mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-05 16:06:43 +00:00
d82efb47aa
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>
944 lines
27 KiB
C++
944 lines
27 KiB
C++
/* GStreamer pitch controller element
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* Copyright (C) 2006 Wouter Paesen <wouter@blue-gate.be>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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/* FIXME: workaround for SoundTouch.h of version 1.3.1 defining those
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* variables while it shouldn't. */
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#undef VERSION
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#undef PACKAGE_VERSION
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#undef PACKAGE_TARNAME
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#undef PACKAGE_STRING
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#undef PACKAGE_NAME
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#undef PACKAGE_BUGREPORT
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#undef PACKAGE
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#include <soundtouch/SoundTouch.h>
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include "gstpitch.hh"
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#include <math.h>
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GST_DEBUG_CATEGORY_STATIC (pitch_debug);
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#define GST_CAT_DEFAULT pitch_debug
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#define GST_PITCH_GET_PRIVATE(o) (o->priv)
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struct _GstPitchPrivate
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{
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gfloat stream_time_ratio;
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GstEvent *pending_segment;
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soundtouch::SoundTouch * st;
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};
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enum
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{
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ARG_0,
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ARG_OUT_RATE,
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ARG_RATE,
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ARG_TEMPO,
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ARG_PITCH
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};
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/* For soundtouch 1.4 */
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#if defined(INTEGER_SAMPLES)
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#define SOUNDTOUCH_INTEGER_SAMPLES 1
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#elif defined(FLOAT_SAMPLES)
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#define SOUNDTOUCH_FLOAT_SAMPLES 1
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#endif
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#if defined(SOUNDTOUCH_FLOAT_SAMPLES)
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#define SUPPORTED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (F32) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"layout = (string) interleaved"
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#elif defined(SOUNDTOUCH_INTEGER_SAMPLES)
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#define SUPPORTED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (S16) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, MAX ]", \
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"layout = (string) interleaved"
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#else
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#error "Only integer or float samples are supported"
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#endif
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static GstStaticPadTemplate gst_pitch_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (SUPPORTED_CAPS));
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static GstStaticPadTemplate gst_pitch_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (SUPPORTED_CAPS));
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static void gst_pitch_dispose (GObject * object);
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static void gst_pitch_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_pitch_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_pitch_setcaps (GstPitch * pitch, GstCaps * caps);
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static GstFlowReturn gst_pitch_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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static GstStateChangeReturn gst_pitch_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_pitch_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_pitch_src_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_pitch_src_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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#define gst_pitch_parent_class parent_class
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G_DEFINE_TYPE_WITH_PRIVATE (GstPitch, gst_pitch, GST_TYPE_ELEMENT);
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GST_ELEMENT_REGISTER_DEFINE (pitch, "pitch", GST_RANK_NONE,
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GST_TYPE_PITCH);
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static void
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gst_pitch_class_init (GstPitchClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *element_class;
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gobject_class = G_OBJECT_CLASS (klass);
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element_class = GST_ELEMENT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (pitch_debug, "pitch", 0,
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"audio pitch control element");
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gobject_class->set_property = gst_pitch_set_property;
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gobject_class->get_property = gst_pitch_get_property;
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_pitch_dispose);
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g_object_class_install_property (gobject_class, ARG_PITCH,
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g_param_spec_float ("pitch", "Pitch",
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"Audio stream pitch", 0.1, 10.0, 1.0,
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(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
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G_PARAM_STATIC_STRINGS)));
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g_object_class_install_property (gobject_class, ARG_TEMPO,
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g_param_spec_float ("tempo", "Tempo",
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"Audio stream tempo", 0.1, 10.0, 1.0,
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(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
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G_PARAM_STATIC_STRINGS)));
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g_object_class_install_property (gobject_class, ARG_RATE,
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g_param_spec_float ("rate", "Rate",
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"Audio stream rate", 0.1, 10.0, 1.0,
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(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
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G_PARAM_STATIC_STRINGS)));
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g_object_class_install_property (gobject_class, ARG_OUT_RATE,
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g_param_spec_float ("output-rate", "Output Rate",
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"Output rate on downstream segment events", 0.1, 10.0, 1.0,
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(GParamFlags) (G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
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G_PARAM_STATIC_STRINGS)));
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_pitch_change_state);
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gst_element_class_add_static_pad_template (element_class, &gst_pitch_src_template);
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gst_element_class_add_static_pad_template (element_class, &gst_pitch_sink_template);
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gst_element_class_set_static_metadata (element_class, "Pitch controller",
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"Filter/Effect/Audio", "Control the pitch of an audio stream",
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"Wouter Paesen <wouter@blue-gate.be>");
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}
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static void
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gst_pitch_init (GstPitch * pitch)
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{
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pitch->priv = (GstPitchPrivate *) gst_pitch_get_instance_private (pitch);
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pitch->sinkpad =
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gst_pad_new_from_static_template (&gst_pitch_sink_template, "sink");
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gst_pad_set_chain_function (pitch->sinkpad,
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GST_DEBUG_FUNCPTR (gst_pitch_chain));
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gst_pad_set_event_function (pitch->sinkpad,
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GST_DEBUG_FUNCPTR (gst_pitch_sink_event));
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GST_PAD_SET_PROXY_CAPS (pitch->sinkpad);
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gst_element_add_pad (GST_ELEMENT (pitch), pitch->sinkpad);
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pitch->srcpad =
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gst_pad_new_from_static_template (&gst_pitch_src_template, "src");
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gst_pad_set_event_function (pitch->srcpad,
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GST_DEBUG_FUNCPTR (gst_pitch_src_event));
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gst_pad_set_query_function (pitch->srcpad,
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GST_DEBUG_FUNCPTR (gst_pitch_src_query));
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GST_PAD_SET_PROXY_CAPS (pitch->sinkpad);
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gst_element_add_pad (GST_ELEMENT (pitch), pitch->srcpad);
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pitch->priv->st = new soundtouch::SoundTouch ();
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pitch->tempo = 1.0;
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pitch->rate = 1.0;
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pitch->out_seg_rate = 1.0;
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pitch->seg_arate = 1.0;
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pitch->pitch = 1.0;
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pitch->next_buffer_time = GST_CLOCK_TIME_NONE;
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pitch->next_buffer_offset = 0;
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pitch->priv->st->setRate (pitch->rate);
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pitch->priv->st->setTempo (pitch->tempo * pitch->seg_arate);
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pitch->priv->st->setPitch (pitch->pitch);
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pitch->priv->stream_time_ratio = 1.0;
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pitch->min_latency = pitch->max_latency = 0;
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}
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static void
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gst_pitch_dispose (GObject * object)
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{
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GstPitch *pitch = GST_PITCH (object);
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if (pitch->priv->st) {
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delete pitch->priv->st;
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pitch->priv->st = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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gst_pitch_update_duration (GstPitch * pitch)
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{
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GstMessage *m;
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m = gst_message_new_duration_changed (GST_OBJECT (pitch));
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gst_element_post_message (GST_ELEMENT (pitch), m);
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}
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static void
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gst_pitch_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstPitch *pitch = GST_PITCH (object);
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GST_OBJECT_LOCK (pitch);
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switch (prop_id) {
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case ARG_TEMPO:
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pitch->tempo = g_value_get_float (value);
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pitch->priv->stream_time_ratio =
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pitch->tempo * pitch->rate * pitch->seg_arate;
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pitch->priv->st->setTempo (pitch->tempo * pitch->seg_arate);
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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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case ARG_RATE:
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pitch->rate = g_value_get_float (value);
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pitch->priv->stream_time_ratio =
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pitch->tempo * pitch->rate * pitch->seg_arate;
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pitch->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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break;
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case ARG_OUT_RATE:
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/* Has no effect until the next input segment */
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pitch->out_seg_rate = g_value_get_float (value);
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GST_OBJECT_UNLOCK (pitch);
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break;
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case ARG_PITCH:
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pitch->pitch = g_value_get_float (value);
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pitch->priv->st->setPitch (pitch->pitch);
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GST_OBJECT_UNLOCK (pitch);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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GST_OBJECT_UNLOCK (pitch);
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break;
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}
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}
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static void
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gst_pitch_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstPitch *pitch = GST_PITCH (object);
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GST_OBJECT_LOCK (pitch);
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switch (prop_id) {
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case ARG_TEMPO:
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g_value_set_float (value, pitch->tempo);
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break;
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case ARG_RATE:
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g_value_set_float (value, pitch->rate);
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break;
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case ARG_OUT_RATE:
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g_value_set_float (value, pitch->out_seg_rate);
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break;
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case ARG_PITCH:
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g_value_set_float (value, pitch->pitch);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (pitch);
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}
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static gboolean
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gst_pitch_setcaps (GstPitch * pitch, GstCaps * caps)
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{
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GstPitchPrivate *priv;
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priv = GST_PITCH_GET_PRIVATE (pitch);
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if (!gst_audio_info_from_caps (&pitch->info, caps))
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return FALSE;
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GST_OBJECT_LOCK (pitch);
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/* notify the soundtouch instance of this change */
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priv->st->setSampleRate (pitch->info.rate);
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priv->st->setChannels (pitch->info.channels);
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GST_OBJECT_UNLOCK (pitch);
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return TRUE;
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}
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/* send a buffer out */
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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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GST_BUFFER_TIMESTAMP (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_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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return gst_pad_push (pitch->srcpad, buffer);
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}
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/* extract a buffer from soundtouch */
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static GstBuffer *
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gst_pitch_prepare_buffer (GstPitch * pitch)
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{
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GstPitchPrivate *priv;
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guint samples;
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GstBuffer *buffer;
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GstMapInfo info;
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priv = GST_PITCH_GET_PRIVATE (pitch);
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GST_LOG_OBJECT (pitch, "preparing buffer");
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samples = pitch->priv->st->numSamples ();
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if (samples == 0)
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return NULL;
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buffer = gst_buffer_new_and_alloc (samples * pitch->info.bpf);
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gst_buffer_map (buffer, &info, (GstMapFlags) GST_MAP_READWRITE);
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samples = priv->st->receiveSamples ((soundtouch::SAMPLETYPE *) info.data, samples);
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gst_buffer_unmap (buffer, &info);
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if (samples <= 0) {
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gst_buffer_unref (buffer);
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return NULL;
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}
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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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return buffer;
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}
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/* process the last samples, in a later stage we should make sure no more
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* samples are sent out here as strictly necessary, because soundtouch could
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* append zero samples, which could disturb looping. */
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static GstFlowReturn
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gst_pitch_flush_buffer (GstPitch * pitch, gboolean send)
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{
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GstBuffer *buffer;
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if (pitch->priv->st->numUnprocessedSamples() != 0) {
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GST_DEBUG_OBJECT (pitch, "flushing buffer");
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pitch->priv->st->flush ();
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}
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if (!send)
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return GST_FLOW_OK;
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buffer = gst_pitch_prepare_buffer (pitch);
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if (!buffer)
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return GST_FLOW_OK;
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return gst_pitch_forward_buffer (pitch, buffer);
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}
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static gboolean
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gst_pitch_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
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{
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GstPitch *pitch;
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gboolean res;
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pitch = GST_PITCH (parent);
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GST_DEBUG_OBJECT (pad, "received %s event", GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_SEEK:{
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/* transform the event upstream, according to the playback rate */
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gdouble rate;
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GstFormat format;
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GstSeekFlags flags;
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GstSeekType cur_type, stop_type;
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gint64 cur, stop;
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gfloat stream_time_ratio;
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guint32 seqnum;
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GST_OBJECT_LOCK (pitch);
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stream_time_ratio = pitch->priv->stream_time_ratio;
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GST_OBJECT_UNLOCK (pitch);
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gst_event_parse_seek (event, &rate, &format, &flags,
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&cur_type, &cur, &stop_type, &stop);
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seqnum = gst_event_get_seqnum (event);
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gst_event_unref (event);
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if (format == GST_FORMAT_TIME || format == GST_FORMAT_DEFAULT) {
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cur = (gint64) (cur * stream_time_ratio);
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if (stop != -1)
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stop = (gint64) (stop * stream_time_ratio);
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event = gst_event_new_seek (rate, format, flags,
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cur_type, cur, stop_type, stop);
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gst_event_set_seqnum (event, seqnum);
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res = gst_pad_event_default (pad, parent, event);
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} else {
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GST_WARNING_OBJECT (pitch,
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"Seeking only supported in TIME or DEFAULT format");
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res = FALSE;
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}
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break;
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}
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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;
|
|
}
|