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pitch: make it work with reverse playback

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
This commit is contained in:
Loïc Le Page 2024-02-28 19:34:15 +01:00 committed by GStreamer Marge Bot
parent 35165da586
commit 8f5325ed11
2 changed files with 123 additions and 59 deletions
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180
subprojects/gst-plugins-bad/ext/soundtouch/gstpitch.cc
View file

@ -265,8 +265,8 @@ gst_pitch_set_property (GObject * object, guint prop_id,
GST_OBJECT_LOCK (pitch);
pitch->tempo = g_value_get_float (value);
priv->stream_time_ratio =
pitch->tempo * pitch->rate * pitch->segment_applied_rate;
priv->st->setTempo (pitch->tempo * pitch->segment_applied_rate);
pitch->tempo * pitch->rate * ABS (pitch->segment_applied_rate);
priv->st->setTempo (pitch->tempo * ABS (pitch->segment_applied_rate));
GST_OBJECT_UNLOCK (pitch);
gst_pitch_update_duration (pitch);
break;
@ -274,7 +274,7 @@ gst_pitch_set_property (GObject * object, guint prop_id,
GST_OBJECT_LOCK (pitch);
pitch->rate = g_value_get_float (value);
priv->stream_time_ratio =
pitch->tempo * pitch->rate * pitch->segment_applied_rate;
pitch->tempo * pitch->rate * ABS (pitch->segment_applied_rate);
priv->st->setRate (pitch->rate);
GST_OBJECT_UNLOCK (pitch);
gst_pitch_update_duration (pitch);
@ -353,24 +353,38 @@ gst_pitch_setcaps (GstPitch * pitch, GstCaps * caps)
static GstFlowReturn
gst_pitch_forward_buffer (GstPitch * pitch, GstBuffer * buffer)
{
gint samples;
guint64 samples = GST_BUFFER_OFFSET (buffer);
GST_OBJECT_LOCK (pitch);
GST_BUFFER_TIMESTAMP (buffer) = pitch->next_buffer_time;
pitch->next_buffer_time += GST_BUFFER_DURATION (buffer);
if (pitch->segment_applied_rate >= 0) {
GST_BUFFER_PTS (buffer) = pitch->next_buffer_time;
pitch->next_buffer_time += GST_BUFFER_DURATION (buffer);
samples = GST_BUFFER_OFFSET (buffer);
GST_BUFFER_OFFSET (buffer) = pitch->next_buffer_offset;
pitch->next_buffer_offset += samples;
GST_BUFFER_OFFSET_END (buffer) = pitch->next_buffer_offset;
GST_BUFFER_OFFSET (buffer) = pitch->next_buffer_offset;
pitch->next_buffer_offset += samples;
GST_BUFFER_OFFSET_END (buffer) = pitch->next_buffer_offset;
} else {
if (pitch->next_buffer_time > GST_BUFFER_DURATION (buffer))
pitch->next_buffer_time -= GST_BUFFER_DURATION (buffer);
else
pitch->next_buffer_time = 0;
GST_BUFFER_PTS (buffer) = pitch->next_buffer_time;
GST_LOG ("pushing buffer [%" GST_TIME_FORMAT "]-[%" GST_TIME_FORMAT
"] (%d samples)", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (pitch->next_buffer_time), samples);
GST_BUFFER_OFFSET_END (buffer) = pitch->next_buffer_offset;
if (pitch->next_buffer_offset > samples)
pitch->next_buffer_offset -= samples;
else
pitch->next_buffer_offset = 0;
GST_BUFFER_OFFSET (buffer) = pitch->next_buffer_offset;
}
GST_OBJECT_UNLOCK (pitch);
GST_LOG ("pushing buffer pts: %" GST_TIME_FORMAT ", duration: %"
GST_TIME_FORMAT ", samples: %" G_GUINT64_FORMAT, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)), samples);
return gst_pad_push (pitch->srcpad, buffer);
}
@ -378,37 +392,59 @@ gst_pitch_forward_buffer (GstPitch * pitch, GstBuffer * buffer)
static GstBuffer *
gst_pitch_prepare_buffer (GstPitch * pitch)
{
GstMapInfo info;
GstBuffer *buffer = NULL;
GST_LOG_OBJECT (pitch, "preparing buffer");
GST_OBJECT_LOCK (pitch);
gboolean reverse_playback = (pitch->segment_applied_rate < 0);
gint rate = pitch->info.rate;
gint bytes_per_frame = pitch->info.bpf;
GstPitchPrivate *priv = GST_PITCH_GET_PRIVATE (pitch);
guint samples = priv->st->numSamples ();
if (samples > 0) {
buffer = gst_buffer_new_and_alloc (samples * pitch->info.bpf);
GstMapInfo info;
buffer = gst_buffer_new_and_alloc (samples * bytes_per_frame);
gst_buffer_map (buffer, &info, (GstMapFlags) GST_MAP_READWRITE);
samples =
priv->st->receiveSamples ((soundtouch::SAMPLETYPE *) info.data,
samples);
gst_buffer_unmap (buffer, &info);
if (samples > 0) {
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale (samples, GST_SECOND, pitch->info.rate);
/* temporary store samples here, to avoid having to recalculate this */
GST_BUFFER_OFFSET (buffer) = (gint64) samples;
} else {
if (samples <= 0) {
gst_buffer_unmap (buffer, &info);
gst_buffer_unref (buffer);
buffer = NULL;
}
}
GST_OBJECT_UNLOCK (pitch);
if (buffer) {
if (reverse_playback && samples > 1) {
// If playing backwards, we need to inverse the order of all received samples
guint8 *dest = info.data;
guint8 val = 0;
guint half_samples = samples >> 1;
for (guint i = samples - 1; i >= half_samples; --i) {
guint8 *src = info.data + i * bytes_per_frame;
if (src == dest)
break;
for (gint j = 0; j < bytes_per_frame; ++j) {
val = *dest;
*dest++ = *src;
*src++ = val;
}
}
}
gst_buffer_unmap (buffer, &info);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale (samples, GST_SECOND, rate);
/* temporary store samples here, to avoid having to recalculate this */
GST_BUFFER_OFFSET (buffer) = (gint64) samples;
}
return buffer;
}
@ -584,7 +620,7 @@ gst_pitch_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
GstPitch *pitch;
gboolean res = FALSE;
gfloat stream_time_ratio;
gint64 next_buffer_offset;
guint64 next_buffer_offset;
GstClockTime next_buffer_time;
GstClockTimeDiff min_latency, max_latency;
@ -740,17 +776,19 @@ gst_pitch_process_segment (GstPitch * pitch, GstEvent ** event)
seg.start = 0;
seg.stop = -1;
seg.time = 0;
seg.position = 0;
seg.duration = -1;
}
/* Figure out how much of the incoming 'rate' we'll apply ourselves */
segment_applied_rate = seg.rate / output_rate;
/* update the output rate variables */
seg.rate = output_rate;
seg.applied_rate *= segment_applied_rate;
seg.rate = (segment_applied_rate < 0.0) ? -output_rate : output_rate;
seg.applied_rate *= ABS (segment_applied_rate);
GST_LOG_OBJECT (pitch->sinkpad, "in segment %" GST_SEGMENT_FORMAT, &seg);
stream_time_ratio = pitch->tempo * pitch->rate * segment_applied_rate;
stream_time_ratio = pitch->tempo * pitch->rate * ABS (segment_applied_rate);
if (stream_time_ratio == 0) {
GST_LOG_OBJECT (pitch->sinkpad, "stream_time_ratio is zero");
@ -761,14 +799,16 @@ gst_pitch_process_segment (GstPitch * pitch, GstEvent ** event)
GST_OBJECT_LOCK (pitch);
pitch->segment_applied_rate = segment_applied_rate;
priv->stream_time_ratio = stream_time_ratio;
priv->st->setTempo (pitch->tempo * pitch->segment_applied_rate);
priv->st->setTempo (pitch->tempo * ABS (pitch->segment_applied_rate));
GST_OBJECT_UNLOCK (pitch);
seg.start = (gint64) (seg.start / stream_time_ratio);
seg.position = (gint64) (seg.position / stream_time_ratio);
if (seg.stop != (guint64) - 1)
seg.stop = (gint64) (seg.stop / stream_time_ratio);
seg.time = (gint64) (seg.time / stream_time_ratio);
seg.position = (gint64) (seg.position / stream_time_ratio);
if (seg.duration != (guint64) - 1)
seg.duration = (gint64) (seg.duration / stream_time_ratio);
GST_LOG_OBJECT (pitch->sinkpad, "out segment %" GST_SEGMENT_FORMAT, &seg);
@ -861,8 +901,8 @@ gst_pitch_update_latency (GstPitch * pitch, GstClockTime timestamp)
(GstClockTimeDiff) (timestamp / priv->stream_time_ratio) -
pitch->next_buffer_time;
min_latency = MIN (pitch->min_latency, current_latency);
max_latency = MAX (pitch->max_latency, current_latency);
min_latency = MIN (pitch->min_latency, ABS (current_latency));
max_latency = MAX (pitch->max_latency, ABS (current_latency));
if (pitch->min_latency != min_latency || pitch->max_latency != max_latency) {
pitch->min_latency = min_latency;
@ -887,38 +927,42 @@ gst_pitch_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
GstPitch *pitch = GST_PITCH (parent);
GstPitchPrivate *priv = GST_PITCH_GET_PRIVATE (pitch);
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
GstClockTime timestamp = GST_BUFFER_PTS (buffer);
GST_OBJECT_LOCK (pitch);
GstClockTime next_buffer_time = pitch->next_buffer_time;
gfloat stream_time_ratio = priv->stream_time_ratio;
gint bytes_per_frame = pitch->info.bpf;
GstEvent *pending_segment = priv->pending_segment;
priv->pending_segment = NULL;
GST_OBJECT_UNLOCK (pitch);
// Remember the first time and corresponding offset
if (!GST_CLOCK_TIME_IS_VALID (next_buffer_time)) {
GstFormat out_format = GST_FORMAT_DEFAULT;
gint64 next_buffer_offset;
next_buffer_time = timestamp / stream_time_ratio;
gst_pitch_convert (pitch, GST_FORMAT_TIME, timestamp, &out_format,
&next_buffer_offset);
GST_OBJECT_LOCK (pitch);
pitch->next_buffer_time = next_buffer_time;
pitch->next_buffer_offset = next_buffer_offset;
GST_OBJECT_UNLOCK (pitch);
}
gst_object_sync_values (GST_OBJECT (pitch), next_buffer_time);
/* push the received samples on the soundtouch buffer */
GST_LOG_OBJECT (pitch, "incoming buffer (%d samples) %" GST_TIME_FORMAT,
(gint) (gst_buffer_get_size (buffer) / bytes_per_frame),
GST_TIME_ARGS (timestamp));
gboolean reverse_playback = (pitch->segment_applied_rate < 0);
if (reverse_playback) {
// In reverse playback we need to take the end of the buffer as the
// reference timestamp as the output buffer produced by libsoundtouch may
// have a duration different from this current input buffer.
timestamp += GST_BUFFER_DURATION (buffer);
}
gfloat stream_time_ratio = priv->stream_time_ratio;
GstClockTime next_buffer_time = pitch->next_buffer_time;
if (!GST_CLOCK_TIME_IS_VALID (next_buffer_time)) {
next_buffer_time = timestamp / stream_time_ratio;
pitch->next_buffer_time = next_buffer_time;
pitch->next_buffer_offset =
gst_util_uint64_scale_int (next_buffer_time, pitch->info.rate,
GST_SECOND);
}
GstEvent *pending_segment = priv->pending_segment;
priv->pending_segment = NULL;
GST_OBJECT_UNLOCK (pitch);
gst_object_sync_values (GST_OBJECT (pitch), next_buffer_time);
/* push the received samples on the soundtouch buffer */
if (pending_segment) {
GST_LOG_OBJECT (pitch, "processing stalled segment");
@ -936,11 +980,31 @@ gst_pitch_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
GstMapInfo info;
gst_buffer_map (buffer, &info, GST_MAP_READ);
guint8 *data = info.data;
gboolean free_data = FALSE;
if (reverse_playback && (gint) info.size > bytes_per_frame) {
// If playing backwards, we need to inverse the order of all samples before
// sending them to the libsoundtouch library.
data = (guint8 *) g_malloc (info.size);
free_data = TRUE;
guint8 *p = data;
for (gint i = info.size - bytes_per_frame; i >= 0; i -= bytes_per_frame) {
memcpy (p, info.data + i, bytes_per_frame);
p += bytes_per_frame;
}
}
GST_OBJECT_LOCK (pitch);
priv->st->putSamples ((soundtouch::SAMPLETYPE *) info.data,
priv->st->putSamples ((soundtouch::SAMPLETYPE *) data,
info.size / bytes_per_frame);
gboolean has_output_samples_available = !priv->st->isEmpty ();
GST_OBJECT_UNLOCK (pitch);
if (free_data)
g_free (data);
gst_buffer_unmap (buffer, &info);
gst_buffer_unref (buffer);

2
subprojects/gst-plugins-bad/ext/soundtouch/gstpitch.hh
View file

@ -73,7 +73,7 @@ struct _GstPitch
/* stream tracking */
gfloat segment_applied_rate; /* Rate applied from input segment */
GstClockTime next_buffer_time;
gint64 next_buffer_offset;
guint64 next_buffer_offset;
GstClockTimeDiff min_latency, max_latency;