop(pd->stream);
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pa_threaded_mainloop_unlock(pd->mainloop);
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av_shrink_packet(pkt, pos);
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if (pd->wallclock)
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pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);
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pd->last_period = pkt->size / pd->pa_frame_size;
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return 0;
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unlock_and_fail:
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av_packet_unref(pkt);
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pa_threaded_mainloop_unlock(pd->mainloop);
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return ret;
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}
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