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gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
Original commit message from CVS: 2007-03-14 Julien MOUTTE <julien@moutte.net> * gst/audioresample/gstaudioresample.c: (gst_audioresample_init), (audioresample_transform_size), (audioresample_do_output), (audioresample_transform), (audioresample_pushthrough): Handle discontinuous streams. * gst/audioresample/gstaudioresample.h: * tests/check/elements/audioresample.c: (test_discont_stream_instance), (GST_START_TEST), (audioresample_suite): Add a test for discontinuous streams. * win32/common/config.h: Updated.
This commit is contained in:
Julien Moutte
parent
8b80c6f13a
commit
6940042ecf
5 changed files with 151 additions and 15 deletions
12
ChangeLog
12
ChangeLog
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@ -1,3 +1,15 @@
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2007-03-14 Julien MOUTTE <julien@moutte.net>
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* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
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(audioresample_transform_size), (audioresample_do_output),
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(audioresample_transform), (audioresample_pushthrough): Handle
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discontinuous streams.
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* gst/audioresample/gstaudioresample.h:
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* tests/check/elements/audioresample.c:
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(test_discont_stream_instance), (GST_START_TEST),
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(audioresample_suite): Add a test for discontinuous streams.
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* win32/common/config.h: Updated.
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2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
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* po/af.po:
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@ -194,6 +194,8 @@ gst_audioresample_init (GstAudioresample * audioresample,
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gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
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audioresample->filter_length = DEFAULT_FILTERLEN;
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audioresample->need_discont = FALSE;
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}
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/* vmethods */
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@ -371,7 +373,7 @@ audioresample_transform_size (GstBaseTransform * base,
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gboolean use_internal = FALSE; /* whether we use the internal state */
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gboolean ret = TRUE;
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GST_DEBUG_OBJECT (base, "asked to transform size %d in direction %s",
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GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
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size, direction == GST_PAD_SINK ? "SINK" : "SRC");
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if (direction == GST_PAD_SINK) {
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sinkcaps = caps;
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@ -406,7 +408,7 @@ audioresample_transform_size (GstBaseTransform * base,
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/* we make room for one extra sample, given that the resampling filter
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* can output an extra one for non-integral i_rate/o_rate */
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GST_DEBUG_OBJECT (base, "transformed size %d to %d", size, *othersize);
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GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);
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if (!use_internal) {
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resample_free (state);
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@ -492,8 +494,7 @@ audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
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r = audioresample->resample;
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outsize = resample_get_output_size (r);
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GST_DEBUG_OBJECT (audioresample, "audioresample can give me %d bytes",
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outsize);
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GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
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/* protect against mem corruption */
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if (outsize > GST_BUFFER_SIZE (outbuf)) {
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@ -556,6 +557,13 @@ audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
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}
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GST_BUFFER_SIZE (outbuf) = outsize;
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if (G_UNLIKELY (audioresample->need_discont)) {
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GST_DEBUG_OBJECT (audioresample,
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"marking this buffer with the DISCONT flag");
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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audioresample->need_discont = FALSE;
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}
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GST_LOG_OBJECT (audioresample, "transformed to buffer of %ld bytes, ts %"
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GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
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G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
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@ -591,6 +599,25 @@ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
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GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
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GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
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/* check for timestamp discontinuities and flush/reset if needed */
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if (GST_CLOCK_TIME_IS_VALID (audioresample->prev_ts) &&
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GST_CLOCK_TIME_IS_VALID (audioresample->prev_duration)) {
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GstClockTime ts_expected = audioresample->prev_ts +
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audioresample->prev_duration;
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GstClockTimeDiff ts_diff = GST_CLOCK_DIFF (ts_expected, timestamp);
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if (G_UNLIKELY (ts_diff != 0)) {
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GST_WARNING_OBJECT (audioresample,
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"encountered timestamp discontinuity of %" G_GINT64_FORMAT, ts_diff);
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/* Flush internal samples */
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audioresample_pushthrough (audioresample);
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/* Inform downstream element about discontinuity */
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audioresample->need_discont = TRUE;
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/* We want to recalculate the offset */
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audioresample->ts_offset = -1;
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}
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}
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if (audioresample->ts_offset == -1) {
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/* if we don't know the initial offset yet, calculate it based on the
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* input timestamp. */
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@ -610,6 +637,8 @@ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
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gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
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}
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}
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audioresample->prev_ts = timestamp;
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audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
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/* need to memdup, resample takes ownership. */
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datacopy = g_memdup (data, size);
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@ -631,17 +660,25 @@ audioresample_pushthrough (GstAudioresample * audioresample)
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r = audioresample->resample;
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outsize = resample_get_output_size (r);
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if (outsize == 0)
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goto done;
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outbuf = gst_buffer_new_and_alloc (outsize);
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res = audioresample_do_output (audioresample, outbuf);
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if (res != GST_FLOW_OK)
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if (outsize == 0) {
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GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
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goto done;
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}
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trans = GST_BASE_TRANSFORM (audioresample);
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res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
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GST_PAD_CAPS (trans->srcpad), &outbuf);
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if (G_UNLIKELY (res != GST_FLOW_OK)) {
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GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
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outsize);
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goto done;
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}
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res = audioresample_do_output (audioresample, outbuf);
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if (G_UNLIKELY (res != GST_FLOW_OK))
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goto done;
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res = gst_pad_push (trans->srcpad, outbuf);
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done:
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@ -53,10 +53,12 @@ struct _GstAudioresample {
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GstCaps *srccaps, *sinkcaps;
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gboolean passthru;
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gboolean need_discont;
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guint64 offset;
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guint64 ts_offset;
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GstClockTime next_ts;
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GstClockTime prev_ts, prev_duration;
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int channels;
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int i_rate;
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@ -144,6 +144,7 @@ fail_unless_perfect_stream ()
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buffers = NULL;
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}
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/* this tests that the output is a perfect stream if the input is */
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static void
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test_perfect_stream_instance (int inrate, int outrate, int samples,
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int numbuffers)
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@ -224,6 +225,89 @@ GST_START_TEST (test_perfect_stream)
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GST_END_TEST;
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/* this tests that the output is a correct discontinuous stream
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* if the input is; ie input drops in time come out the same way */
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static void
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test_discont_stream_instance (int inrate, int outrate, int samples,
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int numbuffers)
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{
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GstElement *audioresample;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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GstClockTime ints;
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int i, j;
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gint16 *p;
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audioresample = setup_audioresample (2, inrate, outrate);
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caps = gst_pad_get_negotiated_caps (mysrcpad);
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fail_unless (gst_caps_is_fixed (caps));
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fail_unless (gst_element_set_state (audioresample,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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for (j = 1; j <= numbuffers; ++j) {
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inbuffer = gst_buffer_new_and_alloc (samples * 4);
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GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
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/* "drop" half the buffers */
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ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
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GST_BUFFER_TIMESTAMP (inbuffer) = ints;
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GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
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GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
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gst_buffer_set_caps (inbuffer, caps);
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p = (gint16 *) GST_BUFFER_DATA (inbuffer);
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/* create a 16 bit signed ramp */
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for (i = 0; i < samples; ++i) {
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*p = -32767 + i * (65535 / samples);
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++p;
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*p = -32767 + i * (65535 / samples);
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++p;
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}
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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/* check if the timestamp of the pushed buffer matches the incoming one */
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outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
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fail_if (outbuffer == NULL);
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fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
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if (j > 1) {
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fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
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"expected discont buffer");
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}
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}
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/* cleanup */
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gst_caps_unref (caps);
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cleanup_audioresample (audioresample);
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}
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GST_START_TEST (test_discont_stream)
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{
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/* integral scalings */
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test_discont_stream_instance (48000, 24000, 500, 20);
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test_discont_stream_instance (48000, 12000, 500, 20);
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test_discont_stream_instance (12000, 24000, 500, 20);
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test_discont_stream_instance (12000, 48000, 500, 20);
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/* non-integral scalings */
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test_discont_stream_instance (44100, 8000, 500, 20);
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test_discont_stream_instance (8000, 44100, 500, 20);
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/* wacky scalings */
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test_discont_stream_instance (12345, 54321, 500, 20);
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test_discont_stream_instance (101, 99, 500, 20);
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}
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GST_END_TEST;
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GST_START_TEST (test_reuse)
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{
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GstElement *audioresample;
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@ -295,6 +379,7 @@ audioresample_suite (void)
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_perfect_stream);
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tcase_add_test (tc_chain, test_discont_stream);
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tcase_add_test (tc_chain, test_reuse);
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return s;
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@ -39,7 +39,7 @@
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#define GST_LICENSE "LGPL"
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/* package name in plugins */
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#define GST_PACKAGE_NAME "GStreamer Base Plug-ins source release"
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#define GST_PACKAGE_NAME "GStreamer Base Plug-ins CVS/prerelease"
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/* package origin */
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#define GST_PACKAGE_ORIGIN "Unknown package origin"
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@ -211,13 +211,13 @@
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#undef PACKAGE_NAME "GStreamer Base Plug-ins"
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/* Define to the full name and version of this package. */
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#undef PACKAGE_STRING "GStreamer Base Plug-ins 0.10.12"
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#undef PACKAGE_STRING "GStreamer Base Plug-ins 0.10.12.1"
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/* Define to the one symbol short name of this package. */
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#undef PACKAGE_TARNAME "gst-plugins-base"
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/* Define to the version of this package. */
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#undef PACKAGE_VERSION "0.10.12"
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#undef PACKAGE_VERSION "0.10.12.1"
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/* directory where plugins are located */
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#undef PLUGINDIR
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@ -241,7 +241,7 @@
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#undef STDC_HEADERS
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/* Version number of package */
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#define VERSION "0.10.12"
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#define VERSION "0.10.12.1"
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/* Define to 1 if your processor stores words with the most significant byte
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first (like Motorola and SPARC, unlike Intel and VAX). */
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