/* GStreamer * Copyright (C) 1999 Erik Walthinsen * Copyright (C) 2003,2004 David A. Schleef * Copyright (C) 2007-2008 Sebastian Dröge * * 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:element-audioresample * @title: audioresample * * audioresample resamples raw audio buffers to different sample rates using * a configurable windowing function to enhance quality. * * By default, the resampler uses a reduced sinc table, with cubic interpolation filling in * the gaps. This ensures that the table does not become too big. However, the interpolation * increases the CPU usage considerably. As an alternative, a full sinc table can be used. * Doing so can drastically reduce CPU usage (4x faster with 44.1 -> 48 kHz conversions for * example), at the cost of increased memory consumption, plus the sinc table takes longer * to initialize when the element is created. A third mode exists, which uses the full table * unless said table would become too large, in which case the interpolated one is used instead. * * ## Example launch line * |[ * gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! audio/x-raw, rate=8000 ! autoaudiosink * ]| * Decode an audio file and downsample it to 8Khz and play sound. * To create the Ogg/Vorbis file refer to the documentation of vorbisenc. * This assumes there is an audio sink that will accept/handle 8kHz audio. * */ /* TODO: * - Enable SSE/ARM optimizations and select at runtime */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "gstaudioresample.h" #include #include #include GST_DEBUG_CATEGORY (audio_resample_debug); #define GST_CAT_DEFAULT audio_resample_debug #undef USE_SPEEX #define DEFAULT_QUALITY GST_AUDIO_RESAMPLER_QUALITY_DEFAULT #define DEFAULT_RESAMPLE_METHOD GST_AUDIO_RESAMPLER_METHOD_KAISER #define DEFAULT_SINC_FILTER_MODE GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO #define DEFAULT_SINC_FILTER_AUTO_THRESHOLD (1*1048576) #define DEFAULT_SINC_FILTER_INTERPOLATION GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC enum { PROP_0, PROP_QUALITY, PROP_RESAMPLE_METHOD, PROP_SINC_FILTER_MODE, PROP_SINC_FILTER_AUTO_THRESHOLD, PROP_SINC_FILTER_INTERPOLATION }; #define SUPPORTED_CAPS \ GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \ ", layout = (string) { interleaved, non-interleaved }" static GstStaticPadTemplate gst_audio_resample_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (SUPPORTED_CAPS)); static GstStaticPadTemplate gst_audio_resample_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (SUPPORTED_CAPS)); /* cached quark to avoid contention on the global quark table lock */ #define META_TAG_AUDIO meta_tag_audio_quark static GQuark meta_tag_audio_quark; static void gst_audio_resample_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_resample_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); /* vmethods */ static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps, gsize * size); static GstCaps *gst_audio_resample_transform_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * filter); static GstCaps *gst_audio_resample_fixate_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * othercaps); static gboolean gst_audio_resample_transform_size (GstBaseTransform * trans, GstPadDirection direction, GstCaps * incaps, gsize insize, GstCaps * outcaps, gsize * outsize); static gboolean gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps); static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf); static gboolean gst_audio_resample_transform_meta (GstBaseTransform * trans, GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf); static GstFlowReturn gst_audio_resample_submit_input_buffer (GstBaseTransform * base, gboolean is_discont, GstBuffer * input); static gboolean gst_audio_resample_sink_event (GstBaseTransform * base, GstEvent * event); static gboolean gst_audio_resample_start (GstBaseTransform * base); static gboolean gst_audio_resample_stop (GstBaseTransform * base); static gboolean gst_audio_resample_query (GstPad * pad, GstObject * parent, GstQuery * query); static void gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len); #define gst_audio_resample_parent_class parent_class G_DEFINE_TYPE (GstAudioResample, gst_audio_resample, GST_TYPE_BASE_TRANSFORM); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (audioresample, "audioresample", GST_RANK_PRIMARY, GST_TYPE_AUDIO_RESAMPLE, GST_DEBUG_CATEGORY_INIT (audio_resample_debug, "audioresample", 0, "audio resampling element")); static void gst_audio_resample_class_init (GstAudioResampleClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; gobject_class->set_property = gst_audio_resample_set_property; gobject_class->get_property = gst_audio_resample_get_property; g_object_class_install_property (gobject_class, PROP_QUALITY, g_param_spec_int ("quality", "Quality", "Resample quality with 0 being " "the lowest and 10 being the best", GST_AUDIO_RESAMPLER_QUALITY_MIN, GST_AUDIO_RESAMPLER_QUALITY_MAX, DEFAULT_QUALITY, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RESAMPLE_METHOD, g_param_spec_enum ("resample-method", "Resample method to use", "What resample method to use", GST_TYPE_AUDIO_RESAMPLER_METHOD, DEFAULT_RESAMPLE_METHOD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SINC_FILTER_MODE, g_param_spec_enum ("sinc-filter-mode", "Sinc filter table mode", "What sinc filter table mode to use", GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE, DEFAULT_SINC_FILTER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SINC_FILTER_AUTO_THRESHOLD, g_param_spec_uint ("sinc-filter-auto-threshold", "Sinc filter auto mode threshold", "Memory usage threshold to use if sinc filter mode is AUTO, given in bytes", 0, G_MAXUINT, DEFAULT_SINC_FILTER_AUTO_THRESHOLD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SINC_FILTER_INTERPOLATION, g_param_spec_enum ("sinc-filter-interpolation", "Sinc filter interpolation", "How to interpolate the sinc filter table", GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION, DEFAULT_SINC_FILTER_INTERPOLATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &gst_audio_resample_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_audio_resample_sink_template); gst_element_class_set_static_metadata (gstelement_class, "Audio resampler", "Filter/Converter/Audio", "Resamples audio", "Sebastian Dröge "); GST_BASE_TRANSFORM_CLASS (klass)->start = GST_DEBUG_FUNCPTR (gst_audio_resample_start); GST_BASE_TRANSFORM_CLASS (klass)->stop = GST_DEBUG_FUNCPTR (gst_audio_resample_stop); GST_BASE_TRANSFORM_CLASS (klass)->transform_size = GST_DEBUG_FUNCPTR (gst_audio_resample_transform_size); GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size = GST_DEBUG_FUNCPTR (gst_audio_resample_get_unit_size); GST_BASE_TRANSFORM_CLASS (klass)->transform_caps = GST_DEBUG_FUNCPTR (gst_audio_resample_transform_caps); GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps = GST_DEBUG_FUNCPTR (gst_audio_resample_fixate_caps); GST_BASE_TRANSFORM_CLASS (klass)->set_caps = GST_DEBUG_FUNCPTR (gst_audio_resample_set_caps); GST_BASE_TRANSFORM_CLASS (klass)->transform = GST_DEBUG_FUNCPTR (gst_audio_resample_transform); GST_BASE_TRANSFORM_CLASS (klass)->sink_event = GST_DEBUG_FUNCPTR (gst_audio_resample_sink_event); GST_BASE_TRANSFORM_CLASS (klass)->transform_meta = GST_DEBUG_FUNCPTR (gst_audio_resample_transform_meta); GST_BASE_TRANSFORM_CLASS (klass)->submit_input_buffer = GST_DEBUG_FUNCPTR (gst_audio_resample_submit_input_buffer); GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE; gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_RESAMPLER_METHOD, 0); gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION, 0); gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE, 0); meta_tag_audio_quark = g_quark_from_static_string (GST_META_TAG_AUDIO_STR); } static void gst_audio_resample_init (GstAudioResample * resample) { GstBaseTransform *trans = GST_BASE_TRANSFORM (resample); resample->method = DEFAULT_RESAMPLE_METHOD; resample->quality = DEFAULT_QUALITY; resample->sinc_filter_mode = DEFAULT_SINC_FILTER_MODE; resample->sinc_filter_auto_threshold = DEFAULT_SINC_FILTER_AUTO_THRESHOLD; resample->sinc_filter_interpolation = DEFAULT_SINC_FILTER_INTERPOLATION; gst_base_transform_set_gap_aware (trans, TRUE); gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query); } /* vmethods */ static gboolean gst_audio_resample_start (GstBaseTransform * base) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); resample->need_discont = TRUE; resample->num_gap_samples = 0; resample->num_nongap_samples = 0; resample->t0 = GST_CLOCK_TIME_NONE; resample->in_offset0 = GST_BUFFER_OFFSET_NONE; resample->out_offset0 = GST_BUFFER_OFFSET_NONE; resample->samples_in = 0; resample->samples_out = 0; return TRUE; } static gboolean gst_audio_resample_stop (GstBaseTransform * base) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); if (resample->converter) { gst_audio_converter_free (resample->converter); resample->converter = NULL; } return TRUE; } static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps, gsize * size) { GstAudioInfo info; if (!gst_audio_info_from_caps (&info, caps)) goto invalid_caps; *size = GST_AUDIO_INFO_BPF (&info); return TRUE; /* ERRORS */ invalid_caps: { GST_ERROR_OBJECT (base, "invalid caps"); return FALSE; } } static GstCaps * gst_audio_resample_transform_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * filter) { const GValue *val; GstStructure *s; GstCaps *res; gint i, n; /* transform single caps into input_caps + input_caps with the rate * field set to our supported range. This ensures that upstream knows * about downstream's preferred rate(s) and can negotiate accordingly. */ res = gst_caps_new_empty (); n = gst_caps_get_size (caps); for (i = 0; i < n; i++) { s = gst_caps_get_structure (caps, i); /* If this is already expressed by the existing caps * skip this structure */ if (i > 0 && gst_caps_is_subset_structure (res, s)) continue; /* first, however, check if the caps contain a range for the rate field, in * which case that side isn't going to care much about the exact sample rate * chosen and we should just assume things will get fixated to something sane * and we may just as well offer our full range instead of the range in the * caps. If the rate is not an int range value, it's likely to express a * real preference or limitation and we should maintain that structure as * preference by putting it first into the transformed caps, and only add * our full rate range as second option */ s = gst_structure_copy (s); val = gst_structure_get_value (s, "rate"); if (val == NULL || GST_VALUE_HOLDS_INT_RANGE (val)) { /* overwrite existing range, or add field if it doesn't exist yet */ gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); } else { /* append caps with full range to existing caps with non-range rate field */ gst_caps_append_structure (res, gst_structure_copy (s)); gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); } gst_caps_append_structure (res, s); } if (filter) { GstCaps *intersection; intersection = gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (res); res = intersection; } return res; } /* Fixate rate to the allowed rate that has the smallest difference */ static GstCaps * gst_audio_resample_fixate_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * othercaps) { GstStructure *s; gint rate; s = gst_caps_get_structure (caps, 0); if (G_UNLIKELY (!gst_structure_get_int (s, "rate", &rate))) return othercaps; othercaps = gst_caps_truncate (othercaps); othercaps = gst_caps_make_writable (othercaps); s = gst_caps_get_structure (othercaps, 0); gst_structure_fixate_field_nearest_int (s, "rate", rate); return gst_caps_fixate (othercaps); } static GstStructure * make_options (GstAudioResample * resample, GstAudioInfo * in, GstAudioInfo * out) { GstStructure *options; options = gst_structure_new_empty ("resampler-options"); if (in != NULL && out != NULL) gst_audio_resampler_options_set_quality (resample->method, resample->quality, in->rate, out->rate, options); gst_structure_set (options, GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD, GST_TYPE_AUDIO_RESAMPLER_METHOD, resample->method, GST_AUDIO_RESAMPLER_OPT_FILTER_MODE, GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE, resample->sinc_filter_mode, GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD, G_TYPE_UINT, resample->sinc_filter_auto_threshold, GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION, GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION, resample->sinc_filter_interpolation, NULL); return options; } static gboolean gst_audio_resample_update_state (GstAudioResample * resample, GstAudioInfo * in, GstAudioInfo * out) { gboolean updated_latency = FALSE; gsize old_latency = -1; GstStructure *options; if (resample->converter == NULL && in == NULL && out == NULL) return TRUE; options = make_options (resample, in, out); if (resample->converter) old_latency = gst_audio_converter_get_max_latency (resample->converter); /* if channels and layout changed, destroy existing resampler */ if (in != NULL && (in->finfo != resample->in.finfo || in->channels != resample->in.channels || in->layout != resample->in.layout) && resample->converter) { gst_audio_converter_free (resample->converter); resample->converter = NULL; } if (resample->converter == NULL) { resample->converter = gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE, in, out, options); if (resample->converter == NULL) goto resampler_failed; } else if (in && out) { gboolean ret; ret = gst_audio_converter_update_config (resample->converter, in->rate, out->rate, options); if (!ret) goto update_failed; } else { gst_structure_free (options); } if (old_latency != -1) updated_latency = old_latency != gst_audio_converter_get_max_latency (resample->converter); if (updated_latency) gst_element_post_message (GST_ELEMENT (resample), gst_message_new_latency (GST_OBJECT (resample))); return TRUE; /* ERRORS */ resampler_failed: { GST_ERROR_OBJECT (resample, "failed to create resampler"); return FALSE; } update_failed: { GST_ERROR_OBJECT (resample, "failed to update resampler"); return FALSE; } } static void gst_audio_resample_reset_state (GstAudioResample * resample) { if (resample->converter) gst_audio_converter_reset (resample->converter); } static gboolean gst_audio_resample_transform_size (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps, gsize * othersize) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); gboolean ret = TRUE; gint bpf; GST_LOG_OBJECT (base, "asked to transform size %" G_GSIZE_FORMAT " in direction %s", size, direction == GST_PAD_SINK ? "SINK" : "SRC"); /* Number of samples in either buffer is size / (width*channels) -> * calculate the factor */ bpf = GST_AUDIO_INFO_BPF (&resample->in); /* Convert source buffer size to samples */ size /= bpf; if (direction == GST_PAD_SINK) { /* asked to convert size of an incoming buffer */ *othersize = gst_audio_converter_get_out_frames (resample->converter, size); *othersize *= bpf; } else { /* asked to convert size of an outgoing buffer */ *othersize = gst_audio_converter_get_in_frames (resample->converter, size); *othersize *= bpf; } GST_LOG_OBJECT (base, "transformed size %" G_GSIZE_FORMAT " to %" G_GSIZE_FORMAT, size * bpf, *othersize); return ret; } static gboolean gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); GstAudioInfo in, out; GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %" GST_PTR_FORMAT, incaps, outcaps); if (!gst_audio_info_from_caps (&in, incaps)) goto invalid_incaps; if (!gst_audio_info_from_caps (&out, outcaps)) goto invalid_outcaps; /* Reset timestamp tracking and drain the resampler if the audio format is * changing. Especially when changing the sample rate our timestamp tracking * will be completely off, but even otherwise we would usually lose the last * few samples if we don't drain here */ if (!gst_audio_info_is_equal (&in, &resample->in) || !gst_audio_info_is_equal (&out, &resample->out)) { if (resample->converter) { gsize latency = gst_audio_converter_get_max_latency (resample->converter); gst_audio_resample_push_drain (resample, latency); } gst_audio_resample_reset_state (resample); resample->num_gap_samples = 0; resample->num_nongap_samples = 0; resample->t0 = GST_CLOCK_TIME_NONE; resample->in_offset0 = GST_BUFFER_OFFSET_NONE; resample->out_offset0 = GST_BUFFER_OFFSET_NONE; resample->samples_in = 0; resample->samples_out = 0; resample->need_discont = TRUE; } gst_audio_resample_update_state (resample, &in, &out); resample->in = in; resample->out = out; return TRUE; /* ERROR */ invalid_incaps: { GST_ERROR_OBJECT (base, "invalid incaps"); return FALSE; } invalid_outcaps: { GST_ERROR_OBJECT (base, "invalid outcaps"); return FALSE; } } /* Push history_len zeros into the filter, but discard the output. */ static void gst_audio_resample_dump_drain (GstAudioResample * resample, guint history_len) { gsize out_len, outsize; GstBuffer *outbuf; GstAudioBuffer abuf; out_len = gst_audio_converter_get_out_frames (resample->converter, history_len); if (out_len == 0) return; outsize = out_len * resample->out.bpf; outbuf = gst_buffer_new_and_alloc (outsize); if (GST_AUDIO_INFO_LAYOUT (&resample->out) == GST_AUDIO_LAYOUT_NON_INTERLEAVED) { gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL); } gst_audio_buffer_map (&abuf, &resample->out, outbuf, GST_MAP_WRITE); gst_audio_converter_samples (resample->converter, 0, NULL, history_len, abuf.planes, out_len); gst_audio_buffer_unmap (&abuf); gst_buffer_unref (outbuf); } static void gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len) { GstBuffer *outbuf; GstFlowReturn res; gint outsize; gsize out_len; GstAudioBuffer abuf; g_assert (resample->converter != NULL); /* Don't drain samples if we were reset. */ if (!GST_CLOCK_TIME_IS_VALID (resample->t0)) return; out_len = gst_audio_converter_get_out_frames (resample->converter, history_len); if (out_len == 0) return; outsize = out_len * resample->in.bpf; outbuf = gst_buffer_new_and_alloc (outsize); if (GST_AUDIO_INFO_LAYOUT (&resample->out) == GST_AUDIO_LAYOUT_NON_INTERLEAVED) { gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL); } gst_audio_buffer_map (&abuf, &resample->out, outbuf, GST_MAP_WRITE); gst_audio_converter_samples (resample->converter, 0, NULL, history_len, abuf.planes, out_len); gst_audio_buffer_unmap (&abuf); /* time */ if (GST_CLOCK_TIME_IS_VALID (resample->t0)) { GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 + gst_util_uint64_scale_int_round (resample->samples_out, GST_SECOND, resample->out.rate); GST_BUFFER_DURATION (outbuf) = resample->t0 + gst_util_uint64_scale_int_round (resample->samples_out + out_len, GST_SECOND, resample->out.rate) - GST_BUFFER_TIMESTAMP (outbuf); } else { GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE; GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE; } /* offset */ if (resample->out_offset0 != GST_BUFFER_OFFSET_NONE) { GST_BUFFER_OFFSET (outbuf) = resample->out_offset0 + resample->samples_out; GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_len; } else { GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE; GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE; } /* move along */ resample->samples_out += out_len; resample->samples_in += history_len; GST_LOG_OBJECT (resample, "Pushing drain buffer of %u bytes with timestamp %" GST_TIME_FORMAT " duration %" GST_TIME_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %" G_GUINT64_FORMAT, outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf)); res = gst_pad_push (GST_BASE_TRANSFORM_SRC_PAD (resample), outbuf); if (G_UNLIKELY (res != GST_FLOW_OK)) GST_WARNING_OBJECT (resample, "Failed to push drain: %s", gst_flow_get_name (res)); return; } static gboolean gst_audio_resample_sink_event (GstBaseTransform * base, GstEvent * event) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_STOP: gst_audio_resample_reset_state (resample); resample->num_gap_samples = 0; resample->num_nongap_samples = 0; resample->t0 = GST_CLOCK_TIME_NONE; resample->in_offset0 = GST_BUFFER_OFFSET_NONE; resample->out_offset0 = GST_BUFFER_OFFSET_NONE; resample->samples_in = 0; resample->samples_out = 0; resample->need_discont = TRUE; break; case GST_EVENT_STREAM_START: case GST_EVENT_SEGMENT: case GST_EVENT_EOS: if (resample->converter) { gsize latency = gst_audio_converter_get_max_latency (resample->converter); gst_audio_resample_push_drain (resample, latency); } gst_audio_resample_reset_state (resample); resample->num_gap_samples = 0; resample->num_nongap_samples = 0; resample->t0 = GST_CLOCK_TIME_NONE; resample->in_offset0 = GST_BUFFER_OFFSET_NONE; resample->out_offset0 = GST_BUFFER_OFFSET_NONE; resample->samples_in = 0; resample->samples_out = 0; resample->need_discont = TRUE; break; default: break; } return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (base, event); } static gboolean gst_audio_resample_check_discont (GstAudioResample * resample, GstBuffer * buf) { guint64 offset; guint64 delta; /* is the incoming buffer a discontinuity? */ if (G_UNLIKELY (GST_BUFFER_IS_DISCONT (buf))) return TRUE; /* no valid timestamps or offsets to compare --> no discontinuity */ if (G_UNLIKELY (!(GST_BUFFER_TIMESTAMP_IS_VALID (buf) && GST_CLOCK_TIME_IS_VALID (resample->t0)))) return FALSE; /* convert the inbound timestamp to an offset. */ offset = gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf) - resample->t0, resample->in.rate, GST_SECOND); /* many elements generate imperfect streams due to rounding errors, so we * permit a small error (up to one sample) without triggering a filter * flush/restart (if triggered incorrectly, this will be audible) */ /* allow even up to more samples, since sink is not so strict anyway, * so give that one a chance to handle this as configured */ delta = ABS ((gint64) (offset - resample->samples_in)); if (delta <= (resample->in.rate >> 5)) return FALSE; GST_WARNING_OBJECT (resample, "encountered timestamp discontinuity of %" G_GUINT64_FORMAT " samples = %" GST_TIME_FORMAT, delta, GST_TIME_ARGS (gst_util_uint64_scale_int_round (delta, GST_SECOND, resample->in.rate))); return TRUE; } static GstFlowReturn gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf, GstBuffer * outbuf) { GstAudioBuffer srcabuf, dstabuf; gsize outsize; gsize in_len; gsize out_len; guint filt_len = gst_audio_converter_get_max_latency (resample->converter) * 2; gboolean inbuf_writable; inbuf_writable = gst_buffer_is_writable (inbuf) && gst_buffer_n_memory (inbuf) == 1 && gst_memory_is_writable (gst_buffer_peek_memory (inbuf, 0)); gst_audio_buffer_map (&srcabuf, &resample->in, inbuf, inbuf_writable ? GST_MAP_READWRITE : GST_MAP_READ); in_len = srcabuf.n_samples; out_len = gst_audio_converter_get_out_frames (resample->converter, in_len); GST_DEBUG_OBJECT (resample, "in %" G_GSIZE_FORMAT " frames, out %" G_GSIZE_FORMAT " frames", in_len, out_len); /* ensure that the output buffer is not bigger than what we need */ gst_buffer_set_size (outbuf, out_len * resample->in.bpf); if (GST_AUDIO_INFO_LAYOUT (&resample->out) == GST_AUDIO_LAYOUT_NON_INTERLEAVED) { gst_buffer_add_audio_meta (outbuf, &resample->out, out_len, NULL); } gst_audio_buffer_map (&dstabuf, &resample->out, outbuf, GST_MAP_WRITE); if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) { resample->num_nongap_samples = 0; if (resample->num_gap_samples < filt_len) { guint zeros_to_push; if (in_len >= filt_len - resample->num_gap_samples) zeros_to_push = filt_len - resample->num_gap_samples; else zeros_to_push = in_len; gst_audio_resample_push_drain (resample, zeros_to_push); in_len -= zeros_to_push; resample->num_gap_samples += zeros_to_push; } { guint num, den; gint i; num = resample->in.rate; den = resample->out.rate; if (resample->samples_in + in_len >= filt_len / 2) out_len = gst_util_uint64_scale_int_ceil (resample->samples_in + in_len - filt_len / 2, den, num) - resample->samples_out; else out_len = 0; for (i = 0; i < dstabuf.n_planes; i++) memset (dstabuf.planes[i], 0, GST_AUDIO_BUFFER_PLANE_SIZE (&dstabuf)); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP); resample->num_gap_samples += in_len; } } else { /* not a gap */ if (resample->num_gap_samples > filt_len) { /* push in enough zeros to restore the filter to the right offset */ guint num; num = resample->in.rate; gst_audio_resample_dump_drain (resample, (resample->num_gap_samples - filt_len) % num); } resample->num_gap_samples = 0; if (resample->num_nongap_samples < filt_len) { resample->num_nongap_samples += in_len; if (resample->num_nongap_samples > filt_len) resample->num_nongap_samples = filt_len; } { /* process */ GstAudioConverterFlags flags; flags = 0; if (inbuf_writable) flags |= GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE; gst_audio_converter_samples (resample->converter, flags, srcabuf.planes, in_len, dstabuf.planes, out_len); } } /* time */ if (GST_CLOCK_TIME_IS_VALID (resample->t0)) { GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 + gst_util_uint64_scale_int_round (resample->samples_out, GST_SECOND, resample->out.rate); GST_BUFFER_DURATION (outbuf) = resample->t0 + gst_util_uint64_scale_int_round (resample->samples_out + out_len, GST_SECOND, resample->out.rate) - GST_BUFFER_TIMESTAMP (outbuf); } else { GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE; GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE; } /* offset */ if (resample->out_offset0 != GST_BUFFER_OFFSET_NONE) { GST_BUFFER_OFFSET (outbuf) = resample->out_offset0 + resample->samples_out; GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_len; } else { GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE; GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE; } /* move along */ resample->samples_out += out_len; resample->samples_in += in_len; gst_audio_buffer_unmap (&srcabuf); gst_audio_buffer_unmap (&dstabuf); outsize = out_len * resample->in.bpf; GST_LOG_OBJECT (resample, "Converted to buffer of %" G_GSIZE_FORMAT " samples (%" G_GSIZE_FORMAT " bytes) with timestamp %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", offset_end %" G_GUINT64_FORMAT, out_len, outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf)); if (outsize == 0) return GST_BASE_TRANSFORM_FLOW_DROPPED; else return GST_FLOW_OK; } static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); GstFlowReturn ret; GST_LOG_OBJECT (resample, "transforming buffer of %" G_GSIZE_FORMAT " bytes," " ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT, gst_buffer_get_size (inbuf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)), GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf)); /* check for timestamp discontinuities; flush/reset if needed, and set * flag to resync timestamp and offset counters and send event * downstream */ if (G_UNLIKELY (gst_audio_resample_check_discont (resample, inbuf))) { if (resample->converter) { gsize latency = gst_audio_converter_get_max_latency (resample->converter); gst_audio_resample_push_drain (resample, latency); } gst_audio_resample_reset_state (resample); resample->need_discont = TRUE; } /* handle discontinuity */ if (G_UNLIKELY (resample->need_discont)) { resample->num_gap_samples = 0; resample->num_nongap_samples = 0; /* reset */ resample->samples_in = 0; resample->samples_out = 0; GST_DEBUG_OBJECT (resample, "found discontinuity; resyncing"); /* resync the timestamp and offset counters if possible */ if (GST_BUFFER_TIMESTAMP_IS_VALID (inbuf)) { resample->t0 = GST_BUFFER_TIMESTAMP (inbuf); } else { GST_DEBUG_OBJECT (resample, "... but new timestamp is invalid"); resample->t0 = GST_CLOCK_TIME_NONE; } if (GST_BUFFER_OFFSET_IS_VALID (inbuf)) { resample->in_offset0 = GST_BUFFER_OFFSET (inbuf); resample->out_offset0 = gst_util_uint64_scale_int_round (resample->in_offset0, resample->out.rate, resample->in.rate); } else { GST_DEBUG_OBJECT (resample, "... but new offset is invalid"); resample->in_offset0 = GST_BUFFER_OFFSET_NONE; resample->out_offset0 = GST_BUFFER_OFFSET_NONE; } /* set DISCONT flag on output buffer */ GST_DEBUG_OBJECT (resample, "marking this buffer with the DISCONT flag"); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); resample->need_discont = FALSE; } ret = gst_audio_resample_process (resample, inbuf, outbuf); if (G_UNLIKELY (ret != GST_FLOW_OK)) return ret; GST_DEBUG_OBJECT (resample, "input = samples [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ") = [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ") ns; output = samples [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ") = [%" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ") ns", GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf), GST_BUFFER_TIMESTAMP (inbuf), GST_BUFFER_TIMESTAMP (inbuf) + GST_BUFFER_DURATION (inbuf), GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf), GST_BUFFER_TIMESTAMP (outbuf), GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf)); return GST_FLOW_OK; } static gboolean gst_audio_resample_transform_meta (GstBaseTransform * trans, GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf) { const GstMetaInfo *info = meta->info; const gchar *const *tags; tags = gst_meta_api_type_get_tags (info->api); if (!tags || (g_strv_length ((gchar **) tags) == 1 && gst_meta_api_type_has_tag (info->api, META_TAG_AUDIO))) return TRUE; return FALSE; } static GstFlowReturn gst_audio_resample_submit_input_buffer (GstBaseTransform * base, gboolean is_discont, GstBuffer * input) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (base); if (base->segment.format == GST_FORMAT_TIME) { if (!GST_AUDIO_INFO_IS_VALID (&resample->in)) { GST_WARNING_OBJECT (resample, "Got buffer, but not negotiated yet!"); return GST_FLOW_NOT_NEGOTIATED; } input = gst_audio_buffer_clip (input, &base->segment, resample->in.rate, resample->in.bpf); if (!input) return GST_FLOW_OK; } return GST_BASE_TRANSFORM_CLASS (parent_class)->submit_input_buffer (base, is_discont, input); } static gboolean gst_audio_resample_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstAudioResample *resample = GST_AUDIO_RESAMPLE (parent); GstBaseTransform *trans; gboolean res = TRUE; trans = GST_BASE_TRANSFORM (resample); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: { GstClockTime min, max; gboolean live; guint64 latency; gint rate = resample->in.rate; gint resampler_latency; if (resample->converter) resampler_latency = gst_audio_converter_get_max_latency (resample->converter); else resampler_latency = 0; if (gst_base_transform_is_passthrough (trans)) resampler_latency = 0; if ((res = gst_pad_peer_query (GST_BASE_TRANSFORM_SINK_PAD (trans), query))) { gst_query_parse_latency (query, &live, &min, &max); GST_DEBUG_OBJECT (resample, "Peer latency: min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max)); /* add our own latency */ if (rate != 0 && resampler_latency != 0) latency = gst_util_uint64_scale_round (resampler_latency, GST_SECOND, rate); else latency = 0; GST_DEBUG_OBJECT (resample, "Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency)); min += latency; if (GST_CLOCK_TIME_IS_VALID (max)) max += latency; GST_DEBUG_OBJECT (resample, "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); } break; } default: res = gst_pad_query_default (pad, parent, query); break; } return res; } static void gst_audio_resample_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioResample *resample; resample = GST_AUDIO_RESAMPLE (object); switch (prop_id) { case PROP_QUALITY: /* FIXME locking! */ resample->quality = g_value_get_int (value); GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality); gst_audio_resample_update_state (resample, NULL, NULL); break; case PROP_RESAMPLE_METHOD: resample->method = g_value_get_enum (value); gst_audio_resample_update_state (resample, NULL, NULL); break; case PROP_SINC_FILTER_MODE: /* FIXME locking! */ resample->sinc_filter_mode = g_value_get_enum (value); gst_audio_resample_update_state (resample, NULL, NULL); break; case PROP_SINC_FILTER_AUTO_THRESHOLD: /* FIXME locking! */ resample->sinc_filter_auto_threshold = g_value_get_uint (value); gst_audio_resample_update_state (resample, NULL, NULL); break; case PROP_SINC_FILTER_INTERPOLATION: /* FIXME locking! */ resample->sinc_filter_interpolation = g_value_get_enum (value); gst_audio_resample_update_state (resample, NULL, NULL); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_resample_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioResample *resample; resample = GST_AUDIO_RESAMPLE (object); switch (prop_id) { case PROP_QUALITY: g_value_set_int (value, resample->quality); break; case PROP_RESAMPLE_METHOD: g_value_set_enum (value, resample->method); break; case PROP_SINC_FILTER_MODE: g_value_set_enum (value, resample->sinc_filter_mode); break; case PROP_SINC_FILTER_AUTO_THRESHOLD: g_value_set_uint (value, resample->sinc_filter_auto_threshold); break; case PROP_SINC_FILTER_INTERPOLATION: g_value_set_enum (value, resample->sinc_filter_interpolation); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean plugin_init (GstPlugin * plugin) { return GST_ELEMENT_REGISTER (audioresample, plugin); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, audioresample, "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);