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pitch: make it work with reverse playback
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
This commit is contained in:
Loïc Le Page
GStreamer Marge Bot
parent
35165da586
commit
8f5325ed11
2 changed files with 123 additions and 59 deletions
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@ -265,8 +265,8 @@ gst_pitch_set_property (GObject * object, guint prop_id,
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GST_OBJECT_LOCK (pitch);
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pitch->tempo = g_value_get_float (value);
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priv->stream_time_ratio =
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pitch->tempo * pitch->rate * pitch->segment_applied_rate;
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priv->st->setTempo (pitch->tempo * pitch->segment_applied_rate);
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pitch->tempo * pitch->rate * ABS (pitch->segment_applied_rate);
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priv->st->setTempo (pitch->tempo * ABS (pitch->segment_applied_rate));
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GST_OBJECT_UNLOCK (pitch);
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gst_pitch_update_duration (pitch);
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break;
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@ -274,7 +274,7 @@ gst_pitch_set_property (GObject * object, guint prop_id,
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GST_OBJECT_LOCK (pitch);
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pitch->rate = g_value_get_float (value);
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priv->stream_time_ratio =
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pitch->tempo * pitch->rate * pitch->segment_applied_rate;
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pitch->tempo * pitch->rate * ABS (pitch->segment_applied_rate);
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priv->st->setRate (pitch->rate);
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GST_OBJECT_UNLOCK (pitch);
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gst_pitch_update_duration (pitch);
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@ -353,24 +353,38 @@ gst_pitch_setcaps (GstPitch * pitch, GstCaps * caps)
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static GstFlowReturn
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gst_pitch_forward_buffer (GstPitch * pitch, GstBuffer * buffer)
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{
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gint samples;
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guint64 samples = GST_BUFFER_OFFSET (buffer);
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GST_OBJECT_LOCK (pitch);
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GST_BUFFER_TIMESTAMP (buffer) = pitch->next_buffer_time;
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pitch->next_buffer_time += GST_BUFFER_DURATION (buffer);
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if (pitch->segment_applied_rate >= 0) {
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GST_BUFFER_PTS (buffer) = pitch->next_buffer_time;
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pitch->next_buffer_time += GST_BUFFER_DURATION (buffer);
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samples = GST_BUFFER_OFFSET (buffer);
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GST_BUFFER_OFFSET (buffer) = pitch->next_buffer_offset;
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pitch->next_buffer_offset += samples;
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GST_BUFFER_OFFSET_END (buffer) = pitch->next_buffer_offset;
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GST_BUFFER_OFFSET (buffer) = pitch->next_buffer_offset;
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pitch->next_buffer_offset += samples;
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GST_BUFFER_OFFSET_END (buffer) = pitch->next_buffer_offset;
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} else {
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if (pitch->next_buffer_time > GST_BUFFER_DURATION (buffer))
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pitch->next_buffer_time -= GST_BUFFER_DURATION (buffer);
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else
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pitch->next_buffer_time = 0;
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GST_BUFFER_PTS (buffer) = pitch->next_buffer_time;
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GST_LOG ("pushing buffer [%" GST_TIME_FORMAT "]-[%" GST_TIME_FORMAT
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"] (%d samples)", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
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GST_TIME_ARGS (pitch->next_buffer_time), samples);
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GST_BUFFER_OFFSET_END (buffer) = pitch->next_buffer_offset;
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if (pitch->next_buffer_offset > samples)
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pitch->next_buffer_offset -= samples;
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else
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pitch->next_buffer_offset = 0;
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GST_BUFFER_OFFSET (buffer) = pitch->next_buffer_offset;
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}
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GST_OBJECT_UNLOCK (pitch);
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GST_LOG ("pushing buffer pts: %" GST_TIME_FORMAT ", duration: %"
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GST_TIME_FORMAT ", samples: %" G_GUINT64_FORMAT, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)), samples);
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return gst_pad_push (pitch->srcpad, buffer);
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}
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@ -378,37 +392,59 @@ gst_pitch_forward_buffer (GstPitch * pitch, GstBuffer * buffer)
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static GstBuffer *
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gst_pitch_prepare_buffer (GstPitch * pitch)
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{
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GstMapInfo info;
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GstBuffer *buffer = NULL;
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GST_LOG_OBJECT (pitch, "preparing buffer");
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GST_OBJECT_LOCK (pitch);
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gboolean reverse_playback = (pitch->segment_applied_rate < 0);
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gint rate = pitch->info.rate;
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gint bytes_per_frame = pitch->info.bpf;
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GstPitchPrivate *priv = GST_PITCH_GET_PRIVATE (pitch);
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guint samples = priv->st->numSamples ();
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if (samples > 0) {
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buffer = gst_buffer_new_and_alloc (samples * pitch->info.bpf);
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GstMapInfo info;
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buffer = gst_buffer_new_and_alloc (samples * bytes_per_frame);
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gst_buffer_map (buffer, &info, (GstMapFlags) GST_MAP_READWRITE);
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samples =
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priv->st->receiveSamples ((soundtouch::SAMPLETYPE *) info.data,
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samples);
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gst_buffer_unmap (buffer, &info);
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if (samples > 0) {
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GST_BUFFER_DURATION (buffer) =
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gst_util_uint64_scale (samples, GST_SECOND, pitch->info.rate);
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/* temporary store samples here, to avoid having to recalculate this */
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GST_BUFFER_OFFSET (buffer) = (gint64) samples;
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} else {
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if (samples <= 0) {
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gst_buffer_unmap (buffer, &info);
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gst_buffer_unref (buffer);
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buffer = NULL;
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}
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}
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GST_OBJECT_UNLOCK (pitch);
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if (buffer) {
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if (reverse_playback && samples > 1) {
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// If playing backwards, we need to inverse the order of all received samples
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guint8 *dest = info.data;
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guint8 val = 0;
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guint half_samples = samples >> 1;
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for (guint i = samples - 1; i >= half_samples; --i) {
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guint8 *src = info.data + i * bytes_per_frame;
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if (src == dest)
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break;
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for (gint j = 0; j < bytes_per_frame; ++j) {
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val = *dest;
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*dest++ = *src;
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*src++ = val;
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}
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}
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}
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gst_buffer_unmap (buffer, &info);
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GST_BUFFER_DURATION (buffer) =
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gst_util_uint64_scale (samples, GST_SECOND, rate);
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/* temporary store samples here, to avoid having to recalculate this */
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GST_BUFFER_OFFSET (buffer) = (gint64) samples;
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}
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return buffer;
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}
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@ -584,7 +620,7 @@ gst_pitch_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
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GstPitch *pitch;
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gboolean res = FALSE;
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gfloat stream_time_ratio;
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gint64 next_buffer_offset;
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guint64 next_buffer_offset;
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GstClockTime next_buffer_time;
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GstClockTimeDiff min_latency, max_latency;
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@ -740,17 +776,19 @@ gst_pitch_process_segment (GstPitch * pitch, GstEvent ** event)
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seg.start = 0;
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seg.stop = -1;
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seg.time = 0;
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seg.position = 0;
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seg.duration = -1;
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}
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/* Figure out how much of the incoming 'rate' we'll apply ourselves */
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segment_applied_rate = seg.rate / output_rate;
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/* update the output rate variables */
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seg.rate = output_rate;
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seg.applied_rate *= segment_applied_rate;
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seg.rate = (segment_applied_rate < 0.0) ? -output_rate : output_rate;
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seg.applied_rate *= ABS (segment_applied_rate);
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GST_LOG_OBJECT (pitch->sinkpad, "in segment %" GST_SEGMENT_FORMAT, &seg);
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stream_time_ratio = pitch->tempo * pitch->rate * segment_applied_rate;
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stream_time_ratio = pitch->tempo * pitch->rate * ABS (segment_applied_rate);
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if (stream_time_ratio == 0) {
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GST_LOG_OBJECT (pitch->sinkpad, "stream_time_ratio is zero");
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@ -761,14 +799,16 @@ gst_pitch_process_segment (GstPitch * pitch, GstEvent ** event)
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GST_OBJECT_LOCK (pitch);
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pitch->segment_applied_rate = segment_applied_rate;
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priv->stream_time_ratio = stream_time_ratio;
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priv->st->setTempo (pitch->tempo * pitch->segment_applied_rate);
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priv->st->setTempo (pitch->tempo * ABS (pitch->segment_applied_rate));
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GST_OBJECT_UNLOCK (pitch);
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seg.start = (gint64) (seg.start / stream_time_ratio);
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seg.position = (gint64) (seg.position / stream_time_ratio);
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if (seg.stop != (guint64) - 1)
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seg.stop = (gint64) (seg.stop / stream_time_ratio);
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seg.time = (gint64) (seg.time / stream_time_ratio);
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seg.position = (gint64) (seg.position / stream_time_ratio);
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if (seg.duration != (guint64) - 1)
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seg.duration = (gint64) (seg.duration / stream_time_ratio);
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GST_LOG_OBJECT (pitch->sinkpad, "out segment %" GST_SEGMENT_FORMAT, &seg);
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@ -861,8 +901,8 @@ gst_pitch_update_latency (GstPitch * pitch, GstClockTime timestamp)
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(GstClockTimeDiff) (timestamp / priv->stream_time_ratio) -
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pitch->next_buffer_time;
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min_latency = MIN (pitch->min_latency, current_latency);
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max_latency = MAX (pitch->max_latency, current_latency);
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min_latency = MIN (pitch->min_latency, ABS (current_latency));
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max_latency = MAX (pitch->max_latency, ABS (current_latency));
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if (pitch->min_latency != min_latency || pitch->max_latency != max_latency) {
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pitch->min_latency = min_latency;
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@ -887,38 +927,42 @@ gst_pitch_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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GstPitch *pitch = GST_PITCH (parent);
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GstPitchPrivate *priv = GST_PITCH_GET_PRIVATE (pitch);
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GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
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GstClockTime timestamp = GST_BUFFER_PTS (buffer);
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GST_OBJECT_LOCK (pitch);
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GstClockTime next_buffer_time = pitch->next_buffer_time;
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gfloat stream_time_ratio = priv->stream_time_ratio;
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gint bytes_per_frame = pitch->info.bpf;
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GstEvent *pending_segment = priv->pending_segment;
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priv->pending_segment = NULL;
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GST_OBJECT_UNLOCK (pitch);
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// Remember the first time and corresponding offset
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if (!GST_CLOCK_TIME_IS_VALID (next_buffer_time)) {
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GstFormat out_format = GST_FORMAT_DEFAULT;
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gint64 next_buffer_offset;
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next_buffer_time = timestamp / stream_time_ratio;
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gst_pitch_convert (pitch, GST_FORMAT_TIME, timestamp, &out_format,
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&next_buffer_offset);
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GST_OBJECT_LOCK (pitch);
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pitch->next_buffer_time = next_buffer_time;
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pitch->next_buffer_offset = next_buffer_offset;
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GST_OBJECT_UNLOCK (pitch);
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}
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gst_object_sync_values (GST_OBJECT (pitch), next_buffer_time);
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/* push the received samples on the soundtouch buffer */
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GST_LOG_OBJECT (pitch, "incoming buffer (%d samples) %" GST_TIME_FORMAT,
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(gint) (gst_buffer_get_size (buffer) / bytes_per_frame),
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GST_TIME_ARGS (timestamp));
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gboolean reverse_playback = (pitch->segment_applied_rate < 0);
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if (reverse_playback) {
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// In reverse playback we need to take the end of the buffer as the
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// reference timestamp as the output buffer produced by libsoundtouch may
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// have a duration different from this current input buffer.
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timestamp += GST_BUFFER_DURATION (buffer);
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}
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gfloat stream_time_ratio = priv->stream_time_ratio;
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GstClockTime next_buffer_time = pitch->next_buffer_time;
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if (!GST_CLOCK_TIME_IS_VALID (next_buffer_time)) {
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next_buffer_time = timestamp / stream_time_ratio;
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pitch->next_buffer_time = next_buffer_time;
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pitch->next_buffer_offset =
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gst_util_uint64_scale_int (next_buffer_time, pitch->info.rate,
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GST_SECOND);
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}
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GstEvent *pending_segment = priv->pending_segment;
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priv->pending_segment = NULL;
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GST_OBJECT_UNLOCK (pitch);
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gst_object_sync_values (GST_OBJECT (pitch), next_buffer_time);
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/* push the received samples on the soundtouch buffer */
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if (pending_segment) {
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GST_LOG_OBJECT (pitch, "processing stalled segment");
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@ -936,11 +980,31 @@ gst_pitch_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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GstMapInfo info;
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gst_buffer_map (buffer, &info, GST_MAP_READ);
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guint8 *data = info.data;
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gboolean free_data = FALSE;
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if (reverse_playback && (gint) info.size > bytes_per_frame) {
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// If playing backwards, we need to inverse the order of all samples before
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// sending them to the libsoundtouch library.
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data = (guint8 *) g_malloc (info.size);
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free_data = TRUE;
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guint8 *p = data;
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for (gint i = info.size - bytes_per_frame; i >= 0; i -= bytes_per_frame) {
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memcpy (p, info.data + i, bytes_per_frame);
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p += bytes_per_frame;
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}
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}
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GST_OBJECT_LOCK (pitch);
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priv->st->putSamples ((soundtouch::SAMPLETYPE *) info.data,
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priv->st->putSamples ((soundtouch::SAMPLETYPE *) data,
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info.size / bytes_per_frame);
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gboolean has_output_samples_available = !priv->st->isEmpty ();
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GST_OBJECT_UNLOCK (pitch);
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if (free_data)
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g_free (data);
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gst_buffer_unmap (buffer, &info);
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gst_buffer_unref (buffer);
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@ -73,7 +73,7 @@ struct _GstPitch
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/* stream tracking */
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gfloat segment_applied_rate; /* Rate applied from input segment */
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GstClockTime next_buffer_time;
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gint64 next_buffer_offset;
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guint64 next_buffer_offset;
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GstClockTimeDiff min_latency, max_latency;
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