mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1009 lines
31 KiB
C
1009 lines
31 KiB
C
/* GStreamer Wavpack encoder plugin
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* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* gstwavpackdec.c: Wavpack audio encoder
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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 Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 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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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-wavpackenc
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* @title: wavpackenc
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*
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* WavpackEnc encodes raw audio into a framed Wavpack stream.
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* [Wavpack](http://www.wavpack.com/) is an open-source audio codec that
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* features both lossless and lossy encoding.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
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* ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
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* as the Wavpack encoder only accepts input with 32 bit width.
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* |[
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* gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
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* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
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* lossless encoding (the file output will be fairly large).
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* |[
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* gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
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* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
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* lossy encoding at a certain bitrate (the file will be fairly small).
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*
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*/
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/*
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* TODO: - add 32 bit float mode. CONFIG_FLOAT_DATA
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*/
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#include <string.h>
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#include <gst/gst.h>
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#include <glib/gprintf.h>
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#include <wavpack/wavpack.h>
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#include "gstwavpackenc.h"
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#include "gstwavpackcommon.h"
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static gboolean gst_wavpack_enc_start (GstAudioEncoder * enc);
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static gboolean gst_wavpack_enc_stop (GstAudioEncoder * enc);
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static gboolean gst_wavpack_enc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_wavpack_enc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static gboolean gst_wavpack_enc_sink_event (GstAudioEncoder * enc,
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GstEvent * event);
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static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
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static GstFlowReturn gst_wavpack_enc_drain (GstWavpackEnc * enc);
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static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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enum
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{
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ARG_0,
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ARG_MODE,
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ARG_BITRATE,
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ARG_BITSPERSAMPLE,
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ARG_CORRECTION_MODE,
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ARG_MD5,
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ARG_EXTRA_PROCESSING,
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ARG_JOINT_STEREO_MODE
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};
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GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
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#define GST_CAT_DEFAULT gst_wavpack_enc_debug
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static GstStaticPadTemplate sink_factory = 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 ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S32) ", "
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"layout = (string) interleaved, "
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"channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
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);
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static GstStaticPadTemplate src_factory = 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 ("audio/x-wavpack, "
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"depth = (int) [ 1, 32 ], "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
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);
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static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) TRUE")
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);
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enum
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{
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GST_WAVPACK_ENC_MODE_VERY_FAST = 0,
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GST_WAVPACK_ENC_MODE_FAST,
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GST_WAVPACK_ENC_MODE_DEFAULT,
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GST_WAVPACK_ENC_MODE_HIGH,
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GST_WAVPACK_ENC_MODE_VERY_HIGH
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};
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#define GST_TYPE_WAVPACK_ENC_MODE (gst_wavpack_enc_mode_get_type ())
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static GType
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gst_wavpack_enc_mode_get_type (void)
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{
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static GType qtype = 0;
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if (qtype == 0) {
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static const GEnumValue values[] = {
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#if 0
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/* Very Fast Compression is not supported yet, but will be supported
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* in future wavpack versions */
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{GST_WAVPACK_ENC_MODE_VERY_FAST, "Very Fast Compression", "veryfast"},
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#endif
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{GST_WAVPACK_ENC_MODE_FAST, "Fast Compression", "fast"},
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{GST_WAVPACK_ENC_MODE_DEFAULT, "Normal Compression", "normal"},
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{GST_WAVPACK_ENC_MODE_HIGH, "High Compression", "high"},
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{GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"},
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{0, NULL, NULL}
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};
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qtype = g_enum_register_static ("GstWavpackEncMode", values);
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}
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return qtype;
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}
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enum
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{
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GST_WAVPACK_CORRECTION_MODE_OFF = 0,
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GST_WAVPACK_CORRECTION_MODE_ON,
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GST_WAVPACK_CORRECTION_MODE_OPTIMIZED
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};
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#define GST_TYPE_WAVPACK_ENC_CORRECTION_MODE (gst_wavpack_enc_correction_mode_get_type ())
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static GType
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gst_wavpack_enc_correction_mode_get_type (void)
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{
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static GType qtype = 0;
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if (qtype == 0) {
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static const GEnumValue values[] = {
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{GST_WAVPACK_CORRECTION_MODE_OFF, "Create no correction file", "off"},
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{GST_WAVPACK_CORRECTION_MODE_ON, "Create correction file", "on"},
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{GST_WAVPACK_CORRECTION_MODE_OPTIMIZED,
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"Create optimized correction file", "optimized"},
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{0, NULL, NULL}
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};
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qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values);
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}
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return qtype;
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}
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enum
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{
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GST_WAVPACK_JS_MODE_AUTO = 0,
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GST_WAVPACK_JS_MODE_LEFT_RIGHT,
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GST_WAVPACK_JS_MODE_MID_SIDE
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};
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#define GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE (gst_wavpack_enc_joint_stereo_mode_get_type ())
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static GType
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gst_wavpack_enc_joint_stereo_mode_get_type (void)
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{
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static GType qtype = 0;
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if (qtype == 0) {
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static const GEnumValue values[] = {
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{GST_WAVPACK_JS_MODE_AUTO, "auto", "auto"},
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{GST_WAVPACK_JS_MODE_LEFT_RIGHT, "left/right", "leftright"},
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{GST_WAVPACK_JS_MODE_MID_SIDE, "mid/side", "midside"},
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{0, NULL, NULL}
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};
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qtype = g_enum_register_static ("GstWavpackEncJSMode", values);
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}
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return qtype;
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}
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#define gst_wavpack_enc_parent_class parent_class
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G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) (klass);
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GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
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/* add pad templates */
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gst_element_class_add_static_pad_template (element_class, &sink_factory);
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gst_element_class_add_static_pad_template (element_class, &src_factory);
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gst_element_class_add_static_pad_template (element_class, &wvcsrc_factory);
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/* set element details */
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gst_element_class_set_static_metadata (element_class, "Wavpack audio encoder",
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"Codec/Encoder/Audio",
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"Encodes audio with the Wavpack lossless/lossy audio codec",
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"Sebastian Dröge <slomo@circular-chaos.org>");
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/* set property handlers */
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gobject_class->set_property = gst_wavpack_enc_set_property;
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gobject_class->get_property = gst_wavpack_enc_get_property;
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base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame);
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base_class->sink_event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event);
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/* install all properties */
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g_object_class_install_property (gobject_class, ARG_MODE,
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g_param_spec_enum ("mode", "Encoding mode",
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"Speed versus compression tradeoff.",
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GST_TYPE_WAVPACK_ENC_MODE, GST_WAVPACK_ENC_MODE_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_BITRATE,
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g_param_spec_uint ("bitrate", "Bitrate",
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"Try to encode with this average bitrate (bits/sec). "
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"This enables lossy encoding, values smaller than 24000 disable it again.",
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0, 9600000, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_BITSPERSAMPLE,
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g_param_spec_double ("bits-per-sample", "Bits per sample",
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"Try to encode with this amount of bits per sample. "
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"This enables lossy encoding, values smaller than 2.0 disable it again.",
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0.0, 24.0, 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
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g_param_spec_enum ("correction-mode", "Correction stream mode",
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"Use this mode for the correction stream. Only works in lossy mode!",
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GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, GST_WAVPACK_CORRECTION_MODE_OFF,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_MD5,
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g_param_spec_boolean ("md5", "MD5",
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"Store MD5 hash of raw samples within the file.", FALSE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
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g_param_spec_uint ("extra-processing", "Extra processing",
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"Use better but slower filters for better compression/quality.",
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0, 6, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_JOINT_STEREO_MODE,
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g_param_spec_enum ("joint-stereo-mode", "Joint-Stereo mode",
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"Use this joint-stereo mode.", GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE,
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GST_WAVPACK_JS_MODE_AUTO,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_type_mark_as_plugin_api (GST_TYPE_WAVPACK_ENC_MODE, 0);
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gst_type_mark_as_plugin_api (GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, 0);
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gst_type_mark_as_plugin_api (GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE, 0);
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}
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static void
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gst_wavpack_enc_reset (GstWavpackEnc * enc)
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{
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/* close and free everything stream related if we already did something */
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if (enc->wp_context) {
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WavpackCloseFile (enc->wp_context);
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enc->wp_context = NULL;
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}
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if (enc->wp_config) {
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g_free (enc->wp_config);
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enc->wp_config = NULL;
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}
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if (enc->first_block) {
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g_free (enc->first_block);
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enc->first_block = NULL;
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}
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enc->first_block_size = 0;
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if (enc->md5_context) {
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g_checksum_free (enc->md5_context);
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enc->md5_context = NULL;
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}
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if (enc->pending_segment)
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gst_event_unref (enc->pending_segment);
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enc->pending_segment = NULL;
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if (enc->pending_buffer) {
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gst_buffer_unref (enc->pending_buffer);
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enc->pending_buffer = NULL;
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enc->pending_offset = 0;
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}
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/* reset the last returns to GST_FLOW_OK. This is only set to something else
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* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
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* so not valid anymore */
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enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
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/* reset stream information */
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enc->samplerate = 0;
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enc->depth = 0;
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enc->channels = 0;
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enc->channel_mask = 0;
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enc->need_channel_remap = FALSE;
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enc->timestamp_offset = GST_CLOCK_TIME_NONE;
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enc->next_ts = GST_CLOCK_TIME_NONE;
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}
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static void
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gst_wavpack_enc_init (GstWavpackEnc * enc)
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{
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GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
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/* initialize object attributes */
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enc->wp_config = NULL;
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enc->wp_context = NULL;
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enc->first_block = NULL;
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enc->md5_context = NULL;
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gst_wavpack_enc_reset (enc);
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enc->wv_id.correction = FALSE;
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enc->wv_id.wavpack_enc = enc;
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enc->wv_id.passthrough = FALSE;
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enc->wvc_id.correction = TRUE;
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enc->wvc_id.wavpack_enc = enc;
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enc->wvc_id.passthrough = FALSE;
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/* set default values of params */
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enc->mode = GST_WAVPACK_ENC_MODE_DEFAULT;
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enc->bitrate = 0;
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enc->bps = 0.0;
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enc->correction_mode = GST_WAVPACK_CORRECTION_MODE_OFF;
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enc->md5 = FALSE;
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enc->extra_processing = 0;
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enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
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/* require perfect ts */
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gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
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}
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static gboolean
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gst_wavpack_enc_start (GstAudioEncoder * enc)
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{
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GST_DEBUG_OBJECT (enc, "start");
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return TRUE;
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}
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static gboolean
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gst_wavpack_enc_stop (GstAudioEncoder * enc)
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{
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GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
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GST_DEBUG_OBJECT (enc, "stop");
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gst_wavpack_enc_reset (wpenc);
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return TRUE;
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}
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static gboolean
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gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
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GstAudioChannelPosition *pos;
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GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
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GstCaps *caps;
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guint64 mask = 0;
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/* we may be configured again, but that change should have cleanup context */
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g_assert (enc->wp_context == NULL);
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enc->channels = GST_AUDIO_INFO_CHANNELS (info);
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enc->depth = GST_AUDIO_INFO_DEPTH (info);
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enc->samplerate = GST_AUDIO_INFO_RATE (info);
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pos = info->position;
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g_assert (pos);
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/* If one channel is NONE they'll be all undefined */
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if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
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goto invalid_channels;
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}
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enc->channel_mask =
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gst_wavpack_get_channel_mask_from_positions (pos, enc->channels);
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enc->need_channel_remap =
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gst_wavpack_set_channel_mapping (pos, enc->channels,
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enc->channel_mapping);
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/* wavpack caps hold gst mask, not wavpack mask */
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gst_audio_channel_positions_to_mask (opos, enc->channels, FALSE, &mask);
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/* set fixed src pad caps now that we know what we will get */
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caps = gst_caps_new_simple ("audio/x-wavpack",
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"channels", G_TYPE_INT, enc->channels,
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"rate", G_TYPE_INT, enc->samplerate,
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"depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
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if (mask)
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gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
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if (!gst_audio_encoder_set_output_format (benc, caps))
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goto setting_src_caps_failed;
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gst_caps_unref (caps);
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/* no special feedback to base class; should provide all available samples */
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return TRUE;
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/* ERRORS */
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setting_src_caps_failed:
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{
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GST_DEBUG_OBJECT (enc,
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"Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps);
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gst_caps_unref (caps);
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return FALSE;
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}
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invalid_channels:
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{
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GST_DEBUG_OBJECT (enc, "input has invalid channel layout");
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return FALSE;
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}
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}
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static void
|
|
gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc)
|
|
{
|
|
enc->wp_config = g_new0 (WavpackConfig, 1);
|
|
/* set general stream information in the WavpackConfig */
|
|
enc->wp_config->bytes_per_sample = GST_ROUND_UP_8 (enc->depth) / 8;
|
|
enc->wp_config->bits_per_sample = enc->depth;
|
|
enc->wp_config->num_channels = enc->channels;
|
|
enc->wp_config->channel_mask = enc->channel_mask;
|
|
enc->wp_config->sample_rate = enc->samplerate;
|
|
|
|
/*
|
|
* Set parameters in WavpackConfig
|
|
*/
|
|
|
|
/* Encoding mode */
|
|
switch (enc->mode) {
|
|
#if 0
|
|
case GST_WAVPACK_ENC_MODE_VERY_FAST:
|
|
enc->wp_config->flags |= CONFIG_VERY_FAST_FLAG;
|
|
enc->wp_config->flags |= CONFIG_FAST_FLAG;
|
|
break;
|
|
#endif
|
|
case GST_WAVPACK_ENC_MODE_FAST:
|
|
enc->wp_config->flags |= CONFIG_FAST_FLAG;
|
|
break;
|
|
case GST_WAVPACK_ENC_MODE_DEFAULT:
|
|
break;
|
|
case GST_WAVPACK_ENC_MODE_HIGH:
|
|
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
|
|
break;
|
|
case GST_WAVPACK_ENC_MODE_VERY_HIGH:
|
|
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
|
|
enc->wp_config->flags |= CONFIG_VERY_HIGH_FLAG;
|
|
break;
|
|
}
|
|
|
|
/* Bitrate, enables lossy mode */
|
|
if (enc->bitrate) {
|
|
enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
|
|
enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
|
|
enc->wp_config->bitrate = enc->bitrate / 1000.0;
|
|
} else if (enc->bps) {
|
|
enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
|
|
enc->wp_config->bitrate = enc->bps;
|
|
}
|
|
|
|
/* Correction Mode, only in lossy mode */
|
|
if (enc->wp_config->flags & CONFIG_HYBRID_FLAG) {
|
|
if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
|
|
GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
|
|
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
|
|
enc->wvcsrcpad =
|
|
gst_pad_new_from_static_template (&wvcsrc_factory, "wvcsrc");
|
|
|
|
/* try to add correction src pad, don't set correction mode on failure */
|
|
GST_DEBUG_OBJECT (enc, "Adding correction pad with caps %"
|
|
GST_PTR_FORMAT, caps);
|
|
if (!gst_pad_set_caps (enc->wvcsrcpad, caps)) {
|
|
enc->correction_mode = 0;
|
|
GST_WARNING_OBJECT (enc, "setting correction caps failed");
|
|
} else {
|
|
gst_pad_use_fixed_caps (enc->wvcsrcpad);
|
|
gst_pad_set_active (enc->wvcsrcpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT (enc), enc->wvcsrcpad);
|
|
enc->wp_config->flags |= CONFIG_CREATE_WVC;
|
|
if (enc->correction_mode == GST_WAVPACK_CORRECTION_MODE_OPTIMIZED) {
|
|
enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC;
|
|
}
|
|
}
|
|
gst_caps_unref (caps);
|
|
}
|
|
} else {
|
|
if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
|
|
enc->correction_mode = 0;
|
|
GST_WARNING_OBJECT (enc, "setting correction mode only has "
|
|
"any effect if a bitrate is provided.");
|
|
}
|
|
}
|
|
gst_element_no_more_pads (GST_ELEMENT (enc));
|
|
|
|
/* MD5, setup MD5 context */
|
|
if ((enc->md5) && !(enc->md5_context)) {
|
|
enc->wp_config->flags |= CONFIG_MD5_CHECKSUM;
|
|
enc->md5_context = g_checksum_new (G_CHECKSUM_MD5);
|
|
}
|
|
|
|
/* Extra encode processing */
|
|
if (enc->extra_processing) {
|
|
enc->wp_config->flags |= CONFIG_EXTRA_MODE;
|
|
enc->wp_config->xmode = enc->extra_processing;
|
|
}
|
|
|
|
/* Joint stereo mode */
|
|
switch (enc->joint_stereo_mode) {
|
|
case GST_WAVPACK_JS_MODE_AUTO:
|
|
break;
|
|
case GST_WAVPACK_JS_MODE_LEFT_RIGHT:
|
|
enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE;
|
|
enc->wp_config->flags &= ~CONFIG_JOINT_STEREO;
|
|
break;
|
|
case GST_WAVPACK_JS_MODE_MID_SIDE:
|
|
enc->wp_config->flags |= (CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static int
|
|
gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
|
|
{
|
|
GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id;
|
|
GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc);
|
|
GstFlowReturn *flow;
|
|
GstBuffer *buffer;
|
|
GstPad *pad;
|
|
guchar *block = (guchar *) data;
|
|
gint samples = 0;
|
|
|
|
pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
|
|
flow =
|
|
(wid->correction) ? &enc->
|
|
wvcsrcpad_last_return : &enc->srcpad_last_return;
|
|
|
|
buffer = gst_buffer_new_and_alloc (count);
|
|
gst_buffer_fill (buffer, 0, data, count);
|
|
|
|
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
|
|
/* if it's a Wavpack block set buffer timestamp and duration, etc */
|
|
WavpackHeader wph;
|
|
|
|
GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata",
|
|
count, (wid->correction) ? "correction " : "");
|
|
|
|
gst_wavpack_read_header (&wph, block);
|
|
|
|
/* Only set when pushing the first buffer again, in that case
|
|
* we don't want to delay the buffer or push newsegment events
|
|
*/
|
|
if (!wid->passthrough) {
|
|
/* Only push complete blocks */
|
|
if (enc->pending_buffer == NULL) {
|
|
enc->pending_buffer = buffer;
|
|
enc->pending_offset = wph.block_index;
|
|
} else if (enc->pending_offset == wph.block_index) {
|
|
enc->pending_buffer = gst_buffer_append (enc->pending_buffer, buffer);
|
|
} else {
|
|
GST_ERROR ("Got incomplete block, dropping");
|
|
gst_buffer_unref (enc->pending_buffer);
|
|
enc->pending_buffer = buffer;
|
|
enc->pending_offset = wph.block_index;
|
|
}
|
|
|
|
/* Is this the not-final block of multi-channel data? If so, just
|
|
* accumulate and return here. */
|
|
if (!(wph.flags & FINAL_BLOCK) && ((block[32] & ID_OPTIONAL_DATA) == 0))
|
|
return TRUE;
|
|
|
|
buffer = enc->pending_buffer;
|
|
enc->pending_buffer = NULL;
|
|
enc->pending_offset = 0;
|
|
|
|
/* only send segment on correction pad,
|
|
* regular pad is handled normally by baseclass */
|
|
if (wid->correction && enc->pending_segment) {
|
|
gst_pad_push_event (pad, enc->pending_segment);
|
|
enc->pending_segment = NULL;
|
|
}
|
|
|
|
if (wph.block_index == 0) {
|
|
/* save header for later reference, so we can re-send it later on
|
|
* EOS with fixed up values for total sample count etc. */
|
|
if (enc->first_block == NULL && !wid->correction) {
|
|
GstMapInfo map;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
enc->first_block = g_memdup (map.data, map.size);
|
|
enc->first_block_size = map.size;
|
|
gst_buffer_unmap (buffer, &map);
|
|
}
|
|
}
|
|
}
|
|
samples = wph.block_samples;
|
|
|
|
/* decorate buffer */
|
|
/* NOTE: this will get overwritten by baseclass, but stay for those
|
|
* that are pushed directly
|
|
* FIXME: add setting to baseclass to avoid overwriting it ?? */
|
|
GST_BUFFER_OFFSET (buffer) = wph.block_index;
|
|
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
|
|
} else {
|
|
/* if it's something else set no timestamp and duration on the buffer */
|
|
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
|
|
}
|
|
|
|
if (wid->correction || wid->passthrough) {
|
|
/* push the buffer and forward errors */
|
|
GST_DEBUG_OBJECT (enc, "pushing buffer with %" G_GSIZE_FORMAT " bytes",
|
|
gst_buffer_get_size (buffer));
|
|
*flow = gst_pad_push (pad, buffer);
|
|
} else {
|
|
GST_DEBUG_OBJECT (enc, "handing frame of %" G_GSIZE_FORMAT " bytes",
|
|
gst_buffer_get_size (buffer));
|
|
*flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
|
|
samples);
|
|
}
|
|
|
|
if (*flow != GST_FLOW_OK) {
|
|
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
|
|
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
|
|
gint nsamples)
|
|
{
|
|
gint i, j;
|
|
gint32 tmp[8];
|
|
|
|
for (i = 0; i < nsamples / enc->channels; i++) {
|
|
for (j = 0; j < enc->channels; j++) {
|
|
tmp[enc->channel_mapping[j]] = data[j];
|
|
}
|
|
for (j = 0; j < enc->channels; j++) {
|
|
data[j] = tmp[j];
|
|
}
|
|
data += enc->channels;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|
{
|
|
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
|
|
uint32_t sample_count;
|
|
GstFlowReturn ret;
|
|
GstMapInfo map;
|
|
|
|
/* base class ensures configuration */
|
|
g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
|
|
|
|
/* reset the last returns to GST_FLOW_OK. This is only set to something else
|
|
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
|
|
* so not valid anymore */
|
|
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
|
|
|
|
if (G_UNLIKELY (!buf))
|
|
return gst_wavpack_enc_drain (enc);
|
|
|
|
sample_count = gst_buffer_get_size (buf) / 4;
|
|
GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
|
|
|
|
/* check if we already have a valid WavpackContext, otherwise make one */
|
|
if (!enc->wp_context) {
|
|
/* create raw context */
|
|
enc->wp_context =
|
|
WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
|
|
(enc->correction_mode > 0) ? &enc->wvc_id : NULL);
|
|
if (!enc->wp_context)
|
|
goto context_failed;
|
|
|
|
/* set the WavpackConfig according to our parameters */
|
|
gst_wavpack_enc_set_wp_config (enc);
|
|
|
|
/* set the configuration to the context now that we know everything
|
|
* and initialize the encoder */
|
|
if (!WavpackSetConfiguration (enc->wp_context,
|
|
enc->wp_config, (uint32_t) (-1))
|
|
|| !WavpackPackInit (enc->wp_context)) {
|
|
WavpackCloseFile (enc->wp_context);
|
|
goto config_failed;
|
|
}
|
|
GST_DEBUG_OBJECT (enc, "setup of encoding context successful");
|
|
}
|
|
|
|
if (enc->need_channel_remap) {
|
|
buf = gst_buffer_make_writable (buf);
|
|
gst_buffer_map (buf, &map, GST_MAP_WRITE);
|
|
gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count);
|
|
gst_buffer_unmap (buf, &map);
|
|
}
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
|
|
/* if we want to append the MD5 sum to the stream update it here
|
|
* with the current raw samples */
|
|
if (enc->md5) {
|
|
g_checksum_update (enc->md5_context, map.data, map.size);
|
|
}
|
|
|
|
/* encode and handle return values from encoding */
|
|
if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data,
|
|
sample_count / enc->channels)) {
|
|
GST_DEBUG_OBJECT (enc, "encoding samples successful");
|
|
gst_buffer_unmap (buf, &map);
|
|
ret = GST_FLOW_OK;
|
|
} else {
|
|
gst_buffer_unmap (buf, &map);
|
|
if ((enc->srcpad_last_return == GST_FLOW_OK) ||
|
|
(enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
|
|
ret = GST_FLOW_OK;
|
|
} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
|
|
(enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
|
|
ret = GST_FLOW_NOT_LINKED;
|
|
} else if ((enc->srcpad_last_return == GST_FLOW_FLUSHING) &&
|
|
(enc->wvcsrcpad_last_return == GST_FLOW_FLUSHING)) {
|
|
ret = GST_FLOW_FLUSHING;
|
|
} else {
|
|
goto encoding_failed;
|
|
}
|
|
}
|
|
|
|
exit:
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
encoding_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
|
|
("encoding samples failed"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto exit;
|
|
}
|
|
config_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
|
|
("error setting up wavpack encoding context"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto exit;
|
|
}
|
|
context_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
|
|
("error creating Wavpack context"));
|
|
ret = GST_FLOW_ERROR;
|
|
goto exit;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
|
|
{
|
|
GstSegment segment;
|
|
gboolean ret;
|
|
GstQuery *query;
|
|
gboolean seekable = FALSE;
|
|
|
|
g_return_if_fail (enc);
|
|
g_return_if_fail (enc->first_block);
|
|
|
|
/* update the sample count in the first block */
|
|
WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
|
|
|
|
/* try to seek to the beginning of the output */
|
|
query = gst_query_new_seeking (GST_FORMAT_BYTES);
|
|
if (gst_pad_peer_query (GST_AUDIO_ENCODER_SRC_PAD (enc), query)) {
|
|
GstFormat format;
|
|
|
|
gst_query_parse_seeking (query, &format, &seekable, NULL, NULL);
|
|
if (format != GST_FORMAT_BYTES)
|
|
seekable = FALSE;
|
|
} else {
|
|
GST_LOG_OBJECT (enc, "SEEKING query not handled");
|
|
}
|
|
gst_query_unref (query);
|
|
|
|
if (!seekable) {
|
|
GST_DEBUG_OBJECT (enc, "downstream not seekable; not rewriting");
|
|
return;
|
|
}
|
|
|
|
gst_segment_init (&segment, GST_FORMAT_BYTES);
|
|
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
|
|
gst_event_new_segment (&segment));
|
|
if (ret) {
|
|
/* try to rewrite the first block */
|
|
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
|
|
enc->wv_id.passthrough = TRUE;
|
|
ret = gst_wavpack_enc_push_block (&enc->wv_id,
|
|
enc->first_block, enc->first_block_size);
|
|
enc->wv_id.passthrough = FALSE;
|
|
g_free (enc->first_block);
|
|
enc->first_block = NULL;
|
|
} else {
|
|
GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
|
|
"Seeking to first block failed!");
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_wavpack_enc_drain (GstWavpackEnc * enc)
|
|
{
|
|
if (!enc->wp_context)
|
|
return GST_FLOW_OK;
|
|
|
|
GST_DEBUG_OBJECT (enc, "draining");
|
|
|
|
/* Encode all remaining samples and flush them to the src pads */
|
|
WavpackFlushSamples (enc->wp_context);
|
|
|
|
/* Drop all remaining data, this is no complete block otherwise
|
|
* it would've been pushed already */
|
|
if (enc->pending_buffer) {
|
|
gst_buffer_unref (enc->pending_buffer);
|
|
enc->pending_buffer = NULL;
|
|
enc->pending_offset = 0;
|
|
}
|
|
|
|
/* write the MD5 sum if we have to write one */
|
|
if ((enc->md5) && (enc->md5_context)) {
|
|
guint8 md5_digest[16];
|
|
gsize digest_len = sizeof (md5_digest);
|
|
|
|
g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
|
|
if (digest_len == sizeof (md5_digest)) {
|
|
WavpackStoreMD5Sum (enc->wp_context, md5_digest);
|
|
WavpackFlushSamples (enc->wp_context);
|
|
} else
|
|
GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
|
|
}
|
|
|
|
/* Try to rewrite the first frame with the correct sample number */
|
|
if (enc->first_block)
|
|
gst_wavpack_enc_rewrite_first_block (enc);
|
|
|
|
/* close the context if not already happened */
|
|
if (enc->wp_context) {
|
|
WavpackCloseFile (enc->wp_context);
|
|
enc->wp_context = NULL;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
|
|
{
|
|
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
|
|
|
|
GST_DEBUG_OBJECT (enc, "Received %s event on sinkpad",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEGMENT:
|
|
if (enc->wp_context) {
|
|
GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
|
|
"already started");
|
|
}
|
|
/* peek and hold NEWSEGMENT events for sending on correction pad */
|
|
if (enc->pending_segment)
|
|
gst_event_unref (enc->pending_segment);
|
|
enc->pending_segment = gst_event_ref (event);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* baseclass handles rest */
|
|
return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
|
|
}
|
|
|
|
static void
|
|
gst_wavpack_enc_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_MODE:
|
|
enc->mode = g_value_get_enum (value);
|
|
break;
|
|
case ARG_BITRATE:{
|
|
guint val = g_value_get_uint (value);
|
|
|
|
if ((val >= 24000) && (val <= 9600000)) {
|
|
enc->bitrate = val;
|
|
enc->bps = 0.0;
|
|
} else {
|
|
enc->bitrate = 0;
|
|
enc->bps = 0.0;
|
|
}
|
|
break;
|
|
}
|
|
case ARG_BITSPERSAMPLE:{
|
|
gdouble val = g_value_get_double (value);
|
|
|
|
if ((val >= 2.0) && (val <= 24.0)) {
|
|
enc->bps = val;
|
|
enc->bitrate = 0;
|
|
} else {
|
|
enc->bps = 0.0;
|
|
enc->bitrate = 0;
|
|
}
|
|
break;
|
|
}
|
|
case ARG_CORRECTION_MODE:
|
|
enc->correction_mode = g_value_get_enum (value);
|
|
break;
|
|
case ARG_MD5:
|
|
enc->md5 = g_value_get_boolean (value);
|
|
break;
|
|
case ARG_EXTRA_PROCESSING:
|
|
enc->extra_processing = g_value_get_uint (value);
|
|
break;
|
|
case ARG_JOINT_STEREO_MODE:
|
|
enc->joint_stereo_mode = g_value_get_enum (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_wavpack_enc_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_MODE:
|
|
g_value_set_enum (value, enc->mode);
|
|
break;
|
|
case ARG_BITRATE:
|
|
if (enc->bps == 0.0) {
|
|
g_value_set_uint (value, enc->bitrate);
|
|
} else {
|
|
g_value_set_uint (value, 0);
|
|
}
|
|
break;
|
|
case ARG_BITSPERSAMPLE:
|
|
if (enc->bitrate == 0) {
|
|
g_value_set_double (value, enc->bps);
|
|
} else {
|
|
g_value_set_double (value, 0.0);
|
|
}
|
|
break;
|
|
case ARG_CORRECTION_MODE:
|
|
g_value_set_enum (value, enc->correction_mode);
|
|
break;
|
|
case ARG_MD5:
|
|
g_value_set_boolean (value, enc->md5);
|
|
break;
|
|
case ARG_EXTRA_PROCESSING:
|
|
g_value_set_uint (value, enc->extra_processing);
|
|
break;
|
|
case ARG_JOINT_STEREO_MODE:
|
|
g_value_set_enum (value, enc->joint_stereo_mode);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_wavpack_enc_plugin_init (GstPlugin * plugin)
|
|
{
|
|
if (!gst_element_register (plugin, "wavpackenc",
|
|
GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
|
|
return FALSE;
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0,
|
|
"Wavpack encoder");
|
|
|
|
return TRUE;
|
|
}
|