gstreamer/subprojects/gst-libav/ext/libav/gstavauddec.c
Edward Hervey f7337b7477 ffmpeg: Don't register elements for which we have no caps
We would previously register a whole bunch of encoder/decoder for which the caps
were ... "unknown/unknown".

Add a function to quickly check (without generating caps) whether a given
AVCodecID has a known mapping (which can include the {video|audio}/x-gst-av-*
ones) without generating the caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6237>
2024-07-25 12:14:50 +00:00

959 lines
30 KiB
C

/* 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
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <assert.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <gst/gst.h>
#include <gst/base/gstbytewriter.h>
#include "gstav.h"
#include "gstavcodecmap.h"
#include "gstavutils.h"
#include "gstavauddec.h"
GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);
/* 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_propose_allocation (GstAudioDecoder * decoder,
GstQuery * query);
static gboolean gst_ffmpegauddec_start (GstAudioDecoder * decoder);
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 GstFlowReturn gst_ffmpegauddec_handle_frame (GstAudioDecoder * decoder,
GstBuffer * inbuf);
static gboolean gst_ffmpegauddec_negotiate (GstFFMpegAudDec * ffmpegdec,
AVCodecContext * context, AVFrame * frame, gboolean force);
static GstFlowReturn gst_ffmpegauddec_drain (GstFFMpegAudDec * ffmpegdec,
gboolean force);
#define GST_FFDEC_PARAMS_QDATA g_quark_from_static_string("avdec-params")
static GstElementClass *parent_class = NULL;
static void
gst_ffmpegauddec_base_init (GstFFMpegAudDecClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstPadTemplate *sinktempl, *srctempl;
GstCaps *sinkcaps, *srccaps;
AVCodec *in_plugin;
gchar *longname, *description;
in_plugin =
(AVCodec *) g_type_get_qdata (G_OBJECT_CLASS_TYPE (klass),
GST_FFDEC_PARAMS_QDATA);
g_assert (in_plugin != NULL);
/* construct the element details struct */
longname = g_strdup_printf ("libav %s decoder", in_plugin->long_name);
description = g_strdup_printf ("libav %s decoder", in_plugin->name);
gst_element_class_set_metadata (element_class, longname,
"Codec/Decoder/Audio", description,
"Wim Taymans <wim.taymans@gmail.com>, "
"Ronald Bultje <rbultje@ronald.bitfreak.net>, "
"Edward Hervey <bilboed@bilboed.com>");
g_free (longname);
g_free (description);
/* get the caps */
sinkcaps = gst_ffmpeg_codecid_to_caps (in_plugin->id, NULL, FALSE);
if (!sinkcaps) {
GST_DEBUG ("Couldn't get sink caps for decoder '%s'", in_plugin->name);
sinkcaps = gst_caps_from_string ("unknown/unknown");
}
srccaps = gst_ffmpeg_codectype_to_audio_caps (NULL,
in_plugin->id, FALSE, in_plugin);
if (!srccaps) {
GST_DEBUG ("Couldn't get source caps for decoder '%s'", in_plugin->name);
srccaps = gst_caps_from_string ("audio/x-raw");
}
/* pad templates */
sinktempl = gst_pad_template_new ("sink", GST_PAD_SINK,
GST_PAD_ALWAYS, sinkcaps);
srctempl = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, srccaps);
gst_element_class_add_pad_template (element_class, srctempl);
gst_element_class_add_pad_template (element_class, sinktempl);
gst_caps_unref (sinkcaps);
gst_caps_unref (srccaps);
klass->in_plugin = in_plugin;
klass->srctempl = srctempl;
klass->sinktempl = sinktempl;
}
static void
gst_ffmpegauddec_class_init (GstFFMpegAudDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstAudioDecoderClass *gstaudiodecoder_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_ffmpegauddec_finalize;
gstaudiodecoder_class->start = GST_DEBUG_FUNCPTR (gst_ffmpegauddec_start);
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);
gstaudiodecoder_class->propose_allocation =
GST_DEBUG_FUNCPTR (gst_ffmpegauddec_propose_allocation);
GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE");
}
static void
gst_ffmpegauddec_init (GstFFMpegAudDec * ffmpegdec)
{
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (ffmpegdec));
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(ffmpegdec), TRUE);
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (ffmpegdec), TRUE);
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (ffmpegdec), TRUE);
}
static void
gst_ffmpegauddec_finalize (GObject * object)
{
GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) object;
av_frame_free (&ffmpegdec->frame);
avcodec_free_context (&ffmpegdec->context);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/* With LOCK */
static void
gst_ffmpegauddec_close (GstFFMpegAudDec * ffmpegdec)
{
GST_LOG_OBJECT (ffmpegdec, "closing libav codec");
gst_caps_replace (&ffmpegdec->last_caps, NULL);
avcodec_free_context (&ffmpegdec->context);
}
static gboolean
gst_ffmpegauddec_start (GstAudioDecoder * decoder)
{
GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) decoder;
GST_OBJECT_LOCK (ffmpegdec);
ffmpegdec->frame = av_frame_alloc ();
avcodec_free_context (&ffmpegdec->context);
GST_OBJECT_UNLOCK (ffmpegdec);
return TRUE;
}
static gboolean
gst_ffmpegauddec_stop (GstAudioDecoder * decoder)
{
GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) decoder;
GST_OBJECT_LOCK (ffmpegdec);
av_frame_free (&ffmpegdec->frame);
g_free (ffmpegdec->padded);
gst_ffmpegauddec_close (ffmpegdec);
ffmpegdec->padded = NULL;
ffmpegdec->padded_size = 0;
GST_OBJECT_UNLOCK (ffmpegdec);
gst_audio_info_init (&ffmpegdec->info);
gst_caps_replace (&ffmpegdec->last_caps, NULL);
return TRUE;
}
/* with LOCK */
static gboolean
gst_ffmpegauddec_open (GstFFMpegAudDec * ffmpegdec)
{
GstFFMpegAudDecClass *oclass;
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
if (gst_ffmpeg_avcodec_open (ffmpegdec->context, oclass->in_plugin) < 0)
goto could_not_open;
GST_LOG_OBJECT (ffmpegdec, "Opened libav codec %s, id %d",
oclass->in_plugin->name, oclass->in_plugin->id);
gst_audio_info_init (&ffmpegdec->info);
return TRUE;
/* ERRORS */
could_not_open:
{
gst_ffmpegauddec_close (ffmpegdec);
GST_DEBUG_OBJECT (ffmpegdec, "avdec_%s: Failed to open libav codec",
oclass->in_plugin->name);
return FALSE;
}
}
static gboolean
gst_ffmpegauddec_propose_allocation (GstAudioDecoder * decoder,
GstQuery * query)
{
GstAllocationParams params;
gst_allocation_params_init (&params);
params.flags = GST_MEMORY_FLAG_ZERO_PADDED;
params.align = 15;
params.padding = AV_INPUT_BUFFER_PADDING_SIZE;
/* we would like to have some padding so that we don't have to
* memcpy. We don't suggest an allocator. */
gst_query_add_allocation_param (query, NULL, &params);
return GST_AUDIO_DECODER_CLASS (parent_class)->propose_allocation (decoder,
query);
}
static gboolean
gst_ffmpegauddec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) decoder;
GstFFMpegAudDecClass *oclass;
gboolean ret = TRUE;
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
GST_DEBUG_OBJECT (ffmpegdec, "setcaps called");
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->context) {
GST_OBJECT_UNLOCK (ffmpegdec);
gst_ffmpegauddec_drain (ffmpegdec, FALSE);
GST_OBJECT_LOCK (ffmpegdec);
gst_ffmpegauddec_close (ffmpegdec);
}
ffmpegdec->context = avcodec_alloc_context3 (oclass->in_plugin);
if (ffmpegdec->context == NULL) {
GST_DEBUG_OBJECT (ffmpegdec, "Failed to allocate context");
GST_OBJECT_UNLOCK (ffmpegdec);
return FALSE;
}
ffmpegdec->context->opaque = ffmpegdec;
/* FIXME: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1474 */
if ((oclass->in_plugin->capabilities & AV_CODEC_CAP_DELAY) != 0
&& (oclass->in_plugin->id == AV_CODEC_ID_WMAV1
|| oclass->in_plugin->id == AV_CODEC_ID_WMAV2)) {
ffmpegdec->context->flags2 |= AV_CODEC_FLAG2_SKIP_MANUAL;
}
/* get size and so */
gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id,
oclass->in_plugin->type, caps, ffmpegdec->context);
/* workaround encoder bugs */
ffmpegdec->context->workaround_bugs |= FF_BUG_AUTODETECT;
ffmpegdec->context->err_recognition = 1;
/* open codec - we don't select an output pix_fmt yet,
* simply because we don't know! We only get it
* during playback... */
if (!gst_ffmpegauddec_open (ffmpegdec))
goto open_failed;
done:
GST_OBJECT_UNLOCK (ffmpegdec);
return ret;
/* ERRORS */
open_failed:
{
GST_DEBUG_OBJECT (ffmpegdec, "Failed to open");
ret = FALSE;
goto done;
}
}
static gboolean
settings_changed (GstFFMpegAudDec * ffmpegdec, AVFrame * frame)
{
GstAudioFormat format;
GstAudioLayout layout;
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(57, 28, 100)
const gint channels = frame->ch_layout.nb_channels;
#else
gint channels = av_get_channel_layout_nb_channels (frame->channel_layout);
if (channels == 0)
channels = frame->channels;
#endif
format = gst_ffmpeg_smpfmt_to_audioformat (frame->format, &layout);
if (format == GST_AUDIO_FORMAT_UNKNOWN)
return TRUE;
return !(ffmpegdec->info.rate == frame->sample_rate &&
ffmpegdec->info.channels == channels &&
ffmpegdec->info.finfo->format == format &&
ffmpegdec->info.layout == layout);
}
static gboolean
gst_ffmpegauddec_negotiate (GstFFMpegAudDec * ffmpegdec,
AVCodecContext * context, AVFrame * frame, gboolean force)
{
GstFFMpegAudDecClass *oclass;
GstAudioFormat format;
GstAudioLayout layout;
gint channels;
GstAudioChannelPosition pos[64] = { 0, };
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
format = gst_ffmpeg_smpfmt_to_audioformat (frame->format, &layout);
if (format == GST_AUDIO_FORMAT_UNKNOWN)
goto no_caps;
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(57, 28, 100)
channels = frame->ch_layout.nb_channels;
#else
channels = av_get_channel_layout_nb_channels (frame->channel_layout);
if (channels == 0)
channels = frame->channels;
#endif
if (channels == 0)
goto no_caps;
if (!force && !settings_changed (ffmpegdec, frame))
return TRUE;
GST_DEBUG_OBJECT (ffmpegdec,
"Renegotiating audio from %dHz@%dchannels (%d, interleaved=%d) "
"to %dHz@%dchannels (%d, interleaved=%d)",
ffmpegdec->info.rate, ffmpegdec->info.channels,
ffmpegdec->info.finfo->format,
ffmpegdec->info.layout == GST_AUDIO_LAYOUT_INTERLEAVED,
frame->sample_rate, channels, format,
layout == GST_AUDIO_LAYOUT_INTERLEAVED);
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(57, 28, 100)
gst_ffmpeg_channel_layout_to_gst (&frame->ch_layout, channels, pos);
#else
gst_ffmpeg_channel_layout_to_gst (frame->channel_layout, channels, pos);
#endif
memcpy (ffmpegdec->ffmpeg_layout, pos,
sizeof (GstAudioChannelPosition) * channels);
/* Get GStreamer channel layout */
gst_audio_channel_positions_to_valid_order (pos, channels);
ffmpegdec->needs_reorder =
memcmp (pos, ffmpegdec->ffmpeg_layout, sizeof (pos[0]) * channels) != 0;
gst_audio_info_set_format (&ffmpegdec->info, format,
frame->sample_rate, channels, pos);
ffmpegdec->info.layout = layout;
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (ffmpegdec),
&ffmpegdec->info))
goto caps_failed;
return TRUE;
/* ERRORS */
no_caps:
{
#ifdef HAVE_LIBAV_UNINSTALLED
/* using internal ffmpeg snapshot */
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION,
("Could not find GStreamer caps mapping for libav codec '%s'.",
oclass->in_plugin->name), (NULL));
#else
/* using external ffmpeg */
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION,
("Could not find GStreamer caps mapping for libav codec '%s', and "
"you are using an external libavcodec. This is most likely due to "
"a packaging problem and/or libavcodec having been upgraded to a "
"version that is not compatible with this version of "
"gstreamer-libav. Make sure your gstreamer-libav and libavcodec "
"packages come from the same source/repository.",
oclass->in_plugin->name), (NULL));
#endif
return FALSE;
}
caps_failed:
{
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION, (NULL),
("Could not set caps for libav decoder (%s), not fixed?",
oclass->in_plugin->name));
memset (&ffmpegdec->info, 0, sizeof (ffmpegdec->info));
return FALSE;
}
}
static void
gst_avpacket_init (AVPacket * packet, guint8 * data, guint size)
{
memset (packet, 0, sizeof (AVPacket));
packet->data = data;
packet->size = size;
}
/*
* Returns: whether a frame was decoded
*/
static gboolean
gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
AVCodec * in_plugin, GstBuffer ** outbuf, GstFlowReturn * ret,
gboolean * need_more_data)
{
gboolean got_frame = FALSE;
gint res;
res = avcodec_receive_frame (ffmpegdec->context, ffmpegdec->frame);
if (res >= 0) {
gint nsamples, channels, byte_per_sample;
gsize output_size;
gboolean planar;
if (!gst_ffmpegauddec_negotiate (ffmpegdec, ffmpegdec->context,
ffmpegdec->frame, FALSE)) {
*outbuf = NULL;
*ret = GST_FLOW_NOT_NEGOTIATED;
goto beach;
}
got_frame = TRUE;
channels = ffmpegdec->info.channels;
nsamples = ffmpegdec->frame->nb_samples;
byte_per_sample = ffmpegdec->info.finfo->width / 8;
planar = av_sample_fmt_is_planar (ffmpegdec->frame->format);
g_return_val_if_fail (ffmpegdec->info.layout == (planar ?
GST_AUDIO_LAYOUT_NON_INTERLEAVED : GST_AUDIO_LAYOUT_INTERLEAVED),
GST_FLOW_NOT_NEGOTIATED);
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
/* ffmpegdec->frame->linesize[0] might contain padding, allocate only what's needed */
output_size = nsamples * byte_per_sample * channels;
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
if (planar) {
gint i;
GstAudioMeta *meta;
meta = gst_buffer_add_audio_meta (*outbuf, &ffmpegdec->info, nsamples,
NULL);
for (i = 0; i < channels; i++) {
gst_buffer_fill (*outbuf, meta->offsets[i],
ffmpegdec->frame->extended_data[i], nsamples * byte_per_sample);
}
} else {
gst_buffer_fill (*outbuf, 0, ffmpegdec->frame->data[0], output_size);
}
GST_DEBUG_OBJECT (ffmpegdec, "Buffer created. Size: %" G_GSIZE_FORMAT,
output_size);
/* Reorder channels to the GStreamer channel order */
if (ffmpegdec->needs_reorder) {
*outbuf = gst_buffer_make_writable (*outbuf);
gst_audio_buffer_reorder_channels (*outbuf, ffmpegdec->info.finfo->format,
ffmpegdec->info.channels, ffmpegdec->ffmpeg_layout,
ffmpegdec->info.position);
}
/* Mark corrupted frames as corrupted */
if (ffmpegdec->frame->flags & AV_FRAME_FLAG_CORRUPT)
GST_BUFFER_FLAG_SET (*outbuf, GST_BUFFER_FLAG_CORRUPTED);
} else if (res == AVERROR (EAGAIN)) {
GST_DEBUG_OBJECT (ffmpegdec, "Need more data");
*outbuf = NULL;
*need_more_data = TRUE;
} else if (res == AVERROR_EOF) {
*ret = GST_FLOW_EOS;
GST_DEBUG_OBJECT (ffmpegdec, "Context was entirely flushed");
} else if (res < 0) {
GST_AUDIO_DECODER_ERROR (ffmpegdec, 1, STREAM, DECODE, (NULL),
("Audio decoding error"), *ret);
}
beach:
av_frame_unref (ffmpegdec->frame);
GST_DEBUG_OBJECT (ffmpegdec, "return flow %s, out %p, got_frame %d",
gst_flow_get_name (*ret), *outbuf, got_frame);
return got_frame;
}
/*
* Returns: whether a frame was decoded
*/
static gboolean
gst_ffmpegauddec_frame (GstFFMpegAudDec * ffmpegdec, GstFlowReturn * ret,
gboolean * need_more_data)
{
GstFFMpegAudDecClass *oclass;
GstBuffer *outbuf = NULL;
gboolean got_frame = FALSE;
if (G_UNLIKELY (!ffmpegdec->context))
goto no_codec;
*ret = GST_FLOW_OK;
#if LIBAVCODEC_VERSION_MAJOR >= 60
ffmpegdec->context->frame_num++;
#else
ffmpegdec->context->frame_number++;
#endif
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
got_frame =
gst_ffmpegauddec_audio_frame (ffmpegdec, oclass->in_plugin, &outbuf, ret,
need_more_data);
if (outbuf) {
GST_LOG_OBJECT (ffmpegdec, "Decoded data, buffer %" GST_PTR_FORMAT, outbuf);
*ret =
gst_audio_decoder_finish_subframe (GST_AUDIO_DECODER_CAST (ffmpegdec),
outbuf);
} else {
GST_DEBUG_OBJECT (ffmpegdec, "We didn't get a decoded buffer");
}
beach:
return got_frame;
/* ERRORS */
no_codec:
{
GST_ERROR_OBJECT (ffmpegdec, "no codec context");
goto beach;
}
}
static GstFlowReturn
gst_ffmpegauddec_drain (GstFFMpegAudDec * ffmpegdec, gboolean force)
{
GstFlowReturn ret = GST_FLOW_OK;
gboolean got_any_frames = FALSE;
gboolean need_more_data = FALSE;
gboolean got_frame;
if (!ffmpegdec->context)
return GST_FLOW_OK;
if (avcodec_send_packet (ffmpegdec->context, NULL))
goto send_packet_failed;
/* FIXME: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1474 */
if (!(ffmpegdec->context->flags2 & AV_CODEC_FLAG2_SKIP_MANUAL)) {
do {
got_frame = gst_ffmpegauddec_frame (ffmpegdec, &ret, &need_more_data);
if (got_frame)
got_any_frames = TRUE;
} while (got_frame && !need_more_data);
}
avcodec_flush_buffers (ffmpegdec->context);
/* FFMpeg will return AVERROR_EOF if it's internal was fully drained
* then we are translating it to GST_FLOW_EOS. However, because this behavior
* is fully internal stuff of this implementation and gstaudiodecoder
* baseclass doesn't convert this GST_FLOW_EOS to GST_FLOW_OK,
* convert this flow returned here */
if (ret == GST_FLOW_EOS)
ret = GST_FLOW_OK;
if (got_any_frames || force) {
GstFlowReturn new_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (ffmpegdec), NULL, 1);
if (ret == GST_FLOW_OK)
ret = new_ret;
}
done:
return ret;
send_packet_failed:
GST_WARNING_OBJECT (ffmpegdec, "send packet failed, could not drain decoder");
goto done;
}
static void
gst_ffmpegauddec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstFFMpegAudDec *ffmpegdec = (GstFFMpegAudDec *) decoder;
if (ffmpegdec->context) {
avcodec_flush_buffers (ffmpegdec->context);
}
}
static GstFlowReturn
gst_ffmpegauddec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstFFMpegAudDec *ffmpegdec;
GstFFMpegAudDecClass *oclass;
guint8 *data;
GstMapInfo map;
gint size;
gboolean got_any_frames = FALSE;
gboolean got_frame;
GstFlowReturn ret = GST_FLOW_OK;
gboolean is_header;
AVPacket packet;
GstAudioClippingMeta *clipping_meta = NULL;
guint32 num_clipped_samples = 0;
gboolean fully_clipped = FALSE;
gboolean need_more_data = FALSE;
ffmpegdec = (GstFFMpegAudDec *) decoder;
if (G_UNLIKELY (!ffmpegdec->context))
goto not_negotiated;
if (inbuf == NULL) {
return gst_ffmpegauddec_drain (ffmpegdec, FALSE);
}
inbuf = gst_buffer_ref (inbuf);
is_header = GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_HEADER);
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
GST_LOG_OBJECT (ffmpegdec,
"Received new data of size %" G_GSIZE_FORMAT ", offset:%" G_GUINT64_FORMAT
", ts:%" GST_TIME_FORMAT ", dur:%" GST_TIME_FORMAT,
gst_buffer_get_size (inbuf), GST_BUFFER_OFFSET (inbuf),
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.
* libavcodec/svq3.c:svq3_decode_slice_header too.
* ffmpeg devs know about it and will fix it (they said). */
if (oclass->in_plugin->id == AV_CODEC_ID_SVQ1 ||
oclass->in_plugin->id == AV_CODEC_ID_SVQ3) {
inbuf = gst_buffer_make_writable (inbuf);
}
/* mpegaudioparse is setting buffer flags for the Xing/LAME header. This
* should not be passed to the decoder as it results in unnecessary silence
* samples to be output */
if (oclass->in_plugin->id == AV_CODEC_ID_MP3 &&
GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DECODE_ONLY) &&
GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DROPPABLE)) {
gst_buffer_unref (inbuf);
return gst_audio_decoder_finish_frame (decoder, NULL, 1);
}
clipping_meta = gst_buffer_get_audio_clipping_meta (inbuf);
gst_buffer_map (inbuf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
if (size > 0 && (!GST_MEMORY_IS_ZERO_PADDED (map.memory)
|| (map.maxsize - map.size) < AV_INPUT_BUFFER_PADDING_SIZE)) {
/* add padding */
if (ffmpegdec->padded_size < size + AV_INPUT_BUFFER_PADDING_SIZE) {
ffmpegdec->padded_size = size + AV_INPUT_BUFFER_PADDING_SIZE;
ffmpegdec->padded = g_realloc (ffmpegdec->padded, ffmpegdec->padded_size);
GST_LOG_OBJECT (ffmpegdec, "resized padding buffer to %d",
ffmpegdec->padded_size);
}
GST_CAT_TRACE_OBJECT (GST_CAT_PERFORMANCE, ffmpegdec,
"Copy input to add padding");
memcpy (ffmpegdec->padded, data, size);
memset (ffmpegdec->padded + size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
data = ffmpegdec->padded;
}
gst_avpacket_init (&packet, data, size);
if (!packet.size)
goto unmap;
if (clipping_meta != NULL) {
if (clipping_meta->format == GST_FORMAT_DEFAULT) {
uint8_t *p = av_packet_new_side_data (&packet, AV_PKT_DATA_SKIP_SAMPLES,
10);
if (p != NULL) {
GstByteWriter writer;
guint32 start = clipping_meta->start;
guint32 end = clipping_meta->end;
num_clipped_samples = start + end;
gst_byte_writer_init_with_data (&writer, p, 10, FALSE);
gst_byte_writer_put_uint32_le (&writer, start);
gst_byte_writer_put_uint32_le (&writer, end);
GST_LOG_OBJECT (ffmpegdec, "buffer has clipping metadata; added skip "
"side data to avpacket with start %u and end %u", start, end);
}
} else {
GST_WARNING_OBJECT (ffmpegdec,
"buffer has clipping metadata in unsupported format %s",
gst_format_get_name (clipping_meta->format));
}
}
if (avcodec_send_packet (ffmpegdec->context, &packet) < 0) {
av_packet_free_side_data (&packet);
goto send_packet_failed;
}
av_packet_free_side_data (&packet);
do {
/* decode a frame of audio now */
got_frame = gst_ffmpegauddec_frame (ffmpegdec, &ret, &need_more_data);
if (got_frame)
got_any_frames = TRUE;
if (ret != GST_FLOW_OK) {
GST_LOG_OBJECT (ffmpegdec, "breaking because of flow ret %s",
gst_flow_get_name (ret));
/* bad flow return, make sure we discard all data and exit */
break;
}
} while (got_frame && !need_more_data);
/* The frame was fully clipped if we have samples to be clipped and
* it's either more than the known fixed frame size, or the decoder returned
* that it needs more data (EAGAIN) and we didn't decode any frames at all.
*/
fully_clipped = (clipping_meta != NULL && num_clipped_samples > 0)
&& ((ffmpegdec->context->frame_size != 0
&& num_clipped_samples >= ffmpegdec->context->frame_size)
|| (need_more_data && !got_any_frames));
if (is_header || got_any_frames || fully_clipped) {
/* Even if previous return wasn't GST_FLOW_OK, we need to call
* _finish_frame() since baseclass is expecting that _finish_frame()
* is followed by _finish_subframe()
*/
GstFlowReturn new_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (ffmpegdec), NULL, 1);
/* Only override the flow return value if previously did have a GST_FLOW_OK.
* Failure to do this would result in skipping downstream issues caught in
* earlier steps. */
if (ret == GST_FLOW_OK)
ret = new_ret;
}
unmap:
gst_buffer_unmap (inbuf, &map);
gst_buffer_unref (inbuf);
done:
return ret;
/* ERRORS */
not_negotiated:
{
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION, (NULL),
("avdec_%s: input format was not set before data start",
oclass->in_plugin->name));
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
send_packet_failed:
{
GST_AUDIO_DECODER_ERROR (ffmpegdec, 1, STREAM, DECODE, (NULL),
("Audio decoding error"), ret);
if (ret == GST_FLOW_OK) {
/* Even if ffmpeg was not able to decode current audio frame,
* we should call gst_audio_decoder_finish_frame() so that baseclass
* can clear its internal status and can respect timestamp of later
* incoming buffers */
ret = gst_ffmpegauddec_drain (ffmpegdec, TRUE);
}
goto unmap;
}
}
gboolean
gst_ffmpegauddec_register (GstPlugin * plugin)
{
GTypeInfo typeinfo = {
sizeof (GstFFMpegAudDecClass),
(GBaseInitFunc) gst_ffmpegauddec_base_init,
NULL,
(GClassInitFunc) gst_ffmpegauddec_class_init,
NULL,
NULL,
sizeof (GstFFMpegAudDec),
0,
(GInstanceInitFunc) gst_ffmpegauddec_init,
};
GType type;
AVCodec *in_plugin;
void *i = 0;
gint rank;
GST_LOG ("Registering decoders");
while ((in_plugin = (AVCodec *) av_codec_iterate (&i))) {
gchar *type_name;
/* only decoders */
if (!av_codec_is_decoder (in_plugin)
|| in_plugin->type != AVMEDIA_TYPE_AUDIO) {
continue;
}
/* Skip formats we don't handle */
if (!gst_ffmpeg_codecid_is_known (in_plugin->id)) {
continue;
}
/* no quasi codecs, please */
if (in_plugin->id == AV_CODEC_ID_PCM_S16LE_PLANAR ||
(in_plugin->id >= AV_CODEC_ID_PCM_S16LE &&
in_plugin->id <= AV_CODEC_ID_PCM_BLURAY) ||
(in_plugin->id >= AV_CODEC_ID_PCM_S8_PLANAR &&
in_plugin->id <= AV_CODEC_ID_PCM_F24LE))
continue;
/* No decoders depending on external libraries (we don't build them, but
* people who build against an external ffmpeg might have them.
* We have native gstreamer plugins for all of those libraries anyway. */
if (!strncmp (in_plugin->name, "lib", 3)) {
GST_DEBUG
("Not using external library decoder %s. Use the gstreamer-native ones instead.",
in_plugin->name);
continue;
}
GST_DEBUG ("Trying plugin %s [%s]", in_plugin->name, in_plugin->long_name);
/* no codecs for which we're GUARANTEED to have better alternatives */
/* MP1 : Use MP3 for decoding */
/* MP2 : Use MP3 for decoding */
/* Theora: Use libtheora based theoradec */
if (!strcmp (in_plugin->name, "vorbis") ||
!strcmp (in_plugin->name, "wavpack") ||
!strcmp (in_plugin->name, "mp1") ||
!strcmp (in_plugin->name, "mp2") ||
!strcmp (in_plugin->name, "libfaad") ||
!strcmp (in_plugin->name, "mpeg4aac") ||
!strcmp (in_plugin->name, "ass") ||
!strcmp (in_plugin->name, "srt") ||
!strcmp (in_plugin->name, "pgssub") ||
!strcmp (in_plugin->name, "dvdsub") ||
!strcmp (in_plugin->name, "dvbsub")) {
GST_LOG ("Ignoring decoder %s", in_plugin->name);
continue;
}
/* construct the type */
type_name = g_strdup_printf ("avdec_%s", in_plugin->name);
g_strdelimit (type_name, ".,|-<> ", '_');
type = g_type_from_name (type_name);
if (!type) {
/* create the gtype now */
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);
}
/* (Ronald) MPEG-4 gets a higher priority because it has been well-
* tested and by far outperforms divxdec/xviddec - so we prefer it.
* msmpeg4v3 same, as it outperforms divxdec for divx3 playback.
* VC1/WMV3 are not working and thus unpreferred for now. */
switch (in_plugin->id) {
case AV_CODEC_ID_RA_144:
case AV_CODEC_ID_RA_288:
case AV_CODEC_ID_COOK:
case AV_CODEC_ID_AAC:
case AV_CODEC_ID_MUSEPACK7:
case AV_CODEC_ID_MUSEPACK8:
rank = GST_RANK_PRIMARY;
break;
/* SIPR: decoder should have a higher rank than realaudiodec.
*/
case AV_CODEC_ID_SIPR:
rank = GST_RANK_SECONDARY;
break;
default:
rank = GST_RANK_MARGINAL;
break;
}
if (!gst_element_register (plugin, type_name, rank, type)) {
g_warning ("Failed to register %s", type_name);
g_free (type_name);
return FALSE;
}
g_free (type_name);
}
GST_LOG ("Finished Registering decoders");
return TRUE;
}