/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:gstaudio * @title: GstAudio * @short_description: Support library for audio elements * * This library contains some helper functions for audio elements. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include "audio.h" #include "audio-enumtypes.h" #ifndef GST_DISABLE_GST_DEBUG #define GST_CAT_DEFAULT ensure_debug_category() static GstDebugCategory * ensure_debug_category (void) { static gsize cat_gonce = 0; if (g_once_init_enter (&cat_gonce)) { gsize cat_done; cat_done = (gsize) _gst_debug_category_new ("audio", 0, "audio library"); g_once_init_leave (&cat_gonce, cat_done); } return (GstDebugCategory *) cat_gonce; } #else #define ensure_debug_category() /* NOOP */ #endif /* GST_DISABLE_GST_DEBUG */ /** * gst_audio_buffer_clip: * @buffer: (transfer full): The buffer to clip. * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which * the buffer should be clipped. * @rate: sample rate. * @bpf: size of one audio frame in bytes. This is the size of one sample * * number of channels. * * Clip the buffer to the given %GstSegment. * * After calling this function the caller does not own a reference to * @buffer anymore. * * Returns: (transfer full): %NULL if the buffer is completely outside the configured segment, * otherwise the clipped buffer is returned. * * If the buffer has no timestamp, it is assumed to be inside the segment and * is not clipped */ GstBuffer * gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment, gint rate, gint bpf) { GstBuffer *ret; GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE; guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE; gsize trim, size, osize; gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end = TRUE; g_return_val_if_fail (segment->format == GST_FORMAT_TIME || segment->format == GST_FORMAT_DEFAULT, buffer); g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL); if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) /* No timestamp - assume the buffer is completely in the segment */ return buffer; /* Get copies of the buffer metadata to change later. * Calculate the missing values for the calculations, * they won't be changed later though. */ trim = 0; osize = size = gst_buffer_get_size (buffer); /* no data, nothing to clip */ if (!size) return buffer; timestamp = GST_BUFFER_TIMESTAMP (buffer); GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp)); if (GST_BUFFER_DURATION_IS_VALID (buffer)) { duration = GST_BUFFER_DURATION (buffer); } else { change_duration = FALSE; duration = gst_util_uint64_scale (size / bpf, GST_SECOND, rate); } if (GST_BUFFER_OFFSET_IS_VALID (buffer)) { offset = GST_BUFFER_OFFSET (buffer); } else { change_offset = FALSE; offset = 0; } if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) { offset_end = GST_BUFFER_OFFSET_END (buffer); } else { change_offset_end = FALSE; offset_end = offset + size / bpf; } if (segment->format == GST_FORMAT_TIME) { /* Handle clipping for GST_FORMAT_TIME */ guint64 start, stop, cstart, cstop, diff; start = timestamp; stop = timestamp + duration; if (gst_segment_clip (segment, GST_FORMAT_TIME, start, stop, &cstart, &cstop)) { diff = cstart - start; if (diff > 0) { timestamp = cstart; if (change_duration) duration -= diff; diff = gst_util_uint64_scale (diff, rate, GST_SECOND); if (change_offset) offset += diff; trim += diff * bpf; size -= diff * bpf; } diff = stop - cstop; if (diff > 0) { /* duration is always valid if stop is valid */ duration -= diff; diff = gst_util_uint64_scale (diff, rate, GST_SECOND); if (change_offset_end) offset_end -= diff; size -= diff * bpf; } } else { gst_buffer_unref (buffer); return NULL; } } else { /* Handle clipping for GST_FORMAT_DEFAULT */ guint64 start, stop, cstart, cstop, diff; g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer); start = offset; stop = offset_end; if (gst_segment_clip (segment, GST_FORMAT_DEFAULT, start, stop, &cstart, &cstop)) { diff = cstart - start; if (diff > 0) { offset = cstart; timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate); if (change_duration) duration -= gst_util_uint64_scale (diff, GST_SECOND, rate); trim += diff * bpf; size -= diff * bpf; } diff = stop - cstop; if (diff > 0) { offset_end = cstop; if (change_duration) duration -= gst_util_uint64_scale (diff, GST_SECOND, rate); size -= diff * bpf; } } else { gst_buffer_unref (buffer); return NULL; } } if (trim == 0 && size == osize) { ret = buffer; if (GST_BUFFER_TIMESTAMP (ret) != timestamp) { ret = gst_buffer_make_writable (ret); GST_BUFFER_TIMESTAMP (ret) = timestamp; } if (GST_BUFFER_DURATION (ret) != duration) { ret = gst_buffer_make_writable (ret); GST_BUFFER_DURATION (ret) = duration; } } else { /* Get a writable buffer and apply all changes */ GST_DEBUG ("trim %" G_GSIZE_FORMAT " size %" G_GSIZE_FORMAT, trim, size); ret = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, trim, size); gst_buffer_unref (buffer); GST_DEBUG ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp)); if (ret) { GST_BUFFER_TIMESTAMP (ret) = timestamp; if (change_duration) GST_BUFFER_DURATION (ret) = duration; if (change_offset) GST_BUFFER_OFFSET (ret) = offset; if (change_offset_end) GST_BUFFER_OFFSET_END (ret) = offset_end; } else { GST_ERROR ("copy_region failed"); } } return ret; }