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audio: Add reverse playback support to GstAudioStreamAlign
https://bugzilla.gnome.org/show_bug.cgi?id=787560
This commit is contained in:
Sebastian Dröge
parent
ec1e20ffe5
commit
d2fd740388
2 changed files with 236 additions and 22 deletions
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@ -64,6 +64,7 @@ struct _GstAudioStreamAlign
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* Allocate a new #GstAudioStreamAlign with the given configuration. All
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* processing happens according to sample rate @rate, until
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* gst_audio_discont_wait_set_rate() is called with a new @rate.
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* A negative rate can be used for reverse playback.
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*
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* @alignment_threshold gives the tolerance in nanoseconds after which a
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* timestamp difference is considered a discontinuity. Once detected,
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@ -82,7 +83,7 @@ gst_audio_stream_align_new (gint rate, GstClockTime alignment_threshold,
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{
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GstAudioStreamAlign *align;
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g_return_val_if_fail (rate > 0, NULL);
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g_return_val_if_fail (rate != 0, NULL);
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align = g_new0 (GstAudioStreamAlign, 1);
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align->rate = rate;
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@ -147,7 +148,7 @@ void
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gst_audio_stream_align_set_rate (GstAudioStreamAlign * align, gint rate)
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{
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g_return_if_fail (align != NULL);
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g_return_if_fail (rate > 0);
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g_return_if_fail (rate != 0);
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if (align->rate == rate)
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return;
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@ -283,10 +284,18 @@ gst_audio_stream_align_mark_discont (GstAudioStreamAlign * align)
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* discontinuity differs by more than the alignment threshold for a duration
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* longer than discont wait.
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*
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* Note: In reverse playback, every buffer is considered discontinuous in the
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* context of buffer flags because the last sample of the previous buffer is
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* discontinuous with the first sample of the current one. However for this
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* function they are only considered discontinuous in reverse playback if the
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* first sample of the previous buffer is discontinuous with the last sample
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* of the current one.
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*
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* Returns: %TRUE if a discontinuity was detected, %FALSE otherwise.
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*
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* Since: 1.14
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*/
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#define ABSDIFF(a, b) ((a) > (b) ? (a) - (b) : (b) - (a))
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gboolean
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gst_audio_stream_align_process (GstAudioStreamAlign * align,
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gboolean discont, GstClockTime timestamp, guint n_samples,
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@ -299,10 +308,12 @@ gst_audio_stream_align_process (GstAudioStreamAlign * align,
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g_return_val_if_fail (align != NULL, FALSE);
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start_time = timestamp;
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start_offset = gst_util_uint64_scale (start_time, align->rate, GST_SECOND);
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start_offset =
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gst_util_uint64_scale (start_time, ABS (align->rate), GST_SECOND);
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end_offset = start_offset + n_samples;
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end_time = gst_util_uint64_scale_int (end_offset, GST_SECOND, align->rate);
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end_time =
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gst_util_uint64_scale_int (end_offset, GST_SECOND, ABS (align->rate));
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duration = end_time - start_time;
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@ -312,24 +323,27 @@ gst_audio_stream_align_process (GstAudioStreamAlign * align,
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guint64 diff, max_sample_diff;
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/* Check discont */
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if (start_offset <= align->next_offset)
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diff = align->next_offset - start_offset;
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else
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diff = start_offset - align->next_offset;
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if (align->rate > 0) {
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diff = ABSDIFF (start_offset, align->next_offset);
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} else {
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diff = ABSDIFF (end_offset, align->next_offset);
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}
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max_sample_diff =
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gst_util_uint64_scale_int (align->alignment_threshold,
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align->rate, GST_SECOND);
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ABS (align->rate), GST_SECOND);
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/* Discont! */
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if (G_UNLIKELY (diff >= max_sample_diff)) {
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if (align->discont_wait > 0) {
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if (align->discont_time == GST_CLOCK_TIME_NONE) {
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align->discont_time = start_time;
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} else if ((start_time >= align->discont_time
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&& start_time - align->discont_time >= align->discont_wait)
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|| (start_time < align->discont_time
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&& align->discont_time - start_time >= align->discont_wait)) {
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align->discont_time = align->rate > 0 ? start_time : end_time;
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} else if ((align->rate > 0
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&& ABSDIFF (start_time,
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align->discont_time) >= align->discont_wait)
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|| (align->rate < 0
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&& ABSDIFF (end_time,
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align->discont_time) >= align->discont_wait)) {
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discont = TRUE;
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align->discont_time = GST_CLOCK_TIME_NONE;
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}
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@ -348,22 +362,41 @@ gst_audio_stream_align_process (GstAudioStreamAlign * align,
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GST_INFO ("Have discont. Expected %"
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G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
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align->next_offset, start_offset);
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align->next_offset = end_offset;
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align->next_offset = align->rate > 0 ? end_offset : start_offset;
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/* Got a discont and adjusted, reset the discont_time marker */
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align->discont_time = GST_CLOCK_TIME_NONE;
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} else {
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/* No discont, just keep counting */
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timestamp =
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gst_util_uint64_scale (align->next_offset, GST_SECOND, align->rate);
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if (align->rate > 0) {
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timestamp =
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gst_util_uint64_scale (align->next_offset, GST_SECOND,
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ABS (align->rate));
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start_offset = align->next_offset;
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align->next_offset += n_samples;
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start_offset = align->next_offset;
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align->next_offset += n_samples;
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duration =
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gst_util_uint64_scale (align->next_offset, GST_SECOND,
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align->rate) - timestamp;
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duration =
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gst_util_uint64_scale (align->next_offset, GST_SECOND,
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ABS (align->rate)) - timestamp;
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} else {
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guint64 old_offset = align->next_offset;
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if (align->next_offset > n_samples)
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align->next_offset -= n_samples;
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else
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align->next_offset = 0;
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start_offset = align->next_offset;
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timestamp =
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gst_util_uint64_scale (align->next_offset, GST_SECOND,
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ABS (align->rate));
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duration =
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gst_util_uint64_scale (old_offset, GST_SECOND,
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ABS (align->rate)) - timestamp;
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}
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}
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if (out_timestamp)
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@ -375,3 +408,5 @@ gst_audio_stream_align_process (GstAudioStreamAlign * align,
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return discont;
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}
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#undef ABSDIFF
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@ -883,6 +883,184 @@ GST_START_TEST (test_stream_align)
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GST_END_TEST;
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GST_START_TEST (test_stream_align_reverse)
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{
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GstAudioStreamAlign *align;
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gint i;
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GstClockTime timestamp;
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GstClockTime out_timestamp, out_duration;
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gboolean discont;
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align = gst_audio_stream_align_new (-1000);
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for (i = 499; i >= 0; i--) {
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timestamp = 10 * GST_MSECOND * i;
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discont = i == 499;
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discont =
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gst_audio_stream_align_process (align, discont, timestamp, 10,
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&out_timestamp, &out_duration, NULL);
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fail_unless_equals_uint64 (out_timestamp, 10 * GST_MSECOND * i);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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if (i == 499)
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fail_unless (discont);
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else
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fail_unless (!discont);
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}
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/* Drift forwards by 1ms per 10ms buffer for the first 40 buffers.
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* - after 40 buffers we're above alignment threshold
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* - after 40 + 100 buffers we're at discont wait
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*/
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for (i = 499; i >= 0; i--) {
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timestamp = 10 * GST_MSECOND * i;
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discont = i == 499;
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if (i < 499)
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timestamp += 1 * GST_MSECOND * MIN (499 - i, 40);
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discont =
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gst_audio_stream_align_process (align, discont, timestamp, 10,
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&out_timestamp, &out_duration, NULL);
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if (i >= 500 - 140) {
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fail_unless_equals_uint64 (out_timestamp, 10 * GST_MSECOND * i);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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if (i == 499)
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fail_unless (discont);
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else
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fail_unless (!discont);
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} else {
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if (i == 499 - 140)
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fail_unless (discont);
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else
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fail_unless (!discont);
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fail_unless_equals_uint64 (out_timestamp,
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10 * GST_MSECOND * i + 40 * GST_MSECOND);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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}
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}
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/* Drift backwards by 1ms per 10ms buffer for the first 40 buffers.
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* - after 40 buffers we're above alignment threshold
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* - after 40 + 100 buffers we're at discont wait
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*/
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for (i = 499; i >= 4; i--) {
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timestamp = 10 * GST_MSECOND * i;
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discont = i == 499;
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if (i < 499)
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timestamp -= 1 * GST_MSECOND * MIN (499 - i, 40);
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discont =
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gst_audio_stream_align_process (align, discont, timestamp, 10,
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&out_timestamp, &out_duration, NULL);
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if (i >= 500 - 140) {
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fail_unless_equals_uint64 (out_timestamp, 10 * GST_MSECOND * i);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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if (i == 499)
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fail_unless (discont);
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else
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fail_unless (!discont);
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} else {
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if (i == 499 - 140)
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fail_unless (discont);
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else
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fail_unless (!discont);
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fail_unless_equals_uint64 (out_timestamp,
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10 * GST_MSECOND * i - 40 * GST_MSECOND);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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}
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}
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/* Shift all buffers but the first by 40ms
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* - after 1 buffers we're above alignment threshold
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* - after 101 buffers we're at discont wait
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*/
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for (i = 499; i >= 0; i--) {
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timestamp = 10 * GST_MSECOND * i;
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discont = i == 499;
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if (i < 499)
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timestamp += 40 * GST_MSECOND;
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discont =
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gst_audio_stream_align_process (align, discont, timestamp, 10,
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&out_timestamp, &out_duration, NULL);
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if (i >= 500 - 101) {
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fail_unless_equals_uint64 (out_timestamp, 10 * GST_MSECOND * i);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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if (i == 499)
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fail_unless (discont);
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else
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fail_unless (!discont);
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} else {
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if (i == 499 - 101)
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fail_unless (discont);
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else
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fail_unless (!discont);
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fail_unless_equals_uint64 (out_timestamp,
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10 * GST_MSECOND * i + 40 * GST_MSECOND);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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}
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}
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/* Shift every second buffer by 40ms:
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* - never discont!
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*/
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for (i = 499; i >= 0; i--) {
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timestamp = 10 * GST_MSECOND * i;
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discont = i == 499;
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if (i % 2 == 0 && i < 499)
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timestamp += 40 * GST_MSECOND;
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discont =
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gst_audio_stream_align_process (align, discont, timestamp, 10,
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&out_timestamp, &out_duration, NULL);
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fail_unless_equals_uint64 (out_timestamp, 10 * GST_MSECOND * i);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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if (i == 499)
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fail_unless (discont);
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else
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fail_unless (!discont);
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}
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/* Shift buffer 100 by 2: discont at buffer 200
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*/
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for (i = 499; i >= 0; i--) {
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timestamp = 10 * GST_MSECOND * i;
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discont = i == 499;
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if (i < 500 - 100)
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timestamp += 2 * GST_SECOND;
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discont =
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gst_audio_stream_align_process (align, discont, timestamp, 10,
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&out_timestamp, &out_duration, NULL);
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if (i >= 500 - 200) {
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fail_unless_equals_uint64 (out_timestamp, 10 * GST_MSECOND * i);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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if (i == 499)
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fail_unless (discont);
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else
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fail_unless (!discont);
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} else {
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fail_unless_equals_uint64 (out_timestamp,
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10 * GST_MSECOND * i + 2 * GST_SECOND);
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fail_unless_equals_uint64 (out_duration, 10 * GST_MSECOND);
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if (i == 499 - 200)
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fail_unless (discont);
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else
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fail_unless (!discont);
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}
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}
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}
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GST_END_TEST;
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static Suite *
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audio_suite (void)
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{
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@ -911,6 +1089,7 @@ audio_suite (void)
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tcase_add_test (tc_chain, test_audio_format_u8);
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tcase_add_test (tc_chain, test_fill_silence);
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tcase_add_test (tc_chain, test_stream_align);
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tcase_add_test (tc_chain, test_stream_align_reverse);
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return s;
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}
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