/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2012> Collabora Ltd.
+ * Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
GST_DEBUG_CATEGORY_EXTERN (GST_CAT_PERFORMANCE);
-#define GST_TS_INFO_NONE &ts_info_none
-static const GstTSInfo ts_info_none = { -1, -1, -1, -1 };
-
-static const GstTSInfo *
-gst_ts_info_store (GstFFMpegAudDec * dec, GstClockTime dts, GstClockTime pts,
- GstClockTime duration, gint64 offset)
-{
- gint idx = dec->ts_idx;
- dec->ts_info[idx].idx = idx;
- dec->ts_info[idx].dts = dts;
- dec->ts_info[idx].pts = pts;
- dec->ts_info[idx].duration = duration;
- dec->ts_info[idx].offset = offset;
- dec->ts_idx = (idx + 1) & MAX_TS_MASK;
-
- return &dec->ts_info[idx];
-}
-
-static const GstTSInfo *
-gst_ts_info_get (GstFFMpegAudDec * dec, gint idx)
-{
- if (G_UNLIKELY (idx < 0 || idx > MAX_TS_MASK))
- return GST_TS_INFO_NONE;
-
- return &dec->ts_info[idx];
-}
-
/* A number of function prototypes are given so we can refer to them later. */
static void gst_ffmpegauddec_base_init (GstFFMpegAudDecClass * klass);
static void gst_ffmpegauddec_class_init (GstFFMpegAudDecClass * klass);
static void gst_ffmpegauddec_init (GstFFMpegAudDec * ffmpegdec);
static void gst_ffmpegauddec_finalize (GObject * object);
-static gboolean gst_ffmpegauddec_setcaps (GstFFMpegAudDec * ffmpegdec,
+static gboolean gst_ffmpegauddec_stop (GstAudioDecoder * decoder);
+static void gst_ffmpegauddec_flush (GstAudioDecoder * decoder, gboolean hard);
+static gboolean gst_ffmpegauddec_set_format (GstAudioDecoder * decoder,
GstCaps * caps);
-static gboolean gst_ffmpegauddec_sink_event (GstPad * pad, GstObject * parent,
- GstEvent * event);
-static gboolean gst_ffmpegauddec_sink_query (GstPad * pad, GstObject * parent,
- GstQuery * query);
-static GstFlowReturn gst_ffmpegauddec_chain (GstPad * pad, GstObject * parent,
- GstBuffer * buf);
-
-static GstStateChangeReturn gst_ffmpegauddec_change_state (GstElement * element,
- GstStateChange transition);
+static GstFlowReturn gst_ffmpegauddec_handle_frame (GstAudioDecoder * decoder,
+ GstBuffer * inbuf);
static gboolean gst_ffmpegauddec_negotiate (GstFFMpegAudDec * ffmpegdec,
gboolean force);
gst_ffmpegauddec_class_init (GstFFMpegAudDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
- GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+ GstAudioDecoderClass *gstaudiodecoder_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_ffmpegauddec_finalize;
- gstelement_class->change_state = gst_ffmpegauddec_change_state;
+ gstaudiodecoder_class->stop = GST_DEBUG_FUNCPTR (gst_ffmpegauddec_stop);
+ gstaudiodecoder_class->set_format =
+ GST_DEBUG_FUNCPTR (gst_ffmpegauddec_set_format);
+ gstaudiodecoder_class->handle_frame =
+ GST_DEBUG_FUNCPTR (gst_ffmpegauddec_handle_frame);
+ gstaudiodecoder_class->flush = GST_DEBUG_FUNCPTR (gst_ffmpegauddec_flush);
}
static void
gst_ffmpegauddec_init (GstFFMpegAudDec * ffmpegdec)
{
- GstFFMpegAudDecClass *oclass;
-
- oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
-
- /* setup pads */
- ffmpegdec->sinkpad = gst_pad_new_from_template (oclass->sinktempl, "sink");
- gst_pad_set_query_function (ffmpegdec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_ffmpegauddec_sink_query));
- gst_pad_set_event_function (ffmpegdec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_ffmpegauddec_sink_event));
- gst_pad_set_chain_function (ffmpegdec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_ffmpegauddec_chain));
- gst_element_add_pad (GST_ELEMENT (ffmpegdec), ffmpegdec->sinkpad);
-
- ffmpegdec->srcpad = gst_pad_new_from_template (oclass->srctempl, "src");
- gst_pad_use_fixed_caps (ffmpegdec->srcpad);
- gst_element_add_pad (GST_ELEMENT (ffmpegdec), ffmpegdec->srcpad);
-
/* some ffmpeg data */
ffmpegdec->context = avcodec_alloc_context ();
- ffmpegdec->pctx = NULL;
- ffmpegdec->pcache = NULL;
ffmpegdec->opened = FALSE;
- gst_segment_init (&ffmpegdec->segment, GST_FORMAT_TIME);
+ gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (ffmpegdec), TRUE);
+ gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (ffmpegdec), TRUE);
}
static void
if (ffmpegdec->context != NULL)
av_free (ffmpegdec->context);
+ ffmpegdec->context = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
-static void
-gst_ffmpegauddec_reset_ts (GstFFMpegAudDec * ffmpegdec)
-{
- ffmpegdec->next_out = GST_CLOCK_TIME_NONE;
-}
-
-/* with LOCK */
+/* With LOCK */
static void
gst_ffmpegauddec_close (GstFFMpegAudDec * ffmpegdec)
{
av_free (ffmpegdec->context->extradata);
ffmpegdec->context->extradata = NULL;
}
+}
- if (ffmpegdec->pctx) {
- if (ffmpegdec->pcache) {
- gst_buffer_unref (ffmpegdec->pcache);
- ffmpegdec->pcache = NULL;
- }
- av_parser_close (ffmpegdec->pctx);
- ffmpegdec->pctx = NULL;
- }
+static gboolean
+gst_ffmpegauddec_stop (GstAudioDecoder * decoder)
+{
+ GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) decoder;
+
+ GST_OBJECT_LOCK (ffmpegdec);
+ gst_ffmpegauddec_close (ffmpegdec);
+ GST_OBJECT_UNLOCK (ffmpegdec);
+ gst_audio_info_init (&ffmpegdec->info);
+ gst_caps_replace (&ffmpegdec->last_caps, NULL);
+
+ return TRUE;
}
/* with LOCK */
GST_LOG_OBJECT (ffmpegdec, "Opened libav codec %s, id %d",
oclass->in_plugin->name, oclass->in_plugin->id);
- if (!ffmpegdec->turnoff_parser) {
- ffmpegdec->pctx = av_parser_init (oclass->in_plugin->id);
- if (ffmpegdec->pctx)
- GST_LOG_OBJECT (ffmpegdec, "Using parser %p", ffmpegdec->pctx);
- else
- GST_LOG_OBJECT (ffmpegdec, "No parser for codec");
- } else {
- GST_LOG_OBJECT (ffmpegdec, "Parser deactivated for format");
- }
-
- ffmpegdec->samplerate = 0;
- ffmpegdec->channels = 0;
- ffmpegdec->depth = 0;
-
- gst_ffmpegauddec_reset_ts (ffmpegdec);
+ gst_audio_info_init (&ffmpegdec->info);
return TRUE;
}
static gboolean
-gst_ffmpegauddec_setcaps (GstFFMpegAudDec * ffmpegdec, GstCaps * caps)
+gst_ffmpegauddec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
+ GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) decoder;
GstFFMpegAudDecClass *oclass;
- GstStructure *structure;
gboolean ret = TRUE;
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
GST_OBJECT_LOCK (ffmpegdec);
+ if (ffmpegdec->last_caps && gst_caps_is_equal (ffmpegdec->last_caps, caps)) {
+ GST_DEBUG_OBJECT (ffmpegdec, "same caps");
+ GST_OBJECT_UNLOCK (ffmpegdec);
+ return TRUE;
+ }
+
+ gst_caps_replace (&ffmpegdec->last_caps, caps);
+
/* close old session */
if (ffmpegdec->opened) {
GST_OBJECT_UNLOCK (ffmpegdec);
avcodec_get_context_defaults (ffmpegdec->context);
}
- /* default is to let format decide if it needs a parser */
- ffmpegdec->turnoff_parser = FALSE;
-
/* get size and so */
gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id,
oclass->in_plugin->type, caps, ffmpegdec->context);
- /* get pixel aspect ratio if it's set */
- structure = gst_caps_get_structure (caps, 0);
-
- /* for AAC we only use av_parse if not on stream-format==raw or ==loas */
- if (oclass->in_plugin->id == CODEC_ID_AAC
- || oclass->in_plugin->id == CODEC_ID_AAC_LATM) {
- const gchar *format = gst_structure_get_string (structure, "stream-format");
-
- if (format == NULL || strcmp (format, "raw") == 0) {
- ffmpegdec->turnoff_parser = TRUE;
- }
- }
-
- /* for FLAC, don't parse if it's already parsed */
- if (oclass->in_plugin->id == CODEC_ID_FLAC) {
- if (gst_structure_has_field (structure, "streamheader"))
- ffmpegdec->turnoff_parser = TRUE;
- }
-
/* workaround encoder bugs */
ffmpegdec->context->workaround_bugs |= FF_BUG_AUTODETECT;
ffmpegdec->context->error_recognition = 1;
gst_ffmpegauddec_negotiate (GstFFMpegAudDec * ffmpegdec, gboolean force)
{
GstFFMpegAudDecClass *oclass;
- GstCaps *caps;
gint depth;
+ GstAudioFormat format;
GstAudioChannelPosition pos[64] = { 0, };
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
- depth = av_smp_format_depth (ffmpegdec->context->sample_fmt);
- gst_ffmpeg_channel_layout_to_gst (ffmpegdec->context, pos);
+ depth = av_smp_format_depth (ffmpegdec->context->sample_fmt) * 8;
+ format = gst_ffmpeg_smpfmt_to_audioformat (ffmpegdec->context->sample_fmt);
+ if (format == GST_AUDIO_FORMAT_UNKNOWN)
+ goto no_caps;
- if (!force && ffmpegdec->samplerate ==
+ if (!force && ffmpegdec->info.rate ==
ffmpegdec->context->sample_rate &&
- ffmpegdec->channels == ffmpegdec->context->channels &&
- ffmpegdec->depth == depth)
+ ffmpegdec->info.channels == ffmpegdec->context->channels &&
+ ffmpegdec->info.finfo->depth == depth)
return TRUE;
GST_DEBUG_OBJECT (ffmpegdec,
"Renegotiating audio from %dHz@%dchannels (%d) to %dHz@%dchannels (%d)",
- ffmpegdec->samplerate, ffmpegdec->channels,
- ffmpegdec->depth,
+ ffmpegdec->info.rate, ffmpegdec->info.channels,
+ ffmpegdec->info.finfo->depth,
ffmpegdec->context->sample_rate, ffmpegdec->context->channels, depth);
- ffmpegdec->samplerate = ffmpegdec->context->sample_rate;
- ffmpegdec->channels = ffmpegdec->context->channels;
- ffmpegdec->depth = depth;
+ gst_ffmpeg_channel_layout_to_gst (ffmpegdec->context, pos);
memcpy (ffmpegdec->ffmpeg_layout, pos,
sizeof (GstAudioChannelPosition) * ffmpegdec->context->channels);
/* Get GStreamer channel layout */
- memcpy (ffmpegdec->gst_layout,
- ffmpegdec->ffmpeg_layout,
- sizeof (GstAudioChannelPosition) * ffmpegdec->channels);
- gst_audio_channel_positions_to_valid_order (ffmpegdec->gst_layout,
- ffmpegdec->channels);
-
- caps = gst_ffmpeg_codectype_to_caps (oclass->in_plugin->type,
- ffmpegdec->context, oclass->in_plugin->id, FALSE);
-
- if (caps == NULL)
- goto no_caps;
+ gst_audio_channel_positions_to_valid_order (pos,
+ ffmpegdec->context->channels);
+ gst_audio_info_set_format (&ffmpegdec->info, format,
+ ffmpegdec->context->sample_rate, ffmpegdec->context->channels, pos);
- GST_LOG_OBJECT (ffmpegdec, "output caps %" GST_PTR_FORMAT, caps);
-
- if (!gst_pad_set_caps (ffmpegdec->srcpad, caps))
+ if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (ffmpegdec),
+ &ffmpegdec->info))
goto caps_failed;
- gst_caps_unref (caps);
-
return TRUE;
/* ERRORS */
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION, (NULL),
("Could not set caps for libav decoder (%s), not fixed?",
oclass->in_plugin->name));
- gst_caps_unref (caps);
return FALSE;
}
}
static void
-clear_queued (GstFFMpegAudDec * ffmpegdec)
-{
- g_list_foreach (ffmpegdec->queued, (GFunc) gst_mini_object_unref, NULL);
- g_list_free (ffmpegdec->queued);
- ffmpegdec->queued = NULL;
-}
-
-static GstFlowReturn
-flush_queued (GstFFMpegAudDec * ffmpegdec)
-{
- GstFlowReturn res = GST_FLOW_OK;
-
- while (ffmpegdec->queued) {
- GstBuffer *buf = GST_BUFFER_CAST (ffmpegdec->queued->data);
-
- GST_LOG_OBJECT (ffmpegdec, "pushing buffer %p, offset %"
- G_GUINT64_FORMAT ", timestamp %"
- GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
- GST_BUFFER_OFFSET (buf),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
-
- /* iterate ouput queue an push downstream */
- res = gst_pad_push (ffmpegdec->srcpad, buf);
-
- ffmpegdec->queued =
- g_list_delete_link (ffmpegdec->queued, ffmpegdec->queued);
- }
- return res;
-}
-
-static void
gst_avpacket_init (AVPacket * packet, guint8 * data, guint size)
{
memset (packet, 0, sizeof (AVPacket));
packet->size = size;
}
-/* returns TRUE if buffer is within segment, else FALSE.
- * if Buffer is on segment border, it's timestamp and duration will be clipped */
-static gboolean
-clip_audio_buffer (GstFFMpegAudDec * dec, GstBuffer ** buf)
-{
- *buf =
- gst_audio_buffer_clip (*buf, &dec->segment, dec->samplerate,
- dec->depth * dec->channels);
-
- return *buf != NULL;
-}
-
static gint
gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
AVCodec * in_plugin, guint8 * data, guint size,
- const GstTSInfo * dec_info, GstBuffer ** outbuf, GstFlowReturn * ret)
+ GstBuffer ** outbuf, GstFlowReturn * ret)
{
gint len = -1;
gint have_data = AVCODEC_MAX_AUDIO_FRAME_SIZE;
- GstClockTime out_pts, out_duration;
GstMapInfo map;
- gint64 out_offset;
int16_t *odata;
AVPacket packet;
- GST_DEBUG_OBJECT (ffmpegdec,
- "size:%d, offset:%" G_GINT64_FORMAT ", dts:%" GST_TIME_FORMAT ", pts:%"
- GST_TIME_FORMAT ", dur:%" GST_TIME_FORMAT ", ffmpegdec->next_out:%"
- GST_TIME_FORMAT, size, dec_info->offset, GST_TIME_ARGS (dec_info->dts),
- GST_TIME_ARGS (dec_info->pts), GST_TIME_ARGS (dec_info->duration),
- GST_TIME_ARGS (ffmpegdec->next_out));
+ GST_DEBUG_OBJECT (ffmpegdec, "size: %d", size);
- *outbuf = new_aligned_buffer (AVCODEC_MAX_AUDIO_FRAME_SIZE);
+ *outbuf =
+ gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (ffmpegdec),
+ AVCODEC_MAX_AUDIO_FRAME_SIZE);
gst_buffer_map (*outbuf, &map, GST_MAP_WRITE);
odata = (int16_t *) map.data;
"Decode audio: len=%d, have_data=%d", len, have_data);
if (len >= 0 && have_data > 0) {
- GstAudioFormat fmt;
-
/* Buffer size */
gst_buffer_unmap (*outbuf, &map);
gst_buffer_resize (*outbuf, 0, have_data);
goto beach;
}
- /*
- * Timestamps:
- *
- * 1) Copy input timestamp if valid
- * 2) else interpolate from previous input timestamp
- */
- /* always take timestamps from the input buffer if any */
- if (GST_CLOCK_TIME_IS_VALID (dec_info->pts)) {
- out_pts = dec_info->pts;
- } else {
- out_pts = ffmpegdec->next_out;
- }
-
- /*
- * Duration:
- *
- * 1) calculate based on number of samples
- */
- out_duration = gst_util_uint64_scale (have_data, GST_SECOND,
- ffmpegdec->depth * ffmpegdec->channels * ffmpegdec->samplerate);
-
- /* offset:
- *
- * Just copy
- */
- out_offset = dec_info->offset;
-
- GST_DEBUG_OBJECT (ffmpegdec,
- "Buffer created. Size:%d , pts:%" GST_TIME_FORMAT " , duration:%"
- GST_TIME_FORMAT, have_data,
- GST_TIME_ARGS (out_pts), GST_TIME_ARGS (out_duration));
-
- GST_BUFFER_PTS (*outbuf) = out_pts;
- GST_BUFFER_DURATION (*outbuf) = out_duration;
- GST_BUFFER_OFFSET (*outbuf) = out_offset;
-
- /* the next timestamp we'll use when interpolating */
- if (GST_CLOCK_TIME_IS_VALID (out_pts))
- ffmpegdec->next_out = out_pts + out_duration;
-
- /* now see if we need to clip the buffer against the segment boundaries. */
- if (G_UNLIKELY (!clip_audio_buffer (ffmpegdec, outbuf)))
- goto clipped;
+ GST_DEBUG_OBJECT (ffmpegdec, "Buffer created. Size: %d", have_data);
/* Reorder channels to the GStreamer channel order */
/* Only the width really matters here... and it's stored as depth */
- fmt =
- gst_audio_format_build_integer (TRUE, G_BYTE_ORDER,
- ffmpegdec->depth * 8, ffmpegdec->depth * 8);
-
- gst_audio_buffer_reorder_channels (*outbuf, fmt,
- ffmpegdec->channels, ffmpegdec->ffmpeg_layout, ffmpegdec->gst_layout);
+ gst_audio_buffer_reorder_channels (*outbuf, ffmpegdec->info.finfo->format,
+ ffmpegdec->info.channels, ffmpegdec->ffmpeg_layout,
+ ffmpegdec->info.position);
} else {
gst_buffer_unmap (*outbuf, &map);
gst_buffer_unref (*outbuf);
GST_DEBUG_OBJECT (ffmpegdec, "return flow %d, out %p, len %d",
*ret, *outbuf, len);
return len;
-
- /* ERRORS */
-clipped:
- {
- GST_DEBUG_OBJECT (ffmpegdec, "buffer clipped");
- if (*outbuf)
- gst_buffer_unref (*outbuf);
- *outbuf = NULL;
- goto beach;
- }
}
/* gst_ffmpegauddec_frame:
static gint
gst_ffmpegauddec_frame (GstFFMpegAudDec * ffmpegdec,
- guint8 * data, guint size, gint * got_data, const GstTSInfo * dec_info,
- GstFlowReturn * ret)
+ guint8 * data, guint size, gint * got_data, GstFlowReturn * ret)
{
GstFFMpegAudDecClass *oclass;
GstBuffer *outbuf = NULL;
if (G_UNLIKELY (ffmpegdec->context->codec == NULL))
goto no_codec;
- GST_LOG_OBJECT (ffmpegdec, "data:%p, size:%d, id:%d", data, size,
- dec_info->idx);
+ GST_LOG_OBJECT (ffmpegdec, "data:%p, size:%d", data, size);
*ret = GST_FLOW_OK;
ffmpegdec->context->frame_number++;
len =
gst_ffmpegauddec_audio_frame (ffmpegdec, oclass->in_plugin, data, size,
- dec_info, &outbuf, ret);
-
- /* if we did not get an output buffer and we have a pending discont, don't
- * clear the input timestamps, we will put them on the next buffer because
- * else we might create the first buffer with a very big timestamp gap. */
- if (outbuf == NULL && ffmpegdec->discont) {
- GST_DEBUG_OBJECT (ffmpegdec, "no buffer but keeping timestamp");
- ffmpegdec->clear_ts = FALSE;
- }
+ &outbuf, ret);
if (outbuf)
have_data = 1;
}
if (outbuf) {
- GST_LOG_OBJECT (ffmpegdec,
- "Decoded data, now pushing buffer %p with offset %" G_GINT64_FORMAT
- ", timestamp %" GST_TIME_FORMAT " and duration %" GST_TIME_FORMAT,
- outbuf, GST_BUFFER_OFFSET (outbuf),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
-
- /* mark pending discont */
- if (ffmpegdec->discont) {
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- ffmpegdec->discont = FALSE;
- }
- if (ffmpegdec->segment.rate > 0.0) {
- /* and off we go */
- *ret = gst_pad_push (ffmpegdec->srcpad, outbuf);
- } else {
- /* reverse playback, queue frame till later when we get a discont. */
- GST_DEBUG_OBJECT (ffmpegdec, "queued frame");
- ffmpegdec->queued = g_list_prepend (ffmpegdec->queued, outbuf);
- *ret = GST_FLOW_OK;
- }
+ GST_LOG_OBJECT (ffmpegdec, "Decoded data, now pushing buffer %p", outbuf);
+
+ *ret =
+ gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (ffmpegdec), outbuf,
+ 1);
} else {
GST_DEBUG_OBJECT (ffmpegdec, "We didn't get a decoded buffer");
}
do {
GstFlowReturn ret;
- len =
- gst_ffmpegauddec_frame (ffmpegdec, NULL, 0, &have_data, &ts_info_none,
- &ret);
+ len = gst_ffmpegauddec_frame (ffmpegdec, NULL, 0, &have_data, &ret);
if (len < 0 || have_data == 0)
break;
} while (try++ < 10);
}
- if (ffmpegdec->segment.rate < 0.0) {
- /* if we have some queued frames for reverse playback, flush them now */
- flush_queued (ffmpegdec);
- }
}
static void
-gst_ffmpegauddec_flush_pcache (GstFFMpegAudDec * ffmpegdec)
-{
- if (ffmpegdec->pctx) {
- gint size, bsize;
- guint8 *data;
- guint8 bdata[FF_INPUT_BUFFER_PADDING_SIZE];
-
- bsize = FF_INPUT_BUFFER_PADDING_SIZE;
- memset (bdata, 0, bsize);
-
- /* parse some dummy data to work around some ffmpeg weirdness where it keeps
- * the previous pts around */
- av_parser_parse2 (ffmpegdec->pctx, ffmpegdec->context,
- &data, &size, bdata, bsize, -1, -1, -1);
- ffmpegdec->pctx->pts = -1;
- ffmpegdec->pctx->dts = -1;
- }
-
- if (ffmpegdec->pcache) {
- gst_buffer_unref (ffmpegdec->pcache);
- ffmpegdec->pcache = NULL;
- }
-}
-
-static gboolean
-gst_ffmpegauddec_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
-{
- GstFFMpegAudDec *ffmpegdec;
- gboolean ret = FALSE;
-
- ffmpegdec = (GstFFMpegAudDec *) parent;
-
- GST_DEBUG_OBJECT (ffmpegdec, "Handling %s event",
- GST_EVENT_TYPE_NAME (event));
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_EOS:
- {
- gst_ffmpegauddec_drain (ffmpegdec);
- break;
- }
- case GST_EVENT_FLUSH_STOP:
- {
- if (ffmpegdec->opened) {
- avcodec_flush_buffers (ffmpegdec->context);
- }
- gst_ffmpegauddec_reset_ts (ffmpegdec);
- gst_ffmpegauddec_flush_pcache (ffmpegdec);
- gst_segment_init (&ffmpegdec->segment, GST_FORMAT_TIME);
- clear_queued (ffmpegdec);
- break;
- }
- case GST_EVENT_CAPS:
- {
- GstCaps *caps;
-
- gst_event_parse_caps (event, &caps);
-
- if (!ffmpegdec->last_caps
- || !gst_caps_is_equal (ffmpegdec->last_caps, caps)) {
- ret = gst_ffmpegauddec_setcaps (ffmpegdec, caps);
- if (ret) {
- gst_caps_replace (&ffmpegdec->last_caps, caps);
- }
- } else {
- ret = TRUE;
- }
-
- gst_event_unref (event);
- goto done;
- }
- case GST_EVENT_SEGMENT:
- {
- GstSegment segment;
-
- gst_event_copy_segment (event, &segment);
-
- switch (segment.format) {
- case GST_FORMAT_TIME:
- /* fine, our native segment format */
- break;
- case GST_FORMAT_BYTES:
- {
- gint bit_rate;
-
- bit_rate = ffmpegdec->context->bit_rate;
-
- /* convert to time or fail */
- if (!bit_rate)
- goto no_bitrate;
-
- GST_DEBUG_OBJECT (ffmpegdec, "bitrate: %d", bit_rate);
-
- /* convert values to TIME */
- if (segment.start != -1)
- segment.start =
- gst_util_uint64_scale_int (segment.start, GST_SECOND, bit_rate);
- if (segment.stop != -1)
- segment.stop =
- gst_util_uint64_scale_int (segment.stop, GST_SECOND, bit_rate);
- if (segment.time != -1)
- segment.time =
- gst_util_uint64_scale_int (segment.time, GST_SECOND, bit_rate);
-
- /* unref old event */
- gst_event_unref (event);
-
- /* create new converted time segment */
- segment.format = GST_FORMAT_TIME;
- /* FIXME, bitrate is not good enough too find a good stop, let's
- * hope start and time were 0... meh. */
- segment.stop = -1;
- event = gst_event_new_segment (&segment);
- break;
- }
- default:
- /* invalid format */
- goto invalid_format;
- }
-
- GST_DEBUG_OBJECT (ffmpegdec, "SEGMENT in time %" GST_SEGMENT_FORMAT,
- &segment);
-
- /* and store the values */
- gst_segment_copy_into (&segment, &ffmpegdec->segment);
- break;
- }
- default:
- break;
- }
-
- /* and push segment downstream */
- ret = gst_pad_push_event (ffmpegdec->srcpad, event);
-
-done:
-
- return ret;
-
- /* ERRORS */
-no_bitrate:
- {
- GST_WARNING_OBJECT (ffmpegdec, "no bitrate to convert BYTES to TIME");
- gst_event_unref (event);
- goto done;
- }
-invalid_format:
- {
- GST_WARNING_OBJECT (ffmpegdec, "unknown format received in NEWSEGMENT");
- gst_event_unref (event);
- goto done;
- }
-}
-
-static gboolean
-gst_ffmpegauddec_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
+gst_ffmpegauddec_flush (GstAudioDecoder * decoder, gboolean hard)
{
- GstFFMpegAudDec *ffmpegdec;
- gboolean ret = FALSE;
-
- ffmpegdec = (GstFFMpegAudDec *) parent;
-
- GST_DEBUG_OBJECT (ffmpegdec, "Handling %s query",
- GST_QUERY_TYPE_NAME (query));
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_ACCEPT_CAPS:
- {
- GstPadTemplate *templ;
+ GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) decoder;
- ret = FALSE;
- if ((templ = GST_PAD_PAD_TEMPLATE (pad))) {
- GstCaps *tcaps;
-
- if ((tcaps = GST_PAD_TEMPLATE_CAPS (templ))) {
- GstCaps *caps;
-
- gst_query_parse_accept_caps (query, &caps);
- gst_query_set_accept_caps_result (query,
- gst_caps_is_subset (caps, tcaps));
- ret = TRUE;
- }
- }
- break;
- }
- default:
- ret = gst_pad_query_default (pad, parent, query);
- break;
+ if (ffmpegdec->opened) {
+ avcodec_flush_buffers (ffmpegdec->context);
}
- return ret;
}
static GstFlowReturn
-gst_ffmpegauddec_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
+gst_ffmpegauddec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstFFMpegAudDec *ffmpegdec;
GstFFMpegAudDecClass *oclass;
GstMapInfo map;
gint size, bsize, len, have_data;
GstFlowReturn ret = GST_FLOW_OK;
- GstClockTime in_pts, in_dts, in_duration;
- gboolean discont;
- gint64 in_offset;
- const GstTSInfo *in_info;
- const GstTSInfo *dec_info;
- ffmpegdec = (GstFFMpegAudDec *) parent;
+ ffmpegdec = (GstFFMpegAudDec *) decoder;
if (G_UNLIKELY (!ffmpegdec->opened))
goto not_negotiated;
- discont = GST_BUFFER_IS_DISCONT (inbuf);
-
- /* The discont flags marks a buffer that is not continuous with the previous
- * buffer. This means we need to clear whatever data we currently have. We let
- * ffmpeg continue with the data that it has. We currently drain the old
- * frames that might be inside the decoder and we clear any partial data in
- * the pcache, we might be able to remove the drain and flush too. */
- if (G_UNLIKELY (discont)) {
- GST_DEBUG_OBJECT (ffmpegdec, "received DISCONT");
- /* drain what we have queued */
+ if (inbuf == NULL) {
gst_ffmpegauddec_drain (ffmpegdec);
- gst_ffmpegauddec_flush_pcache (ffmpegdec);
- ffmpegdec->discont = TRUE;
- gst_ffmpegauddec_reset_ts (ffmpegdec);
- }
- /* by default we clear the input timestamp after decoding each frame so that
- * interpollation can work. */
- ffmpegdec->clear_ts = TRUE;
-
- oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
-
- /* parse cache joining. If there is cached data */
- if (ffmpegdec->pcache) {
- /* join with previous data */
- GST_LOG_OBJECT (ffmpegdec, "join parse cache");
- inbuf = gst_buffer_append (ffmpegdec->pcache, inbuf);
- /* no more cached data, we assume we can consume the complete cache */
- ffmpegdec->pcache = NULL;
+ return GST_FLOW_OK;
}
- in_dts = GST_BUFFER_DTS (inbuf);
- in_pts = GST_BUFFER_PTS (inbuf);
- in_duration = GST_BUFFER_DURATION (inbuf);
- in_offset = GST_BUFFER_OFFSET (inbuf);
+ inbuf = gst_buffer_ref (inbuf);
- /* get handle to timestamp info, we can pass this around to ffmpeg */
- in_info =
- gst_ts_info_store (ffmpegdec, in_dts, in_pts, in_duration, in_offset);
+ oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
GST_LOG_OBJECT (ffmpegdec,
"Received new data of size %u, offset:%" G_GUINT64_FORMAT ", ts:%"
- GST_TIME_FORMAT ", dur:%" GST_TIME_FORMAT ", info %d",
+ GST_TIME_FORMAT ", dur:%" GST_TIME_FORMAT,
gst_buffer_get_size (inbuf), GST_BUFFER_OFFSET (inbuf),
- GST_TIME_ARGS (in_pts), GST_TIME_ARGS (in_duration), in_info->idx);
+ GST_TIME_ARGS (GST_BUFFER_PTS (inbuf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)));
/* workarounds, functions write to buffers:
* libavcodec/svq1.c:svq1_decode_frame writes to the given buffer.
bdata = map.data;
bsize = map.size;
- GST_LOG_OBJECT (ffmpegdec,
- "Received new data of size %u, offset:%" G_GUINT64_FORMAT ", dts:%"
- GST_TIME_FORMAT ", pts:%" GST_TIME_FORMAT ", dur:%" GST_TIME_FORMAT
- ", info %d", bsize, in_offset, GST_TIME_ARGS (in_dts),
- GST_TIME_ARGS (in_pts), GST_TIME_ARGS (in_duration), in_info->idx);
-
do {
- /* parse, if at all possible */
- if (ffmpegdec->pctx) {
- gint res;
-
- GST_LOG_OBJECT (ffmpegdec,
- "Calling av_parser_parse2 with offset %" G_GINT64_FORMAT ", ts:%"
- GST_TIME_FORMAT " size %d", in_offset, GST_TIME_ARGS (in_pts), bsize);
-
- /* feed the parser. We pass the timestamp info so that we can recover all
- * info again later */
- res = av_parser_parse2 (ffmpegdec->pctx, ffmpegdec->context,
- &data, &size, bdata, bsize, in_info->idx, in_info->idx, in_offset);
-
- GST_LOG_OBJECT (ffmpegdec,
- "parser returned res %d and size %d, id %" G_GINT64_FORMAT, res, size,
- (gint64) ffmpegdec->pctx->pts);
-
- /* store pts for decoding */
- if (ffmpegdec->pctx->pts != AV_NOPTS_VALUE && ffmpegdec->pctx->pts != -1)
- dec_info = gst_ts_info_get (ffmpegdec, ffmpegdec->pctx->pts);
- else {
- /* ffmpeg sometimes loses track after a flush, help it by feeding a
- * valid start time */
- ffmpegdec->pctx->pts = in_info->idx;
- ffmpegdec->pctx->dts = in_info->idx;
- dec_info = in_info;
- }
-
- GST_LOG_OBJECT (ffmpegdec, "consuming %d bytes. id %d", size,
- dec_info->idx);
-
- if (res) {
- /* there is output, set pointers for next round. */
- bsize -= res;
- bdata += res;
- } else {
- /* Parser did not consume any data, make sure we don't clear the
- * timestamp for the next round */
- ffmpegdec->clear_ts = FALSE;
- }
-
- /* if there is no output, we must break and wait for more data. also the
- * timestamp in the context is not updated. */
- if (size == 0) {
- if (bsize > 0)
- continue;
- else
- break;
- }
- } else {
- data = bdata;
- size = bsize;
-
- dec_info = in_info;
- }
+ data = bdata;
+ size = bsize;
/* decode a frame of audio now */
- len =
- gst_ffmpegauddec_frame (ffmpegdec, data, size, &have_data, dec_info,
- &ret);
+ len = gst_ffmpegauddec_frame (ffmpegdec, data, size, &have_data, &ret);
if (ret != GST_FLOW_OK) {
GST_LOG_OBJECT (ffmpegdec, "breaking because of flow ret %s",
bsize = 0;
break;
}
- if (!ffmpegdec->pctx) {
- if (len == 0 && !have_data) {
- /* nothing was decoded, this could be because no data was available or
- * because we were skipping frames.
- * If we have no context we must exit and wait for more data, we keep the
- * data we tried. */
- GST_LOG_OBJECT (ffmpegdec, "Decoding didn't return any data, breaking");
- break;
- } else if (len < 0) {
- /* a decoding error happened, we must break and try again with next data. */
- GST_LOG_OBJECT (ffmpegdec, "Decoding error, breaking");
- bsize = 0;
- break;
- }
- /* prepare for the next round, for codecs with a context we did this
- * already when using the parser. */
- bsize -= len;
- bdata += len;
- } else {
- if (len == 0) {
- /* nothing was decoded, this could be because no data was available or
- * because we were skipping frames. Since we have a parser we can
- * continue with the next frame */
- GST_LOG_OBJECT (ffmpegdec,
- "Decoding didn't return any data, trying next");
- } else if (len < 0) {
- /* we have a context that will bring us to the next frame */
- GST_LOG_OBJECT (ffmpegdec, "Decoding error, trying next");
- }
- }
- /* make sure we don't use the same old timestamp for the next frame and let
- * the interpollation take care of it. */
- if (ffmpegdec->clear_ts) {
- in_dts = GST_CLOCK_TIME_NONE;
- in_pts = GST_CLOCK_TIME_NONE;
- in_duration = GST_CLOCK_TIME_NONE;
- in_offset = GST_BUFFER_OFFSET_NONE;
- in_info = GST_TS_INFO_NONE;
- } else {
- ffmpegdec->clear_ts = TRUE;
+ if (len == 0 && !have_data) {
+ /* nothing was decoded, this could be because no data was available or
+ * because we were skipping frames.
+ * If we have no context we must exit and wait for more data, we keep the
+ * data we tried. */
+ GST_LOG_OBJECT (ffmpegdec, "Decoding didn't return any data, breaking");
+ break;
+ } else if (len < 0) {
+ /* a decoding error happened, we must break and try again with next data. */
+ GST_LOG_OBJECT (ffmpegdec, "Decoding error, breaking");
+ bsize = 0;
+ break;
}
+ /* prepare for the next round, for codecs with a context we did this
+ * already when using the parser. */
+ bsize -= len;
+ bdata += len;
GST_LOG_OBJECT (ffmpegdec, "Before (while bsize>0). bsize:%d , bdata:%p",
bsize, bdata);
} while (bsize > 0);
gst_buffer_unmap (inbuf, &map);
+ gst_buffer_unref (inbuf);
- /* keep left-over */
- if (ffmpegdec->pctx && bsize > 0) {
- in_pts = GST_BUFFER_PTS (inbuf);
- in_dts = GST_BUFFER_DTS (inbuf);
- in_offset = GST_BUFFER_OFFSET (inbuf);
-
- GST_LOG_OBJECT (ffmpegdec,
- "Keeping %d bytes of data with offset %" G_GINT64_FORMAT ", pts %"
- GST_TIME_FORMAT, bsize, in_offset, GST_TIME_ARGS (in_pts));
-
- ffmpegdec->pcache = gst_buffer_copy_region (inbuf, GST_BUFFER_COPY_ALL,
- gst_buffer_get_size (inbuf) - bsize, bsize);
- /* we keep timestamp, even though all we really know is that the correct
- * timestamp is not below the one from inbuf */
- GST_BUFFER_PTS (ffmpegdec->pcache) = in_pts;
- GST_BUFFER_DTS (ffmpegdec->pcache) = in_dts;
- GST_BUFFER_OFFSET (ffmpegdec->pcache) = in_offset;
- } else if (bsize > 0) {
+ if (bsize > 0) {
GST_DEBUG_OBJECT (ffmpegdec, "Dropping %d bytes of data", bsize);
}
- gst_buffer_unref (inbuf);
return ret;
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION, (NULL),
("avdec_%s: input format was not set before data start",
oclass->in_plugin->name));
- gst_buffer_unref (inbuf);
return GST_FLOW_NOT_NEGOTIATED;
}
}
-static GstStateChangeReturn
-gst_ffmpegauddec_change_state (GstElement * element, GstStateChange transition)
-{
- GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) element;
- GstStateChangeReturn ret;
-
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- switch (transition) {
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- GST_OBJECT_LOCK (ffmpegdec);
- gst_ffmpegauddec_close (ffmpegdec);
- GST_OBJECT_UNLOCK (ffmpegdec);
- clear_queued (ffmpegdec);
- break;
- default:
- break;
- }
-
- return ret;
-}
-
gboolean
gst_ffmpegauddec_register (GstPlugin * plugin)
{
if (!type) {
/* create the gtype now */
- type = g_type_register_static (GST_TYPE_ELEMENT, type_name, &typeinfo, 0);
+ type =
+ g_type_register_static (GST_TYPE_AUDIO_DECODER, type_name, &typeinfo,
+ 0);
g_type_set_qdata (type, GST_FFDEC_PARAMS_QDATA, (gpointer) in_plugin);
}
projects / platform / upstream / gst-libav.git / commitdiff