| /* GStreamer |
| * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> |
| * |
| * 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-audiorate |
| * @see_also: #GstVideoRate |
| * |
| * This element takes an incoming stream of timestamped raw audio frames and |
| * produces a perfect stream by inserting or dropping samples as needed. |
| * |
| * This operation may be of use to link to elements that require or otherwise |
| * implicitly assume a perfect stream as they do not store timestamps, |
| * but derive this by some means (e.g. bitrate for some AVI cases). |
| * |
| * The properties #GstAudioRate:in, #GstAudioRate:out, #GstAudioRate:add |
| * and #GstAudioRate:drop can be read to obtain information about number of |
| * input samples, output samples, dropped samples (i.e. the number of unused |
| * input samples) and inserted samples (i.e. the number of samples added to |
| * stream). |
| * |
| * When the #GstAudioRate:silent property is set to FALSE, a GObject property |
| * notification will be emitted whenever one of the #GstAudioRate:add or |
| * #GstAudioRate:drop values changes. |
| * This can potentially cause performance degradation. |
| * Note that property notification will happen from the streaming thread, so |
| * applications should be prepared for this. |
| * |
| * If the #GstAudioRate:tolerance property is non-zero, and an incoming buffer's |
| * timestamp deviates less than the property indicates from what would make a |
| * 'perfect time', then no samples will be added or dropped. |
| * Note that the output is still guaranteed to be a perfect stream, which means |
| * that the incoming data is then simply shifted (by less than the indicated |
| * tolerance) to a perfect time. |
| * |
| * <refsect2> |
| * <title>Example pipelines</title> |
| * |[ |
| * gst-launch-1.0 -v autoaudiosrc ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav |
| * ]| Capture audio from the sound card and turn it into a perfect stream |
| * for saving in a raw audio file. |
| * |[ |
| * gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.file ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav |
| * ]| Decodes an audio file and transforms it into a perfect stream for saving |
| * in a raw audio WAV file. Without the audio rate, the timing might not be |
| * preserved correctly in the WAV file in case the decoded stream is jittery |
| * or there are samples missing. |
| * </refsect2> |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include "config.h" |
| #endif |
| |
| #include <string.h> |
| #include <stdlib.h> |
| |
| #include "gstaudiorate.h" |
| |
| #define GST_CAT_DEFAULT audio_rate_debug |
| GST_DEBUG_CATEGORY_STATIC (audio_rate_debug); |
| |
| /* GstAudioRate signals and args */ |
| enum |
| { |
| /* FILL ME */ |
| LAST_SIGNAL |
| }; |
| |
| #define DEFAULT_SILENT TRUE |
| #define DEFAULT_TOLERANCE (40 * GST_MSECOND) |
| #define DEFAULT_SKIP_TO_FIRST FALSE |
| |
| enum |
| { |
| PROP_0, |
| PROP_IN, |
| PROP_OUT, |
| PROP_ADD, |
| PROP_DROP, |
| PROP_SILENT, |
| PROP_TOLERANCE, |
| PROP_SKIP_TO_FIRST |
| }; |
| |
| static GstStaticPadTemplate gst_audio_rate_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) |
| ", layout = (string) { interleaved, non-interleaved }") |
| ); |
| |
| static GstStaticPadTemplate gst_audio_rate_sink_template = |
| GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) |
| ", layout = (string) { interleaved, non-interleaved }") |
| ); |
| |
| static gboolean gst_audio_rate_sink_event (GstPad * pad, GstObject * parent, |
| GstEvent * event); |
| static gboolean gst_audio_rate_src_event (GstPad * pad, GstObject * parent, |
| GstEvent * event); |
| static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstObject * parent, |
| GstBuffer * buf); |
| |
| static void gst_audio_rate_set_property (GObject * object, |
| guint prop_id, const GValue * value, GParamSpec * pspec); |
| static void gst_audio_rate_get_property (GObject * object, |
| guint prop_id, GValue * value, GParamSpec * pspec); |
| |
| static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element, |
| GstStateChange transition); |
| |
| /*static guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */ |
| |
| static GParamSpec *pspec_drop = NULL; |
| static GParamSpec *pspec_add = NULL; |
| |
| #define gst_audio_rate_parent_class parent_class |
| G_DEFINE_TYPE (GstAudioRate, gst_audio_rate, GST_TYPE_ELEMENT); |
| |
| static void |
| gst_audio_rate_class_init (GstAudioRateClass * klass) |
| { |
| GObjectClass *object_class = G_OBJECT_CLASS (klass); |
| GstElementClass *element_class = GST_ELEMENT_CLASS (klass); |
| |
| object_class->set_property = gst_audio_rate_set_property; |
| object_class->get_property = gst_audio_rate_get_property; |
| |
| g_object_class_install_property (object_class, PROP_IN, |
| g_param_spec_uint64 ("in", "In", |
| "Number of input samples", 0, G_MAXUINT64, 0, |
| G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); |
| g_object_class_install_property (object_class, PROP_OUT, |
| g_param_spec_uint64 ("out", "Out", "Number of output samples", 0, |
| G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); |
| pspec_add = g_param_spec_uint64 ("add", "Add", "Number of added samples", |
| 0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS); |
| g_object_class_install_property (object_class, PROP_ADD, pspec_add); |
| pspec_drop = g_param_spec_uint64 ("drop", "Drop", "Number of dropped samples", |
| 0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS); |
| g_object_class_install_property (object_class, PROP_DROP, pspec_drop); |
| g_object_class_install_property (object_class, PROP_SILENT, |
| g_param_spec_boolean ("silent", "silent", |
| "Don't emit notify for dropped and duplicated frames", DEFAULT_SILENT, |
| G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| /** |
| * GstAudioRate:tolerance: |
| * |
| * The difference between incoming timestamp and next timestamp must exceed |
| * the given value for audiorate to add or drop samples. |
| */ |
| g_object_class_install_property (object_class, PROP_TOLERANCE, |
| g_param_spec_uint64 ("tolerance", "tolerance", |
| "Only act if timestamp jitter/imperfection exceeds indicated tolerance (ns)", |
| 0, G_MAXUINT64, DEFAULT_TOLERANCE, |
| G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| /** |
| * GstAudioRate:skip-to-first: |
| * |
| * Don't produce buffers before the first one we receive. |
| */ |
| g_object_class_install_property (object_class, PROP_SKIP_TO_FIRST, |
| g_param_spec_boolean ("skip-to-first", "Skip to first buffer", |
| "Don't produce buffers before the first one we receive", |
| DEFAULT_SKIP_TO_FIRST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| gst_element_class_set_static_metadata (element_class, |
| "Audio rate adjuster", "Filter/Effect/Audio", |
| "Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream", |
| "Wim Taymans <wim@fluendo.com>"); |
| |
| gst_element_class_add_static_pad_template (element_class, |
| &gst_audio_rate_sink_template); |
| gst_element_class_add_static_pad_template (element_class, |
| &gst_audio_rate_src_template); |
| |
| element_class->change_state = gst_audio_rate_change_state; |
| } |
| |
| static void |
| gst_audio_rate_reset (GstAudioRate * audiorate) |
| { |
| audiorate->next_offset = -1; |
| audiorate->next_ts = -1; |
| audiorate->discont = TRUE; |
| gst_segment_init (&audiorate->sink_segment, GST_FORMAT_UNDEFINED); |
| gst_segment_init (&audiorate->src_segment, GST_FORMAT_TIME); |
| |
| GST_DEBUG_OBJECT (audiorate, "handle reset"); |
| } |
| |
| static gboolean |
| gst_audio_rate_setcaps (GstAudioRate * audiorate, GstCaps * caps) |
| { |
| GstAudioInfo info; |
| |
| if (!gst_audio_info_from_caps (&info, caps)) |
| goto wrong_caps; |
| |
| audiorate->info = info; |
| |
| return TRUE; |
| |
| /* ERRORS */ |
| wrong_caps: |
| { |
| GST_DEBUG_OBJECT (audiorate, "could not parse caps"); |
| return FALSE; |
| } |
| } |
| |
| static void |
| gst_audio_rate_init (GstAudioRate * audiorate) |
| { |
| audiorate->sinkpad = |
| gst_pad_new_from_static_template (&gst_audio_rate_sink_template, "sink"); |
| gst_pad_set_event_function (audiorate->sinkpad, gst_audio_rate_sink_event); |
| gst_pad_set_chain_function (audiorate->sinkpad, gst_audio_rate_chain); |
| GST_PAD_SET_PROXY_CAPS (audiorate->sinkpad); |
| gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad); |
| |
| audiorate->srcpad = |
| gst_pad_new_from_static_template (&gst_audio_rate_src_template, "src"); |
| gst_pad_set_event_function (audiorate->srcpad, gst_audio_rate_src_event); |
| GST_PAD_SET_PROXY_CAPS (audiorate->srcpad); |
| gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad); |
| |
| audiorate->in = 0; |
| audiorate->out = 0; |
| audiorate->drop = 0; |
| audiorate->add = 0; |
| audiorate->silent = DEFAULT_SILENT; |
| audiorate->tolerance = DEFAULT_TOLERANCE; |
| } |
| |
| static void |
| gst_audio_rate_fill_to_time (GstAudioRate * audiorate, GstClockTime time) |
| { |
| GstBuffer *buf; |
| |
| GST_DEBUG_OBJECT (audiorate, "next_ts: %" GST_TIME_FORMAT |
| ", filling to %" GST_TIME_FORMAT, GST_TIME_ARGS (audiorate->next_ts), |
| GST_TIME_ARGS (time)); |
| |
| if (!GST_CLOCK_TIME_IS_VALID (time) || |
| !GST_CLOCK_TIME_IS_VALID (audiorate->next_ts)) |
| return; |
| |
| /* feed an empty buffer to chain with the given timestamp, |
| * it will take care of filling */ |
| buf = gst_buffer_new (); |
| GST_BUFFER_TIMESTAMP (buf) = time; |
| gst_audio_rate_chain (audiorate->sinkpad, GST_OBJECT_CAST (audiorate), buf); |
| } |
| |
| static gboolean |
| gst_audio_rate_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) |
| { |
| gboolean res; |
| GstAudioRate *audiorate; |
| |
| audiorate = GST_AUDIO_RATE (parent); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| case GST_EVENT_CAPS: |
| { |
| GstCaps *caps; |
| |
| gst_event_parse_caps (event, &caps); |
| if ((res = gst_audio_rate_setcaps (audiorate, caps))) { |
| res = gst_pad_push_event (audiorate->srcpad, event); |
| } else { |
| gst_event_unref (event); |
| } |
| break; |
| } |
| case GST_EVENT_FLUSH_STOP: |
| GST_DEBUG_OBJECT (audiorate, "handling FLUSH_STOP"); |
| gst_audio_rate_reset (audiorate); |
| res = gst_pad_push_event (audiorate->srcpad, event); |
| break; |
| case GST_EVENT_SEGMENT: |
| { |
| gst_event_copy_segment (event, &audiorate->sink_segment); |
| |
| GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT"); |
| #if 0 |
| /* FIXME: bad things will likely happen if rate < 0 ... */ |
| if (!update) { |
| /* a new segment starts. We need to figure out what will be the next |
| * sample offset. We mark the offsets as invalid so that the _chain |
| * function will perform this calculation. */ |
| gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop); |
| #endif |
| audiorate->next_offset = -1; |
| audiorate->next_ts = -1; |
| #if 0 |
| } else { |
| gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.start); |
| } |
| #endif |
| |
| GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT, |
| &audiorate->sink_segment); |
| |
| if (audiorate->sink_segment.format == GST_FORMAT_TIME) { |
| /* TIME formats can be copied to src and forwarded */ |
| res = gst_pad_push_event (audiorate->srcpad, event); |
| gst_segment_copy_into (&audiorate->sink_segment, |
| &audiorate->src_segment); |
| } else { |
| /* other formats will be handled in the _chain function */ |
| gst_event_unref (event); |
| res = TRUE; |
| } |
| break; |
| } |
| case GST_EVENT_EOS: |
| /* Fill segment until the end */ |
| if (GST_CLOCK_TIME_IS_VALID (audiorate->src_segment.stop)) |
| gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop); |
| res = gst_pad_push_event (audiorate->srcpad, event); |
| break; |
| case GST_EVENT_GAP: |
| { |
| /* Fill until end of gap */ |
| GstClockTime timestamp, duration; |
| gst_event_parse_gap (event, ×tamp, &duration); |
| gst_event_unref (event); |
| if (GST_CLOCK_TIME_IS_VALID (timestamp)) { |
| if (GST_CLOCK_TIME_IS_VALID (duration)) |
| timestamp += duration; |
| gst_audio_rate_fill_to_time (audiorate, timestamp); |
| } |
| res = TRUE; |
| break; |
| } |
| default: |
| res = gst_pad_event_default (pad, parent, event); |
| break; |
| } |
| |
| return res; |
| } |
| |
| static gboolean |
| gst_audio_rate_src_event (GstPad * pad, GstObject * parent, GstEvent * event) |
| { |
| gboolean res; |
| GstAudioRate *audiorate; |
| |
| audiorate = GST_AUDIO_RATE (parent); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| default: |
| res = gst_pad_push_event (audiorate->sinkpad, event); |
| break; |
| } |
| |
| return res; |
| } |
| |
| static gboolean |
| gst_audio_rate_convert (GstAudioRate * audiorate, |
| GstFormat src_fmt, guint64 src_val, GstFormat dest_fmt, guint64 * dest_val) |
| { |
| return gst_audio_info_convert (&audiorate->info, src_fmt, src_val, dest_fmt, |
| (gint64 *) dest_val); |
| } |
| |
| |
| static gboolean |
| gst_audio_rate_convert_segments (GstAudioRate * audiorate) |
| { |
| GstFormat src_fmt, dst_fmt; |
| |
| src_fmt = audiorate->sink_segment.format; |
| dst_fmt = audiorate->src_segment.format; |
| |
| #define CONVERT_VAL(field) gst_audio_rate_convert (audiorate, \ |
| src_fmt, audiorate->sink_segment.field, \ |
| dst_fmt, &audiorate->src_segment.field); |
| |
| audiorate->sink_segment.rate = audiorate->src_segment.rate; |
| audiorate->sink_segment.flags = audiorate->src_segment.flags; |
| audiorate->sink_segment.applied_rate = audiorate->src_segment.applied_rate; |
| CONVERT_VAL (start); |
| CONVERT_VAL (stop); |
| CONVERT_VAL (time); |
| CONVERT_VAL (base); |
| CONVERT_VAL (position); |
| #undef CONVERT_VAL |
| |
| return TRUE; |
| } |
| |
| static void |
| gst_audio_rate_notify_drop (GstAudioRate * audiorate) |
| { |
| g_object_notify_by_pspec ((GObject *) audiorate, pspec_drop); |
| } |
| |
| static void |
| gst_audio_rate_notify_add (GstAudioRate * audiorate) |
| { |
| g_object_notify_by_pspec ((GObject *) audiorate, pspec_add); |
| } |
| |
| static GstFlowReturn |
| gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) |
| { |
| GstAudioRate *audiorate; |
| GstClockTime in_time; |
| guint64 in_offset, in_offset_end, in_samples; |
| guint in_size; |
| GstFlowReturn ret = GST_FLOW_OK; |
| GstClockTimeDiff diff; |
| gint rate, bpf; |
| |
| audiorate = GST_AUDIO_RATE (parent); |
| |
| rate = GST_AUDIO_INFO_RATE (&audiorate->info); |
| bpf = GST_AUDIO_INFO_BPF (&audiorate->info); |
| |
| /* need to be negotiated now */ |
| if (bpf == 0) |
| goto not_negotiated; |
| |
| /* we have a new pending segment */ |
| if (audiorate->next_offset == -1) { |
| gint64 pos; |
| |
| /* update the TIME segment */ |
| gst_audio_rate_convert_segments (audiorate); |
| |
| /* first buffer, we are negotiated and we have a segment, calculate the |
| * current expected offsets based on the segment.start, which is the first |
| * media time of the segment and should match the media time of the first |
| * buffer in that segment, which is the offset expressed in DEFAULT units. |
| */ |
| /* convert first timestamp of segment to sample position */ |
| pos = gst_util_uint64_scale_int_round (audiorate->src_segment.start, |
| GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND); |
| |
| GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos); |
| |
| /* resyncing is a discont */ |
| audiorate->discont = TRUE; |
| |
| audiorate->next_offset = pos; |
| audiorate->next_ts = |
| gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND, |
| GST_AUDIO_INFO_RATE (&audiorate->info)); |
| |
| if (audiorate->skip_to_first && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { |
| GST_DEBUG_OBJECT (audiorate, "but skipping to first buffer instead"); |
| pos = gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf), |
| GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND); |
| GST_DEBUG_OBJECT (audiorate, "so resync to offset %" G_GINT64_FORMAT, |
| pos); |
| audiorate->next_offset = pos; |
| audiorate->next_ts = GST_BUFFER_TIMESTAMP (buf); |
| } |
| } |
| |
| in_time = GST_BUFFER_TIMESTAMP (buf); |
| if (in_time == GST_CLOCK_TIME_NONE) { |
| GST_DEBUG_OBJECT (audiorate, "no timestamp, using expected next time"); |
| in_time = audiorate->next_ts; |
| } |
| |
| in_size = gst_buffer_get_size (buf); |
| in_samples = in_size / bpf; |
| audiorate->in += in_samples; |
| |
| /* calculate the buffer offset */ |
| in_offset = gst_util_uint64_scale_int_round (in_time, rate, GST_SECOND); |
| in_offset_end = in_offset + in_samples; |
| |
| GST_LOG_OBJECT (audiorate, |
| "in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT |
| ", in_size:%u, in_offset:%" G_GUINT64_FORMAT ", in_offset_end:%" |
| G_GUINT64_FORMAT ", ->next_offset:%" G_GUINT64_FORMAT ", ->next_ts:%" |
| GST_TIME_FORMAT, GST_TIME_ARGS (in_time), |
| GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (in_samples, rate)), |
| in_size, in_offset, in_offset_end, audiorate->next_offset, |
| GST_TIME_ARGS (audiorate->next_ts)); |
| |
| diff = in_time - audiorate->next_ts; |
| if (diff <= (GstClockTimeDiff) audiorate->tolerance && |
| diff >= (GstClockTimeDiff) - audiorate->tolerance) { |
| /* buffer time close enough to expected time, |
| * so produce a perfect stream by simply 'shifting' |
| * it to next ts and offset and sending */ |
| GST_LOG_OBJECT (audiorate, "within tolerance %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (audiorate->tolerance)); |
| /* The outgoing buffer's offset will be set to ->next_offset, we also |
| * need to adjust the offset_end value accordingly */ |
| in_offset_end = audiorate->next_offset + in_samples; |
| audiorate->out += in_samples; |
| goto send; |
| } |
| |
| /* do we need to insert samples */ |
| if (in_offset > audiorate->next_offset) { |
| GstBuffer *fill; |
| gint fillsize; |
| guint64 fillsamples; |
| |
| /* We don't want to allocate a single unreasonably huge buffer - it might |
| be hundreds of megabytes. So, limit each output buffer to one second of |
| audio */ |
| fillsamples = in_offset - audiorate->next_offset; |
| |
| while (fillsamples > 0) { |
| guint64 cursamples = MIN (fillsamples, rate); |
| GstMapInfo fillmap; |
| |
| fillsamples -= cursamples; |
| fillsize = cursamples * bpf; |
| |
| fill = gst_buffer_new_and_alloc (fillsize); |
| |
| gst_buffer_map (fill, &fillmap, GST_MAP_WRITE); |
| gst_audio_format_fill_silence (audiorate->info.finfo, fillmap.data, |
| fillmap.size); |
| gst_buffer_unmap (fill, &fillmap); |
| |
| GST_DEBUG_OBJECT (audiorate, "inserting %" G_GUINT64_FORMAT " samples", |
| cursamples); |
| |
| GST_BUFFER_OFFSET (fill) = audiorate->next_offset; |
| audiorate->next_offset += cursamples; |
| GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset; |
| |
| /* Use next timestamp, then calculate following timestamp based on |
| * offset to get duration. Necessary complexity to get 'perfect' |
| * streams */ |
| GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts; |
| audiorate->next_ts = |
| gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND, |
| rate); |
| GST_BUFFER_DURATION (fill) = |
| audiorate->next_ts - GST_BUFFER_TIMESTAMP (fill); |
| |
| /* we created this buffer to fill a gap */ |
| GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP); |
| /* set discont if it's pending, this is mostly done for the first buffer |
| * and after a flushing seek */ |
| if (audiorate->discont) { |
| GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT); |
| audiorate->discont = FALSE; |
| } |
| |
| fill = gst_audio_buffer_clip (fill, &audiorate->src_segment, rate, bpf); |
| if (fill) |
| ret = gst_pad_push (audiorate->srcpad, fill); |
| |
| if (ret != GST_FLOW_OK) |
| goto beach; |
| audiorate->out += cursamples; |
| audiorate->add += cursamples; |
| |
| if (!audiorate->silent) |
| gst_audio_rate_notify_add (audiorate); |
| } |
| |
| } else if (in_offset < audiorate->next_offset) { |
| /* need to remove samples */ |
| if (in_offset_end <= audiorate->next_offset) { |
| guint64 drop = in_size / bpf; |
| |
| audiorate->drop += drop; |
| |
| GST_DEBUG_OBJECT (audiorate, "dropping %" G_GUINT64_FORMAT " samples", |
| drop); |
| |
| /* we can drop the buffer completely */ |
| gst_buffer_unref (buf); |
| buf = NULL; |
| |
| if (!audiorate->silent) |
| gst_audio_rate_notify_drop (audiorate); |
| |
| goto beach; |
| } else { |
| guint64 truncsamples; |
| guint truncsize, leftsize; |
| GstBuffer *trunc; |
| |
| /* truncate buffer */ |
| truncsamples = audiorate->next_offset - in_offset; |
| truncsize = truncsamples * bpf; |
| leftsize = in_size - truncsize; |
| |
| trunc = |
| gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, truncsize, |
| leftsize); |
| |
| gst_buffer_unref (buf); |
| buf = trunc; |
| |
| audiorate->drop += truncsamples; |
| audiorate->out += (leftsize / bpf); |
| GST_DEBUG_OBJECT (audiorate, "truncating %" G_GUINT64_FORMAT " samples", |
| truncsamples); |
| |
| if (!audiorate->silent) |
| gst_audio_rate_notify_drop (audiorate); |
| } |
| } |
| |
| send: |
| if (gst_buffer_get_size (buf) == 0) |
| goto beach; |
| |
| /* Now calculate parameters for whichever buffer (either the original |
| * or truncated one) we're pushing. */ |
| GST_BUFFER_OFFSET (buf) = audiorate->next_offset; |
| GST_BUFFER_OFFSET_END (buf) = in_offset_end; |
| |
| GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts; |
| audiorate->next_ts = gst_util_uint64_scale_int_round (in_offset_end, |
| GST_SECOND, rate); |
| GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf); |
| |
| if (audiorate->discont) { |
| /* we need to output a discont buffer, do so now */ |
| GST_DEBUG_OBJECT (audiorate, "marking DISCONT on output buffer"); |
| buf = gst_buffer_make_writable (buf); |
| GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); |
| audiorate->discont = FALSE; |
| } else if (GST_BUFFER_IS_DISCONT (buf)) { |
| /* else we make everything continuous so we can safely remove the DISCONT |
| * flag from the buffer if there was one */ |
| GST_DEBUG_OBJECT (audiorate, "removing DISCONT from buffer"); |
| buf = gst_buffer_make_writable (buf); |
| GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT); |
| } |
| |
| buf = gst_audio_buffer_clip (buf, &audiorate->src_segment, rate, bpf); |
| if (buf) { |
| /* set last_stop on segment */ |
| audiorate->src_segment.position = |
| GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); |
| |
| ret = gst_pad_push (audiorate->srcpad, buf); |
| } |
| buf = NULL; |
| |
| audiorate->next_offset = in_offset_end; |
| beach: |
| |
| if (buf) |
| gst_buffer_unref (buf); |
| |
| return ret; |
| |
| /* ERRORS */ |
| not_negotiated: |
| { |
| gst_buffer_unref (buf); |
| |
| GST_ELEMENT_ERROR (audiorate, STREAM, FORMAT, |
| (NULL), ("pipeline error, format was not negotiated")); |
| return GST_FLOW_NOT_NEGOTIATED; |
| } |
| } |
| |
| static void |
| gst_audio_rate_set_property (GObject * object, |
| guint prop_id, const GValue * value, GParamSpec * pspec) |
| { |
| GstAudioRate *audiorate = GST_AUDIO_RATE (object); |
| |
| switch (prop_id) { |
| case PROP_SILENT: |
| audiorate->silent = g_value_get_boolean (value); |
| break; |
| case PROP_TOLERANCE: |
| audiorate->tolerance = g_value_get_uint64 (value); |
| break; |
| case PROP_SKIP_TO_FIRST: |
| audiorate->skip_to_first = g_value_get_boolean (value); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_audio_rate_get_property (GObject * object, |
| guint prop_id, GValue * value, GParamSpec * pspec) |
| { |
| GstAudioRate *audiorate = GST_AUDIO_RATE (object); |
| |
| switch (prop_id) { |
| case PROP_IN: |
| g_value_set_uint64 (value, audiorate->in); |
| break; |
| case PROP_OUT: |
| g_value_set_uint64 (value, audiorate->out); |
| break; |
| case PROP_ADD: |
| g_value_set_uint64 (value, audiorate->add); |
| break; |
| case PROP_DROP: |
| g_value_set_uint64 (value, audiorate->drop); |
| break; |
| case PROP_SILENT: |
| g_value_set_boolean (value, audiorate->silent); |
| break; |
| case PROP_TOLERANCE: |
| g_value_set_uint64 (value, audiorate->tolerance); |
| break; |
| case PROP_SKIP_TO_FIRST: |
| g_value_set_boolean (value, audiorate->skip_to_first); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static GstStateChangeReturn |
| gst_audio_rate_change_state (GstElement * element, GstStateChange transition) |
| { |
| GstAudioRate *audiorate = GST_AUDIO_RATE (element); |
| |
| switch (transition) { |
| case GST_STATE_CHANGE_PAUSED_TO_READY: |
| break; |
| case GST_STATE_CHANGE_READY_TO_PAUSED: |
| audiorate->in = 0; |
| audiorate->out = 0; |
| audiorate->drop = 0; |
| audiorate->add = 0; |
| gst_audio_info_init (&audiorate->info); |
| gst_audio_rate_reset (audiorate); |
| break; |
| default: |
| break; |
| } |
| |
| return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); |
| } |
| |
| static gboolean |
| plugin_init (GstPlugin * plugin) |
| { |
| GST_DEBUG_CATEGORY_INIT (audio_rate_debug, "audiorate", 0, |
| "AudioRate stream fixer"); |
| |
| return gst_element_register (plugin, "audiorate", GST_RANK_NONE, |
| GST_TYPE_AUDIO_RATE); |
| } |
| |
| GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, |
| GST_VERSION_MINOR, |
| audiorate, |
| "Adjusts audio frames", |
| plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) |