/* 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
static void gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc);
static void gst_ffmpegaudenc_finalize (GObject * object);
-static gboolean gst_ffmpegaudenc_setcaps (GstFFMpegAudEnc * ffmpegenc,
- GstCaps * caps);
-static GstCaps *gst_ffmpegaudenc_getcaps (GstFFMpegAudEnc * ffmpegenc,
+static GstCaps *gst_ffmpegaudenc_getcaps (GstAudioEncoder * encoder,
GstCaps * filter);
-static GstFlowReturn gst_ffmpegaudenc_chain_audio (GstPad * pad,
- GstObject * parent, GstBuffer * buffer);
-static gboolean gst_ffmpegaudenc_query_sink (GstPad * pad, GstObject * parent,
- GstQuery * query);
-static gboolean gst_ffmpegaudenc_event_sink (GstPad * pad, GstObject * parent,
- GstEvent * event);
+static gboolean gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder,
+ GstAudioInfo * info);
+static GstFlowReturn gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder,
+ GstBuffer * inbuf);
+static gboolean gst_ffmpegaudenc_stop (GstAudioEncoder * encoder);
static void gst_ffmpegaudenc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_ffmpegaudenc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
-static GstStateChangeReturn gst_ffmpegaudenc_change_state (GstElement * element,
- GstStateChange transition);
-
#define GST_FFENC_PARAMS_QDATA g_quark_from_static_string("avenc-params")
static GstElementClass *parent_class = NULL;
gst_ffmpegaudenc_class_init (GstFFMpegAudEncClass * klass)
{
GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
+ GstAudioEncoderClass *gstaudioencoder_class;
gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
+ gstaudioencoder_class = (GstAudioEncoderClass *) klass;
parent_class = g_type_class_peek_parent (klass);
"Target Audio Bitrate", 0, G_MAXINT, DEFAULT_AUDIO_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- gstelement_class->change_state = gst_ffmpegaudenc_change_state;
-
gobject_class->finalize = gst_ffmpegaudenc_finalize;
+
+ gstaudioencoder_class->stop = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_stop);
+ gstaudioencoder_class->getcaps = GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_getcaps);
+ gstaudioencoder_class->set_format =
+ GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_set_format);
+ gstaudioencoder_class->handle_frame =
+ GST_DEBUG_FUNCPTR (gst_ffmpegaudenc_handle_frame);
}
static void
gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc)
{
- GstFFMpegAudEncClass *oclass =
- (GstFFMpegAudEncClass *) (G_OBJECT_GET_CLASS (ffmpegaudenc));
-
- /* setup pads */
- ffmpegaudenc->sinkpad = gst_pad_new_from_template (oclass->sinktempl, "sink");
- gst_pad_set_event_function (ffmpegaudenc->sinkpad,
- gst_ffmpegaudenc_event_sink);
- gst_pad_set_query_function (ffmpegaudenc->sinkpad,
- gst_ffmpegaudenc_query_sink);
- gst_pad_set_chain_function (ffmpegaudenc->sinkpad,
- gst_ffmpegaudenc_chain_audio);
-
- ffmpegaudenc->srcpad = gst_pad_new_from_template (oclass->srctempl, "src");
- gst_pad_use_fixed_caps (ffmpegaudenc->srcpad);
-
/* ffmpeg objects */
ffmpegaudenc->context = avcodec_alloc_context ();
ffmpegaudenc->opened = FALSE;
- gst_element_add_pad (GST_ELEMENT (ffmpegaudenc), ffmpegaudenc->sinkpad);
- gst_element_add_pad (GST_ELEMENT (ffmpegaudenc), ffmpegaudenc->srcpad);
-
- ffmpegaudenc->adapter = gst_adapter_new ();
+ gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (ffmpegaudenc), FALSE);
}
static void
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) object;
+ /* clean up remaining allocated data */
+ av_free (ffmpegaudenc->context);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_ffmpegaudenc_stop (GstAudioEncoder * encoder)
+{
+ GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
/* close old session */
if (ffmpegaudenc->opened) {
ffmpegaudenc->opened = FALSE;
}
- /* clean up remaining allocated data */
- av_free (ffmpegaudenc->context);
-
- g_object_unref (ffmpegaudenc->adapter);
-
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ return TRUE;
}
static GstCaps *
-gst_ffmpegaudenc_getcaps (GstFFMpegAudEnc * ffmpegaudenc, GstCaps * filter)
+gst_ffmpegaudenc_getcaps (GstAudioEncoder * encoder, GstCaps * filter)
{
+ GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
GstCaps *caps = NULL;
GST_DEBUG_OBJECT (ffmpegaudenc, "getting caps");
/* audio needs no special care */
- caps = gst_pad_get_pad_template_caps (ffmpegaudenc->sinkpad);
-
- if (filter) {
- GstCaps *tmp;
- tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
- gst_caps_unref (caps);
- caps = tmp;
- }
+ caps = gst_audio_encoder_proxy_getcaps (encoder, NULL, filter);
GST_DEBUG_OBJECT (ffmpegaudenc,
"audio caps, return template %" GST_PTR_FORMAT, caps);
}
static gboolean
-gst_ffmpegaudenc_setcaps (GstFFMpegAudEnc * ffmpegaudenc, GstCaps * caps)
+gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
{
+ GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
GstCaps *other_caps;
GstCaps *allowed_caps;
GstCaps *icaps;
+ gsize frame_size;
GstFFMpegAudEncClass *oclass =
(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
ffmpegaudenc->context->scenechange_threshold = 0;
ffmpegaudenc->context->inter_threshold = 0;
-
/* fetch pix_fmt and so on */
- gst_ffmpeg_caps_with_codectype (oclass->in_plugin->type,
- caps, ffmpegaudenc->context);
+ gst_ffmpeg_audioinfo_to_context (info, ffmpegaudenc->context);
if (!ffmpegaudenc->context->time_base.den) {
ffmpegaudenc->context->time_base.den = 25;
ffmpegaudenc->context->time_base.num = 1;
if (gst_ffmpeg_avcodec_open (ffmpegaudenc->context, oclass->in_plugin) < 0) {
if (ffmpegaudenc->context->priv_data)
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
- if (ffmpegaudenc->context->stats_in)
- g_free (ffmpegaudenc->context->stats_in);
GST_DEBUG_OBJECT (ffmpegaudenc, "avenc_%s: Failed to open FFMPEG codec",
oclass->in_plugin->name);
return FALSE;
}
- /* second pass stats buffer no longer needed */
- if (ffmpegaudenc->context->stats_in)
- g_free (ffmpegaudenc->context->stats_in);
-
/* some codecs support more than one format, first auto-choose one */
GST_DEBUG_OBJECT (ffmpegaudenc, "picking an output format ...");
- allowed_caps = gst_pad_get_allowed_caps (ffmpegaudenc->srcpad);
+ allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
if (!allowed_caps) {
GST_DEBUG_OBJECT (ffmpegaudenc, "... but no peer, using template caps");
/* we need to copy because get_allowed_caps returns a ref, and
* get_pad_template_caps doesn't */
- allowed_caps = gst_pad_get_pad_template_caps (ffmpegaudenc->srcpad);
+ allowed_caps =
+ gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
}
GST_DEBUG_OBJECT (ffmpegaudenc, "chose caps %" GST_PTR_FORMAT, allowed_caps);
gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id,
gst_caps_unref (icaps);
return FALSE;
}
+ icaps = gst_caps_truncate (icaps);
- if (gst_caps_get_size (icaps) > 1) {
- GstCaps *newcaps;
-
- newcaps =
- gst_caps_new_full (gst_structure_copy (gst_caps_get_structure (icaps,
- 0)), NULL);
- gst_caps_unref (icaps);
- icaps = newcaps;
- }
-
- if (!gst_pad_set_caps (ffmpegaudenc->srcpad, icaps)) {
+ if (!gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (ffmpegaudenc),
+ icaps)) {
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
gst_caps_unref (icaps);
return FALSE;
}
gst_caps_unref (icaps);
+ frame_size = ffmpegaudenc->context->frame_size;
+ if (frame_size > 1) {
+ gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
+ frame_size);
+ gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
+ frame_size);
+ gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 1);
+ } else {
+ gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
+ 0);
+ gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
+ 0);
+ gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 0);
+ }
+
/* success! */
ffmpegaudenc->opened = TRUE;
static GstFlowReturn
gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
- guint8 * audio_in, guint in_size, guint max_size, GstClockTime timestamp,
- GstClockTime duration, gboolean discont)
+ guint8 * audio_in, guint in_size, guint max_size)
{
GstBuffer *outbuf;
AVCodecContext *ctx;
ctx = ffmpegaudenc->context;
/* We need to provide at least ffmpegs minimal buffer size */
- outbuf = gst_buffer_new_and_alloc (max_size + FF_MIN_BUFFER_SIZE);
+ outbuf =
+ gst_audio_encoder_allocate_output_buffer (GST_AUDIO_ENCODER
+ (ffmpegaudenc), max_size + FF_MIN_BUFFER_SIZE);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer of max size %d", max_size);
gst_buffer_unmap (outbuf, &map);
gst_buffer_resize (outbuf, 0, res);
- GST_BUFFER_PTS (outbuf) = timestamp;
- GST_BUFFER_DURATION (outbuf) = duration;
- if (discont)
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
-
- GST_LOG_OBJECT (ffmpegaudenc, "pushing size %d, timestamp %" GST_TIME_FORMAT,
- res, GST_TIME_ARGS (timestamp));
-
- ret = gst_pad_push (ffmpegaudenc->srcpad, outbuf);
+ if (res > 0) {
+ GST_LOG_OBJECT (ffmpegaudenc, "pushing size %d", res);
+ ret =
+ gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (ffmpegaudenc),
+ outbuf, 1);
+ } else {
+ GST_LOG_OBJECT (ffmpegaudenc, "no output produced");
+ gst_buffer_unref (outbuf);
+ ret = GST_FLOW_OK;
+ }
return ret;
}
static GstFlowReturn
-gst_ffmpegaudenc_chain_audio (GstPad * pad, GstObject * parent,
- GstBuffer * inbuf)
+gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
{
GstFFMpegAudEnc *ffmpegaudenc;
- GstFFMpegAudEncClass *oclass;
- AVCodecContext *ctx;
- GstClockTime timestamp, duration;
- gsize size, frame_size;
- gint osize;
+ gsize size;
GstFlowReturn ret;
gint out_size;
- gboolean discont;
guint8 *in_data;
+ GstMapInfo map;
- ffmpegaudenc = (GstFFMpegAudEnc *) parent;
- oclass = (GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
+ ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
if (G_UNLIKELY (!ffmpegaudenc->opened))
goto not_negotiated;
- ctx = ffmpegaudenc->context;
-
+ inbuf = gst_buffer_ref (inbuf);
size = gst_buffer_get_size (inbuf);
- timestamp = GST_BUFFER_PTS (inbuf);
- duration = GST_BUFFER_DURATION (inbuf);
- discont = GST_BUFFER_IS_DISCONT (inbuf);
GST_DEBUG_OBJECT (ffmpegaudenc,
"Received time %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT
- ", size %" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (duration), size);
-
- frame_size = ctx->frame_size;
- osize = av_get_bits_per_sample_format (ctx->sample_fmt) / 8;
+ ", size %" G_GSIZE_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)), size);
- if (frame_size > 1) {
- /* we have a frame_size, feed the encoder multiples of this frame size */
- guint avail, frame_bytes;
-
- if (discont) {
- GST_LOG_OBJECT (ffmpegaudenc, "DISCONT, clear adapter");
- gst_adapter_clear (ffmpegaudenc->adapter);
- ffmpegaudenc->discont = TRUE;
- }
-
- if (gst_adapter_available (ffmpegaudenc->adapter) == 0) {
- /* lock on to new timestamp */
- GST_LOG_OBJECT (ffmpegaudenc, "taking buffer timestamp %" GST_TIME_FORMAT,
- GST_TIME_ARGS (timestamp));
- ffmpegaudenc->adapter_ts = timestamp;
- ffmpegaudenc->adapter_consumed = 0;
- } else {
- GstClockTime upstream_time;
- GstClockTime consumed_time;
- guint64 bytes;
-
- /* use timestamp at head of the adapter */
- consumed_time =
- gst_util_uint64_scale (ffmpegaudenc->adapter_consumed, GST_SECOND,
- ctx->sample_rate);
- timestamp = ffmpegaudenc->adapter_ts + consumed_time;
- GST_LOG_OBJECT (ffmpegaudenc, "taking adapter timestamp %" GST_TIME_FORMAT
- " and adding consumed time %" GST_TIME_FORMAT,
- GST_TIME_ARGS (ffmpegaudenc->adapter_ts),
- GST_TIME_ARGS (consumed_time));
-
- /* check with upstream timestamps, if too much deviation,
- * forego some timestamp perfection in favour of upstream syncing
- * (particularly in case these do not happen to come in multiple
- * of frame size) */
- upstream_time = gst_adapter_prev_pts (ffmpegaudenc->adapter, &bytes);
- if (GST_CLOCK_TIME_IS_VALID (upstream_time)) {
- GstClockTimeDiff diff;
-
- upstream_time +=
- gst_util_uint64_scale (bytes, GST_SECOND,
- ctx->sample_rate * osize * ctx->channels);
- diff = upstream_time - timestamp;
- /* relaxed difference, rather than half a sample or so ... */
- if (diff > GST_SECOND / 10 || diff < -GST_SECOND / 10) {
- GST_DEBUG_OBJECT (ffmpegaudenc, "adapter timestamp drifting, "
- "taking upstream timestamp %" GST_TIME_FORMAT,
- GST_TIME_ARGS (upstream_time));
- timestamp = upstream_time;
- /* samples corresponding to bytes */
- ffmpegaudenc->adapter_consumed = bytes / (osize * ctx->channels);
- ffmpegaudenc->adapter_ts = upstream_time -
- gst_util_uint64_scale (ffmpegaudenc->adapter_consumed, GST_SECOND,
- ctx->sample_rate);
- ffmpegaudenc->discont = TRUE;
- }
- }
- }
+ /* 4 times the input size should be big enough... */
+ out_size = size * 4;
- GST_LOG_OBJECT (ffmpegaudenc, "pushing buffer in adapter");
- gst_adapter_push (ffmpegaudenc->adapter, inbuf);
+ gst_buffer_map (inbuf, &map, GST_MAP_READ);
+ in_data = map.data;
+ size = map.size;
+ ret = gst_ffmpegaudenc_encode_audio (ffmpegaudenc, in_data, size, out_size);
+ gst_buffer_unmap (inbuf, &map);
+ gst_buffer_unref (inbuf);
- /* first see how many bytes we need to feed to the decoder. */
- frame_bytes = frame_size * osize * ctx->channels;
- avail = gst_adapter_available (ffmpegaudenc->adapter);
-
- GST_LOG_OBJECT (ffmpegaudenc, "frame_bytes %u, avail %u", frame_bytes,
- avail);
-
- /* while there is more than a frame size in the adapter, consume it */
- while (avail >= frame_bytes) {
- GST_LOG_OBJECT (ffmpegaudenc, "taking %u bytes from the adapter",
- frame_bytes);
-
- /* Note that we take frame_bytes and add frame_size.
- * Makes sense when resyncing because you don't have to count channels
- * or samplesize to divide by the samplerate */
-
- /* take an audio buffer out of the adapter */
- in_data = (guint8 *) gst_adapter_map (ffmpegaudenc->adapter, frame_bytes);
- ffmpegaudenc->adapter_consumed += frame_size;
-
- /* calculate timestamp and duration relative to start of adapter and to
- * the amount of samples we consumed */
- duration =
- gst_util_uint64_scale (ffmpegaudenc->adapter_consumed, GST_SECOND,
- ctx->sample_rate);
- duration -= (timestamp - ffmpegaudenc->adapter_ts);
-
- /* 4 times the input size should be big enough... */
- out_size = frame_bytes * 4;
-
- ret =
- gst_ffmpegaudenc_encode_audio (ffmpegaudenc, in_data, frame_bytes,
- out_size, timestamp, duration, ffmpegaudenc->discont);
-
- gst_adapter_unmap (ffmpegaudenc->adapter);
- gst_adapter_flush (ffmpegaudenc->adapter, frame_bytes);
-
- if (ret != GST_FLOW_OK)
- goto push_failed;
-
- /* advance the adapter timestamp with the duration */
- timestamp += duration;
-
- ffmpegaudenc->discont = FALSE;
- avail = gst_adapter_available (ffmpegaudenc->adapter);
- }
- GST_LOG_OBJECT (ffmpegaudenc, "%u bytes left in the adapter", avail);
- } else {
- GstMapInfo map;
- /* we have no frame_size, feed the encoder all the data and expect a fixed
- * output size */
- int coded_bps = av_get_bits_per_sample (oclass->in_plugin->id);
-
- GST_LOG_OBJECT (ffmpegaudenc, "coded bps %d, osize %d", coded_bps, osize);
-
- out_size = size / osize;
- if (coded_bps)
- out_size = (out_size * coded_bps) / 8;
-
- gst_buffer_map (inbuf, &map, GST_MAP_READ);
- in_data = map.data;
- size = map.size;
- ret = gst_ffmpegaudenc_encode_audio (ffmpegaudenc, in_data, size, out_size,
- timestamp, duration, discont);
- gst_buffer_unmap (inbuf, &map);
- gst_buffer_unref (inbuf);
-
- if (ret != GST_FLOW_OK)
- goto push_failed;
- }
+ if (ret != GST_FLOW_OK)
+ goto push_failed;
return GST_FLOW_OK;
}
}
-static gboolean
-gst_ffmpegaudenc_event_sink (GstPad * pad, GstObject * parent, GstEvent * event)
-{
- GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) parent;
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_CAPS:
- {
- GstCaps *caps;
- gboolean ret;
-
- gst_event_parse_caps (event, &caps);
- ret = gst_ffmpegaudenc_setcaps (ffmpegaudenc, caps);
- gst_event_unref (event);
- return ret;
- }
- default:
- break;
- }
-
- return gst_pad_event_default (pad, parent, event);
-}
-
-static gboolean
-gst_ffmpegaudenc_query_sink (GstPad * pad, GstObject * parent, GstQuery * query)
-{
- GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) parent;
- gboolean res = FALSE;
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_CAPS:
- {
- GstCaps *filter, *caps;
-
- gst_query_parse_caps (query, &filter);
- caps = gst_ffmpegaudenc_getcaps (ffmpegaudenc, filter);
- gst_query_set_caps_result (query, caps);
- gst_caps_unref (caps);
- res = TRUE;
- break;
- }
- default:
- res = gst_pad_query_default (pad, parent, query);
- break;
- }
-
- return res;
-}
-
static void
gst_ffmpegaudenc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
}
}
-static GstStateChangeReturn
-gst_ffmpegaudenc_change_state (GstElement * element, GstStateChange transition)
-{
- GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) element;
- GstStateChangeReturn result;
-
- switch (transition) {
- default:
- break;
- }
-
- result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- switch (transition) {
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- if (ffmpegaudenc->opened) {
- gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
- ffmpegaudenc->opened = FALSE;
- }
- gst_adapter_clear (ffmpegaudenc->adapter);
- break;
- default:
- break;
- }
- return result;
-}
-
gboolean
gst_ffmpegaudenc_register (GstPlugin * plugin)
{
if (!type) {
/* create the glib type now */
- type = g_type_register_static (GST_TYPE_ELEMENT, type_name, &typeinfo, 0);
+ type =
+ g_type_register_static (GST_TYPE_AUDIO_ENCODER, type_name, &typeinfo,
+ 0);
g_type_set_qdata (type, GST_FFENC_PARAMS_QDATA, (gpointer) in_plugin);
{
projects / platform / upstream / gst-libav.git / commitdiff