gstreamer/subprojects/gst-plugins-bad/ext/soundtouch/gstpitch.cc
Thibault Saunier d82efb47aa pitch: Specify layout as required for negotiation
There are cases where it might negotiate 'non-interleaved' while it
is wrong.

```
gst-launch-1.0 audiotestsrc !  "audio/x-raw, format=(string)F32LE, layout=(string)non-interleaved" ! audioconvert ! audioresample ! pitch tempo=1.2 ! audioconvert ! "audio/x-raw,format=S16LE" ! fakesink

Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
(gst-launch-1.0:3029628): GStreamer-Audio-CRITICAL **: 11:42:22.477: gst_audio_buffer_map: assertion '(!meta && info->layout == GST_AUDIO_LAYOUT_INTERLEAVED) || (meta && info->layout == meta->info.layout)' failed
ERROR: from element /GstPipeline:pipeline0/GstAudioConvert:audioconvert1: The stream is in the wrong format.
Additional debug info:
../subprojects/gst-plugins-base/gst/audioconvert/gstaudioconvert.c(876): gst_audio_convert_transform (): /GstPipeline:pipeline0/GstAudioConvert:audioconvert1:
failed to map input buffer
ERROR: pipeline doesn't want to preroll.
ERROR: from element /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0: Internal data stream error.
Setting pipeline to NULL ...
Additional debug info:
../subprojects/gstreamer/libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0:
streaming stopped, reason error (-5)
ERROR: pipeline doesn't want to preroll.
Freeing pipeline ...
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1441>
2021-12-11 19:09:09 -03:00

944 lines
27 KiB
C++

/* GStreamer pitch controller element
* Copyright (C) 2006 Wouter Paesen <wouter@blue-gate.be>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
/* FIXME: workaround for SoundTouch.h of version 1.3.1 defining those
* variables while it shouldn't. */
#undef VERSION
#undef PACKAGE_VERSION
#undef PACKAGE_TARNAME
#undef PACKAGE_STRING
#undef PACKAGE_NAME
#undef PACKAGE_BUGREPORT
#undef PACKAGE
#include <soundtouch/SoundTouch.h>
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstpitch.hh"
#include <math.h>
GST_DEBUG_CATEGORY_STATIC (pitch_debug);
#define GST_CAT_DEFAULT pitch_debug
#define GST_PITCH_GET_PRIVATE(o) (o->priv)
struct _GstPitchPrivate
{
gfloat stream_time_ratio;
GstEvent *pending_segment;
soundtouch::SoundTouch * st;
};
enum
{
ARG_0,
ARG_OUT_RATE,
ARG_RATE,
ARG_TEMPO,
ARG_PITCH
};
/* For soundtouch 1.4 */
#if defined(INTEGER_SAMPLES)
#define SOUNDTOUCH_INTEGER_SAMPLES 1
#elif defined(FLOAT_SAMPLES)
#define SOUNDTOUCH_FLOAT_SAMPLES 1
#endif
#if defined(SOUNDTOUCH_FLOAT_SAMPLES)
#define SUPPORTED_CAPS \
"audio/x-raw, " \
"format = (string) " GST_AUDIO_NE (F32) ", " \
"rate = (int) [ 8000, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"layout = (string) interleaved"
#elif defined(SOUNDTOUCH_INTEGER_SAMPLES)
#define SUPPORTED_CAPS \
"audio/x-raw, " \
"format = (string) " GST_AUDIO_NE (S16) ", " \
"rate = (int) [ 8000, MAX ], " \
"channels = (int) [ 1, MAX ]", \
"layout = (string) interleaved"
#else
#error "Only integer or float samples are supported"
#endif
static GstStaticPadTemplate gst_pitch_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SUPPORTED_CAPS));
static GstStaticPadTemplate gst_pitch_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SUPPORTED_CAPS));
static void gst_pitch_dispose (GObject * object);
static void gst_pitch_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_pitch_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_pitch_setcaps (GstPitch * pitch, GstCaps * caps);
static GstFlowReturn gst_pitch_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
static GstStateChangeReturn gst_pitch_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_pitch_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_pitch_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_pitch_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
#define gst_pitch_parent_class parent_class
G_DEFINE_TYPE_WITH_PRIVATE (GstPitch, gst_pitch, GST_TYPE_ELEMENT);
GST_ELEMENT_REGISTER_DEFINE (pitch, "pitch", GST_RANK_NONE,
GST_TYPE_PITCH);
static void
gst_pitch_class_init (GstPitchClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *element_class;
gobject_class = G_OBJECT_CLASS (klass);
element_class = GST_ELEMENT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (pitch_debug, "pitch", 0,
"audio pitch control element");
gobject_class->set_property = gst_pitch_set_property;
gobject_class->get_property = gst_pitch_get_property;
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_pitch_dispose);
g_object_class_install_property (gobject_class, ARG_PITCH,
g_param_spec_float ("pitch", "Pitch",
"Audio stream pitch", 0.1, 10.0, 1.0,
(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (gobject_class, ARG_TEMPO,
g_param_spec_float ("tempo", "Tempo",
"Audio stream tempo", 0.1, 10.0, 1.0,
(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (gobject_class, ARG_RATE,
g_param_spec_float ("rate", "Rate",
"Audio stream rate", 0.1, 10.0, 1.0,
(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (gobject_class, ARG_OUT_RATE,
g_param_spec_float ("output-rate", "Output Rate",
"Output rate on downstream segment events", 0.1, 10.0, 1.0,
(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
G_PARAM_STATIC_STRINGS)));
element_class->change_state = GST_DEBUG_FUNCPTR (gst_pitch_change_state);
gst_element_class_add_static_pad_template (element_class, &gst_pitch_src_template);
gst_element_class_add_static_pad_template (element_class, &gst_pitch_sink_template);
gst_element_class_set_static_metadata (element_class, "Pitch controller",
"Filter/Effect/Audio", "Control the pitch of an audio stream",
"Wouter Paesen <wouter@blue-gate.be>");
}
static void
gst_pitch_init (GstPitch * pitch)
{
pitch->priv = (GstPitchPrivate *) gst_pitch_get_instance_private (pitch);
pitch->sinkpad =
gst_pad_new_from_static_template (&gst_pitch_sink_template, "sink");
gst_pad_set_chain_function (pitch->sinkpad,
GST_DEBUG_FUNCPTR (gst_pitch_chain));
gst_pad_set_event_function (pitch->sinkpad,
GST_DEBUG_FUNCPTR (gst_pitch_sink_event));
GST_PAD_SET_PROXY_CAPS (pitch->sinkpad);
gst_element_add_pad (GST_ELEMENT (pitch), pitch->sinkpad);
pitch->srcpad =
gst_pad_new_from_static_template (&gst_pitch_src_template, "src");
gst_pad_set_event_function (pitch->srcpad,
GST_DEBUG_FUNCPTR (gst_pitch_src_event));
gst_pad_set_query_function (pitch->srcpad,
GST_DEBUG_FUNCPTR (gst_pitch_src_query));
GST_PAD_SET_PROXY_CAPS (pitch->sinkpad);
gst_element_add_pad (GST_ELEMENT (pitch), pitch->srcpad);
pitch->priv->st = new soundtouch::SoundTouch ();
pitch->tempo = 1.0;
pitch->rate = 1.0;
pitch->out_seg_rate = 1.0;
pitch->seg_arate = 1.0;
pitch->pitch = 1.0;
pitch->next_buffer_time = GST_CLOCK_TIME_NONE;
pitch->next_buffer_offset = 0;
pitch->priv->st->setRate (pitch->rate);
pitch->priv->st->setTempo (pitch->tempo * pitch->seg_arate);
pitch->priv->st->setPitch (pitch->pitch);
pitch->priv->stream_time_ratio = 1.0;
pitch->min_latency = pitch->max_latency = 0;
}
static void
gst_pitch_dispose (GObject * object)
{
GstPitch *pitch = GST_PITCH (object);
if (pitch->priv->st) {
delete pitch->priv->st;
pitch->priv->st = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_pitch_update_duration (GstPitch * pitch)
{
GstMessage *m;
m = gst_message_new_duration_changed (GST_OBJECT (pitch));
gst_element_post_message (GST_ELEMENT (pitch), m);
}
static void
gst_pitch_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstPitch *pitch = GST_PITCH (object);
GST_OBJECT_LOCK (pitch);
switch (prop_id) {
case ARG_TEMPO:
pitch->tempo = g_value_get_float (value);
pitch->priv->stream_time_ratio =
pitch->tempo * pitch->rate * pitch->seg_arate;
pitch->priv->st->setTempo (pitch->tempo * pitch->seg_arate);
GST_OBJECT_UNLOCK (pitch);
gst_pitch_update_duration (pitch);
break;
case ARG_RATE:
pitch->rate = g_value_get_float (value);
pitch->priv->stream_time_ratio =
pitch->tempo * pitch->rate * pitch->seg_arate;
pitch->priv->st->setRate (pitch->rate);
GST_OBJECT_UNLOCK (pitch);
gst_pitch_update_duration (pitch);
break;
case ARG_OUT_RATE:
/* Has no effect until the next input segment */
pitch->out_seg_rate = g_value_get_float (value);
GST_OBJECT_UNLOCK (pitch);
break;
case ARG_PITCH:
pitch->pitch = g_value_get_float (value);
pitch->priv->st->setPitch (pitch->pitch);
GST_OBJECT_UNLOCK (pitch);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
GST_OBJECT_UNLOCK (pitch);
break;
}
}
static void
gst_pitch_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstPitch *pitch = GST_PITCH (object);
GST_OBJECT_LOCK (pitch);
switch (prop_id) {
case ARG_TEMPO:
g_value_set_float (value, pitch->tempo);
break;
case ARG_RATE:
g_value_set_float (value, pitch->rate);
break;
case ARG_OUT_RATE:
g_value_set_float (value, pitch->out_seg_rate);
break;
case ARG_PITCH:
g_value_set_float (value, pitch->pitch);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (pitch);
}
static gboolean
gst_pitch_setcaps (GstPitch * pitch, GstCaps * caps)
{
GstPitchPrivate *priv;
priv = GST_PITCH_GET_PRIVATE (pitch);
if (!gst_audio_info_from_caps (&pitch->info, caps))
return FALSE;
GST_OBJECT_LOCK (pitch);
/* notify the soundtouch instance of this change */
priv->st->setSampleRate (pitch->info.rate);
priv->st->setChannels (pitch->info.channels);
GST_OBJECT_UNLOCK (pitch);
return TRUE;
}
/* send a buffer out */
static GstFlowReturn
gst_pitch_forward_buffer (GstPitch * pitch, GstBuffer * buffer)
{
gint samples;
GST_BUFFER_TIMESTAMP (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_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);
return gst_pad_push (pitch->srcpad, buffer);
}
/* extract a buffer from soundtouch */
static GstBuffer *
gst_pitch_prepare_buffer (GstPitch * pitch)
{
GstPitchPrivate *priv;
guint samples;
GstBuffer *buffer;
GstMapInfo info;
priv = GST_PITCH_GET_PRIVATE (pitch);
GST_LOG_OBJECT (pitch, "preparing buffer");
samples = pitch->priv->st->numSamples ();
if (samples == 0)
return NULL;
buffer = gst_buffer_new_and_alloc (samples * pitch->info.bpf);
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_unref (buffer);
return NULL;
}
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;
return buffer;
}
/* process the last samples, in a later stage we should make sure no more
* samples are sent out here as strictly necessary, because soundtouch could
* append zero samples, which could disturb looping. */
static GstFlowReturn
gst_pitch_flush_buffer (GstPitch * pitch, gboolean send)
{
GstBuffer *buffer;
if (pitch->priv->st->numUnprocessedSamples() != 0) {
GST_DEBUG_OBJECT (pitch, "flushing buffer");
pitch->priv->st->flush ();
}
if (!send)
return GST_FLOW_OK;
buffer = gst_pitch_prepare_buffer (pitch);
if (!buffer)
return GST_FLOW_OK;
return gst_pitch_forward_buffer (pitch, buffer);
}
static gboolean
gst_pitch_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstPitch *pitch;
gboolean res;
pitch = GST_PITCH (parent);
GST_DEBUG_OBJECT (pad, "received %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:{
/* transform the event upstream, according to the playback rate */
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gfloat stream_time_ratio;
guint32 seqnum;
GST_OBJECT_LOCK (pitch);
stream_time_ratio = pitch->priv->stream_time_ratio;
GST_OBJECT_UNLOCK (pitch);
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
seqnum = gst_event_get_seqnum (event);
gst_event_unref (event);
if (format == GST_FORMAT_TIME || format == GST_FORMAT_DEFAULT) {
cur = (gint64) (cur * stream_time_ratio);
if (stop != -1)
stop = (gint64) (stop * stream_time_ratio);
event = gst_event_new_seek (rate, format, flags,
cur_type, cur, stop_type, stop);
gst_event_set_seqnum (event, seqnum);
res = gst_pad_event_default (pad, parent, event);
} else {
GST_WARNING_OBJECT (pitch,
"Seeking only supported in TIME or DEFAULT format");
res = FALSE;
}
break;
}
default:
res = gst_pad_event_default (pad, parent, event);
break;
}
return res;
}
/* generic convert function based on caps, no rate
* used here
*/
static gboolean
gst_pitch_convert (GstPitch * pitch,
GstFormat src_format, gint64 src_value,
GstFormat * dst_format, gint64 * dst_value)
{
gboolean res = TRUE;
guint sample_size;
gint samplerate;
g_return_val_if_fail (dst_format && dst_value, FALSE);
GST_OBJECT_LOCK (pitch);
sample_size = pitch->info.bpf;
samplerate = pitch->info.rate;
GST_OBJECT_UNLOCK (pitch);
if (sample_size == 0 || samplerate == 0) {
return FALSE;
}
if (src_format == *dst_format || src_value == -1) {
*dst_value = src_value;
return TRUE;
}
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dst_format) {
case GST_FORMAT_TIME:
*dst_value =
gst_util_uint64_scale_int (src_value, GST_SECOND,
sample_size * samplerate);
break;
case GST_FORMAT_DEFAULT:
*dst_value = gst_util_uint64_scale_int (src_value, 1, sample_size);
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_TIME:
switch (*dst_format) {
case GST_FORMAT_BYTES:
*dst_value =
gst_util_uint64_scale_int (src_value, samplerate * sample_size,
GST_SECOND);
break;
case GST_FORMAT_DEFAULT:
*dst_value =
gst_util_uint64_scale_int (src_value, samplerate, GST_SECOND);
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_DEFAULT:
switch (*dst_format) {
case GST_FORMAT_BYTES:
*dst_value = gst_util_uint64_scale_int (src_value, sample_size, 1);
break;
case GST_FORMAT_TIME:
*dst_value =
gst_util_uint64_scale_int (src_value, GST_SECOND, samplerate);
break;
default:
res = FALSE;
break;
}
break;
default:
res = FALSE;
break;
}
return res;
}
static gboolean
gst_pitch_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstPitch *pitch;
gboolean res = FALSE;
gfloat stream_time_ratio;
gint64 next_buffer_offset;
GstClockTime next_buffer_time;
pitch = GST_PITCH (parent);
GST_LOG ("%s query", GST_QUERY_TYPE_NAME (query));
GST_OBJECT_LOCK (pitch);
stream_time_ratio = pitch->priv->stream_time_ratio;
next_buffer_time = pitch->next_buffer_time;
next_buffer_offset = pitch->next_buffer_offset;
GST_OBJECT_UNLOCK (pitch);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:{
GstFormat format;
gint64 duration;
if (!gst_pad_query_default (pad, parent, query)) {
GST_DEBUG_OBJECT (pitch, "upstream provided no duration");
break;
}
gst_query_parse_duration (query, &format, &duration);
if (format != GST_FORMAT_TIME && format != GST_FORMAT_DEFAULT) {
GST_DEBUG_OBJECT (pitch, "not TIME or DEFAULT format");
break;
}
GST_LOG_OBJECT (pitch, "upstream duration: %" G_GINT64_FORMAT, duration);
duration = (gint64) (duration / stream_time_ratio);
GST_LOG_OBJECT (pitch, "our duration: %" G_GINT64_FORMAT, duration);
gst_query_set_duration (query, format, duration);
res = TRUE;
break;
}
case GST_QUERY_POSITION:{
GstFormat dst_format;
gint64 dst_value;
gst_query_parse_position (query, &dst_format, &dst_value);
if (dst_format != GST_FORMAT_TIME && dst_format != GST_FORMAT_DEFAULT) {
GST_DEBUG_OBJECT (pitch, "not TIME or DEFAULT format");
break;
}
if (dst_format == GST_FORMAT_TIME) {
dst_value = next_buffer_time;
res = TRUE;
} else {
dst_value = next_buffer_offset;
res = TRUE;
}
if (res) {
GST_LOG_OBJECT (pitch, "our position: %" G_GINT64_FORMAT, dst_value);
gst_query_set_position (query, dst_format, dst_value);
}
break;
}
case GST_QUERY_CONVERT:{
GstFormat src_format, dst_format;
gint64 src_value, dst_value;
gst_query_parse_convert (query, &src_format, &src_value,
&dst_format, NULL);
res = gst_pitch_convert (pitch, src_format, src_value,
&dst_format, &dst_value);
if (res) {
gst_query_set_convert (query, src_format, src_value,
dst_format, dst_value);
}
break;
}
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
GstPad *peer;
if ((peer = gst_pad_get_peer (pitch->sinkpad))) {
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG ("Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
GST_DEBUG ("Our latency: min %" GST_TIME_FORMAT
", max %" GST_TIME_FORMAT,
GST_TIME_ARGS (pitch->min_latency),
GST_TIME_ARGS (pitch->max_latency));
min += pitch->min_latency;
if (max != GST_CLOCK_TIME_NONE)
max += pitch->max_latency;
GST_DEBUG ("Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
gst_object_unref (peer);
}
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
/* this function returns FALSE if not enough data is known to transform the
* segment into proper downstream values. If the function does return false
* the segment should be stalled until enough information is available.
* If the function returns TRUE, event will be replaced by the new downstream
* compatible event.
*/
static gboolean
gst_pitch_process_segment (GstPitch * pitch, GstEvent ** event)
{
gint seqnum;
gdouble out_seg_rate, our_arate;
gfloat stream_time_ratio;
GstSegment seg;
g_return_val_if_fail (event, FALSE);
GST_OBJECT_LOCK (pitch);
stream_time_ratio = pitch->priv->stream_time_ratio;
out_seg_rate = pitch->out_seg_rate;
GST_OBJECT_UNLOCK (pitch);
gst_event_copy_segment (*event, &seg);
if (seg.format != GST_FORMAT_TIME && seg.format != GST_FORMAT_DEFAULT) {
GST_WARNING_OBJECT (pitch,
"Only NEWSEGMENT in TIME or DEFAULT format supported, sending"
"open ended NEWSEGMENT in TIME format.");
seg.format = GST_FORMAT_TIME;
seg.start = 0;
seg.stop = -1;
seg.time = 0;
}
/* Figure out how much of the incoming 'rate' we'll apply ourselves */
our_arate = seg.rate / out_seg_rate;
/* update the output rate variables */
seg.rate = out_seg_rate;
seg.applied_rate *= our_arate;
GST_LOG_OBJECT (pitch->sinkpad, "in segment %" GST_SEGMENT_FORMAT, &seg);
stream_time_ratio = pitch->tempo * pitch->rate * pitch->seg_arate;
if (stream_time_ratio == 0) {
GST_LOG_OBJECT (pitch->sinkpad, "stream_time_ratio is zero");
return FALSE;
}
/* Update the playback rate */
GST_OBJECT_LOCK (pitch);
pitch->seg_arate = our_arate;
pitch->priv->stream_time_ratio = stream_time_ratio;
pitch->priv->st->setTempo (pitch->tempo * pitch->seg_arate);
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);
GST_LOG_OBJECT (pitch->sinkpad, "out segment %" GST_SEGMENT_FORMAT, &seg);
seqnum = gst_event_get_seqnum (*event);
gst_event_unref (*event);
*event = gst_event_new_segment (&seg);
gst_event_set_seqnum (*event, seqnum);
return TRUE;
}
static gboolean
gst_pitch_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
gboolean res = TRUE;
GstPitch *pitch;
pitch = GST_PITCH (parent);
GST_LOG_OBJECT (pad, "received %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_pitch_flush_buffer (pitch, FALSE);
pitch->priv->st->clear ();
pitch->next_buffer_offset = 0;
pitch->next_buffer_time = GST_CLOCK_TIME_NONE;
pitch->min_latency = pitch->max_latency = 0;
break;
case GST_EVENT_EOS:
gst_pitch_flush_buffer (pitch, TRUE);
pitch->priv->st->clear ();
pitch->min_latency = pitch->max_latency = 0;
break;
case GST_EVENT_SEGMENT:
if (!gst_pitch_process_segment (pitch, &event)) {
GST_LOG_OBJECT (pad, "not enough data known, stalling segment");
if (GST_PITCH_GET_PRIVATE (pitch)->pending_segment)
gst_event_unref (GST_PITCH_GET_PRIVATE (pitch)->pending_segment);
GST_PITCH_GET_PRIVATE (pitch)->pending_segment = event;
event = NULL;
}
pitch->priv->st->clear ();
pitch->min_latency = pitch->max_latency = 0;
break;
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
res = gst_pitch_setcaps (pitch, caps);
if (!res) {
gst_event_unref (event);
goto done;
}
}
default:
break;
}
/* and forward it */
if (event)
res = gst_pad_event_default (pad, parent, event);
done:
return res;
}
static void
gst_pitch_update_latency (GstPitch * pitch, GstClockTime timestamp)
{
GstClockTimeDiff current_latency, min_latency, max_latency;
current_latency =
(GstClockTimeDiff) (timestamp / pitch->priv->stream_time_ratio) -
pitch->next_buffer_time;
min_latency = MIN (pitch->min_latency, current_latency);
max_latency = MAX (pitch->max_latency, current_latency);
if (pitch->min_latency != min_latency || pitch->max_latency != max_latency) {
pitch->min_latency = min_latency;
pitch->max_latency = max_latency;
/* FIXME: what about the LATENCY event? It only has
* one latency value, should it be current, min or max?
* Should it include upstream latencies?
*/
gst_element_post_message (GST_ELEMENT (pitch),
gst_message_new_latency (GST_OBJECT (pitch)));
}
}
static GstFlowReturn
gst_pitch_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstPitch *pitch;
GstPitchPrivate *priv;
GstClockTime timestamp;
GstMapInfo info;
pitch = GST_PITCH (parent);
priv = GST_PITCH_GET_PRIVATE (pitch);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
// Remember the first time and corresponding offset
if (!GST_CLOCK_TIME_IS_VALID (pitch->next_buffer_time)) {
gfloat stream_time_ratio;
GstFormat out_format = GST_FORMAT_DEFAULT;
GST_OBJECT_LOCK (pitch);
stream_time_ratio = priv->stream_time_ratio;
GST_OBJECT_UNLOCK (pitch);
pitch->next_buffer_time = timestamp / stream_time_ratio;
gst_pitch_convert (pitch, GST_FORMAT_TIME, timestamp, &out_format,
&pitch->next_buffer_offset);
}
gst_object_sync_values (GST_OBJECT (pitch), 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) / pitch->info.bpf),
GST_TIME_ARGS (timestamp));
if (GST_PITCH_GET_PRIVATE (pitch)->pending_segment) {
GstEvent *event =
gst_event_copy (GST_PITCH_GET_PRIVATE (pitch)->pending_segment);
GST_LOG_OBJECT (pitch, "processing stalled segment");
if (!gst_pitch_process_segment (pitch, &event)) {
gst_buffer_unref (buffer);
gst_event_unref (event);
return GST_FLOW_ERROR;
}
if (!gst_pad_event_default (pitch->sinkpad, parent, event)) {
gst_buffer_unref (buffer);
gst_event_unref (event);
return GST_FLOW_ERROR;
}
gst_event_unref (GST_PITCH_GET_PRIVATE (pitch)->pending_segment);
GST_PITCH_GET_PRIVATE (pitch)->pending_segment = NULL;
}
gst_buffer_map (buffer, &info, GST_MAP_READ);
GST_OBJECT_LOCK (pitch);
priv->st->putSamples ((soundtouch::SAMPLETYPE *) info.data, info.size / pitch->info.bpf);
GST_OBJECT_UNLOCK (pitch);
gst_buffer_unmap (buffer, &info);
gst_buffer_unref (buffer);
/* Calculate latency */
gst_pitch_update_latency (pitch, timestamp);
/* and try to extract some samples from the soundtouch buffer */
if (!priv->st->isEmpty ()) {
GstBuffer *out_buffer;
out_buffer = gst_pitch_prepare_buffer (pitch);
if (out_buffer)
return gst_pitch_forward_buffer (pitch, out_buffer);
}
return GST_FLOW_OK;
}
static GstStateChangeReturn
gst_pitch_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstPitch *pitch = GST_PITCH (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
pitch->next_buffer_time = GST_CLOCK_TIME_NONE;
pitch->next_buffer_offset = 0;
pitch->priv->st->clear ();
pitch->min_latency = pitch->max_latency = 0;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret != GST_STATE_CHANGE_SUCCESS)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (GST_PITCH_GET_PRIVATE (pitch)->pending_segment) {
gst_event_unref (GST_PITCH_GET_PRIVATE (pitch)->pending_segment);
GST_PITCH_GET_PRIVATE (pitch)->pending_segment = NULL;
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
default:
break;
}
return ret;
}