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audiodecoder: Don't be too picky about the output frame counter

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With most decoder libraries, and especially when accessing codecs via
OpenMAX or similar APIs, we don't have the ability to properly related
the output buffers to a number of input samples. And could e.g. get
a fractional number of input buffers decoded at a time.

Previously this would in the end lead to an error message and stopped
playback. Change it to a warning message instead and try to handle it
gracefully. In theory the subclass can now get timestamp tracking
wrong if it completely misuses the API, but if on average it behaves
correct (and gst-omx and others do) it will continue to work properly.

Also add a test for the new behaviour.

We don't change it in the encoder yet as that requires more internal logic
changes AFAIU and I'm not aware of a case where this was a problem so far.
This commit is contained in:
Sebastian Dröge 2014-06-20 11:00:14 +02:00
parent 5ebfe5b26b
commit 909dd7831b
2 changed files with 76 additions and 18 deletions
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29
gst-libs/gst/audio/gstaudiodecoder.c
View file

@ -1173,16 +1173,21 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
/* frame and ts book-keeping */
if (G_UNLIKELY (frames < 0)) {
if (G_UNLIKELY (-frames - 1 > priv->frames.length))
goto overflow;
frames = priv->frames.length + frames + 1;
if (G_UNLIKELY (-frames - 1 > priv->frames.length)) {
GST_ELEMENT_WARNING (dec, STREAM, ENCODE,
("received more decoded frames %d than provided %d", frames,
priv->frames.length), (NULL));
frames = 0;
} else {
frames = priv->frames.length + frames + 1;
}
} else if (G_UNLIKELY (frames > priv->frames.length)) {
if (G_LIKELY (!priv->force)) {
/* no way we can let this pass */
g_assert_not_reached ();
/* really no way */
goto overflow;
GST_ELEMENT_WARNING (dec, STREAM, ENCODE,
("received more decoded frames %d than provided %d", frames,
priv->frames.length), (NULL));
}
frames = priv->frames.length;
}
if (G_LIKELY (priv->frames.length))
@ -1291,16 +1296,6 @@ wrong_buffer:
ret = GST_FLOW_ERROR;
goto exit;
}
overflow:
{
GST_ELEMENT_ERROR (dec, STREAM, ENCODE,
("received more decoded frames %d than provided %d", frames,
priv->frames.length), (NULL));
if (buf)
gst_buffer_unref (buf);
ret = GST_FLOW_ERROR;
goto exit;
}
}
static GstFlowReturn

65
tests/check/libs/audiodecoder.c
View file

@ -44,6 +44,7 @@ struct _GstAudioDecoderTester
GstAudioDecoder parent;
gboolean setoutputformat_on_decoding;
gboolean output_too_many_frames;
};
struct _GstAudioDecoderTesterClass
@ -129,7 +130,11 @@ gst_audio_decoder_tester_handle_frame (GstAudioDecoder * dec,
gst_buffer_unmap (buffer, &map);
return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
if (tester->output_too_many_frames) {
return gst_audio_decoder_finish_frame (dec, output_buffer, 2);
} else {
return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
}
}
static void
@ -582,6 +587,63 @@ GST_START_TEST (audiodecoder_buffer_after_segment)
}
GST_END_TEST;
GST_START_TEST (audiodecoder_output_too_many_frames)
{
GstSegment segment;
GstBuffer *buffer;
guint64 i;
setup_audiodecodertester ();
((GstAudioDecoderTester *) dec)->output_too_many_frames = TRUE;
gst_pad_set_active (mysrcpad, TRUE);
gst_element_set_state (dec, GST_STATE_PLAYING);
gst_pad_set_active (mysinkpad, TRUE);
send_startup_events ();
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
/* push buffers, the data is actually a number so we can track them */
for (i = 0; i < 3; i++) {
GstMapInfo map;
guint64 num;
buffer = create_test_buffer (i);
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
/* check that buffer was received by our source pad */
buffer = buffers->data;
gst_buffer_map (buffer, &map, GST_MAP_READ);
num = *(guint64 *) map.data;
fail_unless_equals_uint64 (i, num);
fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer),
gst_util_uint64_scale_round (i, GST_SECOND, TEST_MSECS_PER_SAMPLE));
fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
buffers = g_list_delete_link (buffers, buffers);
}
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
fail_unless (buffers == NULL);
cleanup_audiodecodertest ();
}
GST_END_TEST;
static Suite *
gst_audiodecoder_suite (void)
{
@ -595,6 +657,7 @@ gst_audiodecoder_suite (void)
tcase_add_test (tc, audiodecoder_negotiation_with_gap_event);
tcase_add_test (tc, audiodecoder_delayed_negotiation_with_gap_event);
tcase_add_test (tc, audiodecoder_buffer_after_segment);
tcase_add_test (tc, audiodecoder_output_too_many_frames);
return s;
}