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avaudenc: Port to audio encoder base class

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This commit is contained in:
Sebastian Dröge 2012-11-20 10:36:29 +01:00
parent 970f40b935
commit 7a29cffc50
4 changed files with 122 additions and 323 deletions
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403
ext/libav/gstavaudenc.c
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@ -1,5 +1,7 @@
/* GStreamer /* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> * 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 * This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public * modify it under the terms of the GNU Library General Public
@ -59,25 +61,19 @@ static void gst_ffmpegaudenc_base_init (GstFFMpegAudEncClass * klass);
static void gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc); static void gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc);
static void gst_ffmpegaudenc_finalize (GObject * object); static void gst_ffmpegaudenc_finalize (GObject * object);
static gboolean gst_ffmpegaudenc_setcaps (GstFFMpegAudEnc * ffmpegenc, static GstCaps *gst_ffmpegaudenc_getcaps (GstAudioEncoder * encoder,
GstCaps * caps);
static GstCaps *gst_ffmpegaudenc_getcaps (GstFFMpegAudEnc * ffmpegenc,
GstCaps * filter); GstCaps * filter);
static GstFlowReturn gst_ffmpegaudenc_chain_audio (GstPad * pad, static gboolean gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder,
GstObject * parent, GstBuffer * buffer); GstAudioInfo * info);
static gboolean gst_ffmpegaudenc_query_sink (GstPad * pad, GstObject * parent, static GstFlowReturn gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder,
GstQuery * query); GstBuffer * inbuf);
static gboolean gst_ffmpegaudenc_event_sink (GstPad * pad, GstObject * parent, static gboolean gst_ffmpegaudenc_stop (GstAudioEncoder * encoder);
GstEvent * event);
static void gst_ffmpegaudenc_set_property (GObject * object, static void gst_ffmpegaudenc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec); guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_ffmpegaudenc_get_property (GObject * object, static void gst_ffmpegaudenc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec); 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") #define GST_FFENC_PARAMS_QDATA g_quark_from_static_string("avenc-params")
static GstElementClass *parent_class = NULL; static GstElementClass *parent_class = NULL;
@ -140,10 +136,10 @@ static void
gst_ffmpegaudenc_class_init (GstFFMpegAudEncClass * klass) gst_ffmpegaudenc_class_init (GstFFMpegAudEncClass * klass)
{ {
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstElementClass *gstelement_class; GstAudioEncoderClass *gstaudioencoder_class;
gobject_class = (GObjectClass *) klass; gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass; gstaudioencoder_class = (GstAudioEncoderClass *) klass;
parent_class = g_type_class_peek_parent (klass); parent_class = g_type_class_peek_parent (klass);
@ -156,37 +152,24 @@ gst_ffmpegaudenc_class_init (GstFFMpegAudEncClass * klass)
"Target Audio Bitrate", 0, G_MAXINT, DEFAULT_AUDIO_BITRATE, "Target Audio Bitrate", 0, G_MAXINT, DEFAULT_AUDIO_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state = gst_ffmpegaudenc_change_state;
gobject_class->finalize = gst_ffmpegaudenc_finalize; 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 static void
gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc) 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 */ /* ffmpeg objects */
ffmpegaudenc->context = avcodec_alloc_context (); ffmpegaudenc->context = avcodec_alloc_context ();
ffmpegaudenc->opened = FALSE; ffmpegaudenc->opened = FALSE;
gst_element_add_pad (GST_ELEMENT (ffmpegaudenc), ffmpegaudenc->sinkpad); gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (ffmpegaudenc), FALSE);
gst_element_add_pad (GST_ELEMENT (ffmpegaudenc), ffmpegaudenc->srcpad);
ffmpegaudenc->adapter = gst_adapter_new ();
} }
static void static void
@ -194,6 +177,16 @@ gst_ffmpegaudenc_finalize (GObject * object)
{ {
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) object; 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 */ /* close old session */
if (ffmpegaudenc->opened) { if (ffmpegaudenc->opened) {
@ -201,30 +194,19 @@ gst_ffmpegaudenc_finalize (GObject * object)
ffmpegaudenc->opened = FALSE; ffmpegaudenc->opened = FALSE;
} }
/* clean up remaining allocated data */ return TRUE;
av_free (ffmpegaudenc->context);
g_object_unref (ffmpegaudenc->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
} }
static GstCaps * static GstCaps *
gst_ffmpegaudenc_getcaps (GstFFMpegAudEnc * ffmpegaudenc, GstCaps * filter) gst_ffmpegaudenc_getcaps (GstAudioEncoder * encoder, GstCaps * filter)
{ {
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
GstCaps *caps = NULL; GstCaps *caps = NULL;
GST_DEBUG_OBJECT (ffmpegaudenc, "getting caps"); GST_DEBUG_OBJECT (ffmpegaudenc, "getting caps");
/* audio needs no special care */ /* audio needs no special care */
caps = gst_pad_get_pad_template_caps (ffmpegaudenc->sinkpad); caps = gst_audio_encoder_proxy_getcaps (encoder, NULL, filter);
if (filter) {
GstCaps *tmp;
tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
GST_DEBUG_OBJECT (ffmpegaudenc, GST_DEBUG_OBJECT (ffmpegaudenc,
"audio caps, return template %" GST_PTR_FORMAT, caps); "audio caps, return template %" GST_PTR_FORMAT, caps);
@ -233,11 +215,13 @@ gst_ffmpegaudenc_getcaps (GstFFMpegAudEnc * ffmpegaudenc, GstCaps * filter)
} }
static gboolean 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 *other_caps;
GstCaps *allowed_caps; GstCaps *allowed_caps;
GstCaps *icaps; GstCaps *icaps;
gsize frame_size;
GstFFMpegAudEncClass *oclass = GstFFMpegAudEncClass *oclass =
(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc); (GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
@ -277,10 +261,8 @@ gst_ffmpegaudenc_setcaps (GstFFMpegAudEnc * ffmpegaudenc, GstCaps * caps)
ffmpegaudenc->context->scenechange_threshold = 0; ffmpegaudenc->context->scenechange_threshold = 0;
ffmpegaudenc->context->inter_threshold = 0; ffmpegaudenc->context->inter_threshold = 0;
/* fetch pix_fmt and so on */ /* fetch pix_fmt and so on */
gst_ffmpeg_caps_with_codectype (oclass->in_plugin->type, gst_ffmpeg_audioinfo_to_context (info, ffmpegaudenc->context);
caps, ffmpegaudenc->context);
if (!ffmpegaudenc->context->time_base.den) { if (!ffmpegaudenc->context->time_base.den) {
ffmpegaudenc->context->time_base.den = 25; ffmpegaudenc->context->time_base.den = 25;
ffmpegaudenc->context->time_base.num = 1; ffmpegaudenc->context->time_base.num = 1;
@ -291,25 +273,20 @@ gst_ffmpegaudenc_setcaps (GstFFMpegAudEnc * ffmpegaudenc, GstCaps * caps)
if (gst_ffmpeg_avcodec_open (ffmpegaudenc->context, oclass->in_plugin) < 0) { if (gst_ffmpeg_avcodec_open (ffmpegaudenc->context, oclass->in_plugin) < 0) {
if (ffmpegaudenc->context->priv_data) if (ffmpegaudenc->context->priv_data)
gst_ffmpeg_avcodec_close (ffmpegaudenc->context); 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", GST_DEBUG_OBJECT (ffmpegaudenc, "avenc_%s: Failed to open FFMPEG codec",
oclass->in_plugin->name); oclass->in_plugin->name);
return FALSE; 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 */ /* some codecs support more than one format, first auto-choose one */
GST_DEBUG_OBJECT (ffmpegaudenc, "picking an output format ..."); 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) { if (!allowed_caps) {
GST_DEBUG_OBJECT (ffmpegaudenc, "... but no peer, using template caps"); GST_DEBUG_OBJECT (ffmpegaudenc, "... but no peer, using template caps");
/* we need to copy because get_allowed_caps returns a ref, and /* we need to copy because get_allowed_caps returns a ref, and
* get_pad_template_caps doesn't */ * 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_DEBUG_OBJECT (ffmpegaudenc, "chose caps %" GST_PTR_FORMAT, allowed_caps);
gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id, gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id,
@ -333,24 +310,31 @@ gst_ffmpegaudenc_setcaps (GstFFMpegAudEnc * ffmpegaudenc, GstCaps * caps)
gst_caps_unref (icaps); gst_caps_unref (icaps);
return FALSE; return FALSE;
} }
icaps = gst_caps_truncate (icaps);
if (gst_caps_get_size (icaps) > 1) { if (!gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (ffmpegaudenc),
GstCaps *newcaps; icaps)) {
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)) {
gst_ffmpeg_avcodec_close (ffmpegaudenc->context); gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
gst_caps_unref (icaps); gst_caps_unref (icaps);
return FALSE; return FALSE;
} }
gst_caps_unref (icaps); 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! */ /* success! */
ffmpegaudenc->opened = TRUE; ffmpegaudenc->opened = TRUE;
@ -360,8 +344,7 @@ gst_ffmpegaudenc_setcaps (GstFFMpegAudEnc * ffmpegaudenc, GstCaps * caps)
static GstFlowReturn static GstFlowReturn
gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc, gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
guint8 * audio_in, guint in_size, guint max_size, GstClockTime timestamp, guint8 * audio_in, guint in_size, guint max_size)
GstClockTime duration, gboolean discont)
{ {
GstBuffer *outbuf; GstBuffer *outbuf;
AVCodecContext *ctx; AVCodecContext *ctx;
@ -372,7 +355,9 @@ gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
ctx = ffmpegaudenc->context; ctx = ffmpegaudenc->context;
/* We need to provide at least ffmpegs minimal buffer size */ /* 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_buffer_map (outbuf, &map, GST_MAP_WRITE);
GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer of max size %d", max_size); GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer of max size %d", max_size);
@ -391,187 +376,55 @@ gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
gst_buffer_unmap (outbuf, &map); gst_buffer_unmap (outbuf, &map);
gst_buffer_resize (outbuf, 0, res); gst_buffer_resize (outbuf, 0, res);
GST_BUFFER_PTS (outbuf) = timestamp; if (res > 0) {
GST_BUFFER_DURATION (outbuf) = duration; GST_LOG_OBJECT (ffmpegaudenc, "pushing size %d", res);
if (discont) ret =
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (ffmpegaudenc),
outbuf, 1);
GST_LOG_OBJECT (ffmpegaudenc, "pushing size %d, timestamp %" GST_TIME_FORMAT, } else {
res, GST_TIME_ARGS (timestamp)); GST_LOG_OBJECT (ffmpegaudenc, "no output produced");
gst_buffer_unref (outbuf);
ret = gst_pad_push (ffmpegaudenc->srcpad, outbuf); ret = GST_FLOW_OK;
}
return ret; return ret;
} }
static GstFlowReturn static GstFlowReturn
gst_ffmpegaudenc_chain_audio (GstPad * pad, GstObject * parent, gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
GstBuffer * inbuf)
{ {
GstFFMpegAudEnc *ffmpegaudenc; GstFFMpegAudEnc *ffmpegaudenc;
GstFFMpegAudEncClass *oclass; gsize size;
AVCodecContext *ctx;
GstClockTime timestamp, duration;
gsize size, frame_size;
gint osize;
GstFlowReturn ret; GstFlowReturn ret;
gint out_size; gint out_size;
gboolean discont;
guint8 *in_data; guint8 *in_data;
GstMapInfo map;
ffmpegaudenc = (GstFFMpegAudEnc *) parent; ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
oclass = (GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
if (G_UNLIKELY (!ffmpegaudenc->opened)) if (G_UNLIKELY (!ffmpegaudenc->opened))
goto not_negotiated; goto not_negotiated;
ctx = ffmpegaudenc->context; inbuf = gst_buffer_ref (inbuf);
size = gst_buffer_get_size (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, GST_DEBUG_OBJECT (ffmpegaudenc,
"Received time %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT "Received time %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT
", size %" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp), ", size %" G_GSIZE_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
GST_TIME_ARGS (duration), size); GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)), size);
frame_size = ctx->frame_size; /* 4 times the input size should be big enough... */
osize = av_get_bits_per_sample_format (ctx->sample_fmt) / 8; out_size = size * 4;
if (frame_size > 1) { gst_buffer_map (inbuf, &map, GST_MAP_READ);
/* we have a frame_size, feed the encoder multiples of this frame size */ in_data = map.data;
guint avail, frame_bytes; size = map.size;
ret = gst_ffmpegaudenc_encode_audio (ffmpegaudenc, in_data, size, out_size);
gst_buffer_unmap (inbuf, &map);
gst_buffer_unref (inbuf);
if (discont) { if (ret != GST_FLOW_OK)
GST_LOG_OBJECT (ffmpegaudenc, "DISCONT, clear adapter"); goto push_failed;
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;
}
}
}
GST_LOG_OBJECT (ffmpegaudenc, "pushing buffer in adapter");
gst_adapter_push (ffmpegaudenc->adapter, 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;
}
return GST_FLOW_OK; return GST_FLOW_OK;
@ -591,55 +444,6 @@ push_failed:
} }
} }
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 static void
gst_ffmpegaudenc_set_property (GObject * object, gst_ffmpegaudenc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec) guint prop_id, const GValue * value, GParamSpec * pspec)
@ -698,33 +502,6 @@ gst_ffmpegaudenc_get_property (GObject * object,
} }
} }
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 gboolean
gst_ffmpegaudenc_register (GstPlugin * plugin) gst_ffmpegaudenc_register (GstPlugin * plugin)
{ {
@ -794,7 +571,9 @@ gst_ffmpegaudenc_register (GstPlugin * plugin)
if (!type) { if (!type) {
/* create the glib type now */ /* 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); g_type_set_qdata (type, GST_FFENC_PARAMS_QDATA, (gpointer) in_plugin);
{ {

14
ext/libav/gstavaudenc.h
View file

@ -27,25 +27,17 @@
G_BEGIN_DECLS G_BEGIN_DECLS
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/base/gstadapter.h> #include <gst/audio/gstaudioencoder.h>
#include <libavcodec/avcodec.h> #include <libavcodec/avcodec.h>
typedef struct _GstFFMpegAudEnc GstFFMpegAudEnc; typedef struct _GstFFMpegAudEnc GstFFMpegAudEnc;
struct _GstFFMpegAudEnc struct _GstFFMpegAudEnc
{ {
GstElement element; GstAudioEncoder parent;
/* We need to keep track of our pads, so we do so here. */
GstPad *srcpad;
GstPad *sinkpad;
AVCodecContext *context; AVCodecContext *context;
gboolean opened; gboolean opened;
GstClockTime adapter_ts;
guint64 adapter_consumed;
GstAdapter *adapter;
gboolean discont;
/* cache */ /* cache */
gint bitrate; gint bitrate;
@ -61,7 +53,7 @@ typedef struct _GstFFMpegAudEncClass GstFFMpegAudEncClass;
struct _GstFFMpegAudEncClass struct _GstFFMpegAudEncClass
{ {
GstElementClass parent_class; GstAudioEncoderClass parent_class;
AVCodec *in_plugin; AVCodec *in_plugin;
GstPadTemplate *srctempl, *sinktempl; GstPadTemplate *srctempl, *sinktempl;

24
ext/libav/gstavcodecmap.c
View file

@ -2241,6 +2241,30 @@ gst_ffmpeg_videoinfo_to_context (GstVideoInfo * info, AVCodecContext * context)
gst_ffmpeg_videoformat_to_pixfmt (GST_VIDEO_INFO_FORMAT (info)); gst_ffmpeg_videoformat_to_pixfmt (GST_VIDEO_INFO_FORMAT (info));
} }
void
gst_ffmpeg_audioinfo_to_context (GstAudioInfo * info, AVCodecContext * context)
{
context->channels = info->channels;
context->sample_rate = info->rate;
switch (info->finfo->format) {
case GST_AUDIO_FORMAT_F32:
context->sample_fmt = AV_SAMPLE_FMT_FLT;
break;
case GST_AUDIO_FORMAT_F64:
context->sample_fmt = AV_SAMPLE_FMT_DBL;
break;
case GST_AUDIO_FORMAT_S32:
context->sample_fmt = AV_SAMPLE_FMT_S32;
break;
case GST_AUDIO_FORMAT_S16:
context->sample_fmt = AV_SAMPLE_FMT_S16;
break;
default:
break;
}
}
/* Convert a GstCaps and a FFMPEG codec Type to a /* Convert a GstCaps and a FFMPEG codec Type to a
* AVCodecContext. If the context is ommitted, no fixed values * AVCodecContext. If the context is ommitted, no fixed values
* for video/audio size will be included in the context * for video/audio size will be included in the context

4
ext/libav/gstavcodecmap.h
View file

@ -91,6 +91,10 @@ void
gst_ffmpeg_videoinfo_to_context (GstVideoInfo *info, gst_ffmpeg_videoinfo_to_context (GstVideoInfo *info,
AVCodecContext *context); AVCodecContext *context);
void
gst_ffmpeg_audioinfo_to_context (GstAudioInfo *info,
AVCodecContext *context);
GstVideoFormat gst_ffmpeg_pixfmt_to_videoformat (enum PixelFormat pixfmt); GstVideoFormat gst_ffmpeg_pixfmt_to_videoformat (enum PixelFormat pixfmt);
enum PixelFormat gst_ffmpeg_videoformat_to_pixfmt (GstVideoFormat format); enum PixelFormat gst_ffmpeg_videoformat_to_pixfmt (GstVideoFormat format);