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gstreamer/sys/androidmedia/gstamcaudiodec.c
Matthew Waters 640a65bf96 gst: don't use volatile to mean atomic 
volatile is not sufficient to provide atomic guarantees and real atomics
should be used instead.  GCC 11 has started warning about using volatile
with atomic operations.

https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719

Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2098>
2021-03-22 14:34:36 +11:00

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/*
* Initially based on gst-omx/omx/gstomxvideodec.c
*
* Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
*
* Copyright (C) 2012, Collabora Ltd.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* Copyright (C) 2015, Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation
* version 2.1 of the License.
*
* 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
#ifdef HAVE_ORC
#include <orc/orc.h>
#else
#define orc_memcpy memcpy
#endif
#include "gstamcaudiodec.h"
#include "gstamc-constants.h"
GST_DEBUG_CATEGORY_STATIC (gst_amc_audio_dec_debug_category);
#define GST_CAT_DEFAULT gst_amc_audio_dec_debug_category
#define GST_AUDIO_DECODER_ERROR_FROM_ERROR(el, err) G_STMT_START { \
gchar *__dbg = g_strdup (err->message); \
GstAudioDecoder *__dec = GST_AUDIO_DECODER (el); \
GST_WARNING_OBJECT (el, "error: %s", __dbg); \
_gst_audio_decoder_error (__dec, 1, \
err->domain, err->code, \
NULL, __dbg, __FILE__, GST_FUNCTION, __LINE__); \
g_clear_error (&err); \
} G_STMT_END
/* prototypes */
static void gst_amc_audio_dec_finalize (GObject * object);
static GstStateChangeReturn
gst_amc_audio_dec_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_amc_audio_dec_open (GstAudioDecoder * decoder);
static gboolean gst_amc_audio_dec_close (GstAudioDecoder * decoder);
static gboolean gst_amc_audio_dec_start (GstAudioDecoder * decoder);
static gboolean gst_amc_audio_dec_stop (GstAudioDecoder * decoder);
static gboolean gst_amc_audio_dec_set_format (GstAudioDecoder * decoder,
GstCaps * caps);
static void gst_amc_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard);
static GstFlowReturn gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder,
GstBuffer * buffer);
static GstFlowReturn gst_amc_audio_dec_drain (GstAmcAudioDec * self);
enum
{
PROP_0
};
/* class initialization */
static void gst_amc_audio_dec_class_init (GstAmcAudioDecClass * klass);
static void gst_amc_audio_dec_init (GstAmcAudioDec * self);
static void gst_amc_audio_dec_base_init (gpointer g_class);
static GstAudioDecoderClass *parent_class = NULL;
GType
gst_amc_audio_dec_get_type (void)
{
static gsize type = 0;
if (g_once_init_enter (&type)) {
GType _type;
static const GTypeInfo info = {
sizeof (GstAmcAudioDecClass),
gst_amc_audio_dec_base_init,
NULL,
(GClassInitFunc) gst_amc_audio_dec_class_init,
NULL,
NULL,
sizeof (GstAmcAudioDec),
0,
(GInstanceInitFunc) gst_amc_audio_dec_init,
NULL
};
_type = g_type_register_static (GST_TYPE_AUDIO_DECODER, "GstAmcAudioDec",
&info, 0);
GST_DEBUG_CATEGORY_INIT (gst_amc_audio_dec_debug_category, "amcaudiodec", 0,
"Android MediaCodec audio decoder");
g_once_init_leave (&type, _type);
}
return type;
}
static const gchar *
caps_to_mime (GstCaps * caps)
{
GstStructure *s;
const gchar *name;
s = gst_caps_get_structure (caps, 0);
if (!s)
return NULL;
name = gst_structure_get_name (s);
if (strcmp (name, "audio/mpeg") == 0) {
gint mpegversion;
if (!gst_structure_get_int (s, "mpegversion", &mpegversion))
return NULL;
if (mpegversion == 1) {
gint layer;
if (!gst_structure_get_int (s, "layer", &layer) || layer == 3)
return "audio/mpeg";
else if (layer == 2)
return "audio/mpeg-L2";
} else if (mpegversion == 2 || mpegversion == 4) {
return "audio/mp4a-latm";
}
} else if (strcmp (name, "audio/AMR") == 0) {
return "audio/3gpp";
} else if (strcmp (name, "audio/AMR-WB") == 0) {
return "audio/amr-wb";
} else if (strcmp (name, "audio/x-alaw") == 0) {
return "audio/g711-alaw";
} else if (strcmp (name, "audio/x-mulaw") == 0) {
return "audio/g711-mlaw";
} else if (strcmp (name, "audio/x-vorbis") == 0) {
return "audio/vorbis";
} else if (strcmp (name, "audio/x-opus") == 0) {
return "audio/opus";
}
return NULL;
}
static void
gst_amc_audio_dec_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstAmcAudioDecClass *amcaudiodec_class = GST_AMC_AUDIO_DEC_CLASS (g_class);
const GstAmcCodecInfo *codec_info;
GstPadTemplate *templ;
GstCaps *sink_caps, *src_caps;
gchar *longname;
codec_info =
g_type_get_qdata (G_TYPE_FROM_CLASS (g_class), gst_amc_codec_info_quark);
/* This happens for the base class and abstract subclasses */
if (!codec_info)
return;
amcaudiodec_class->codec_info = codec_info;
gst_amc_codec_info_to_caps (codec_info, &sink_caps, &src_caps);
/* Add pad templates */
templ =
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps);
gst_element_class_add_pad_template (element_class, templ);
gst_caps_unref (sink_caps);
templ = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, src_caps);
gst_element_class_add_pad_template (element_class, templ);
gst_caps_unref (src_caps);
longname = g_strdup_printf ("Android MediaCodec %s", codec_info->name);
gst_element_class_set_metadata (element_class,
codec_info->name,
"Codec/Decoder/Audio",
longname, "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
g_free (longname);
}
static void
gst_amc_audio_dec_class_init (GstAmcAudioDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *audiodec_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_amc_audio_dec_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_amc_audio_dec_change_state);
audiodec_class->start = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_start);
audiodec_class->stop = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_stop);
audiodec_class->open = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_open);
audiodec_class->close = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_close);
audiodec_class->flush = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_flush);
audiodec_class->set_format = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_set_format);
audiodec_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_amc_audio_dec_handle_frame);
}
static void
gst_amc_audio_dec_init (GstAmcAudioDec * self)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE);
gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
g_mutex_init (&self->drain_lock);
g_cond_init (&self->drain_cond);
self->output_adapter = gst_adapter_new ();
}
static gboolean
gst_amc_audio_dec_open (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (decoder);
GstAmcAudioDecClass *klass = GST_AMC_AUDIO_DEC_GET_CLASS (self);
GError *err = NULL;
GST_DEBUG_OBJECT (self, "Opening decoder");
self->codec = gst_amc_codec_new (klass->codec_info->name, FALSE, &err);
if (!self->codec) {
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
self->started = FALSE;
self->flushing = TRUE;
GST_DEBUG_OBJECT (self, "Opened decoder");
return TRUE;
}
static gboolean
gst_amc_audio_dec_close (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Closing decoder");
if (self->codec) {
GError *err = NULL;
gst_amc_codec_release (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_amc_codec_free (self->codec);
}
self->codec = NULL;
self->started = FALSE;
self->flushing = TRUE;
GST_DEBUG_OBJECT (self, "Closed decoder");
return TRUE;
}
static void
gst_amc_audio_dec_finalize (GObject * object)
{
GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (object);
if (self->output_adapter)
gst_object_unref (self->output_adapter);
self->output_adapter = NULL;
g_mutex_clear (&self->drain_lock);
g_cond_clear (&self->drain_cond);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstStateChangeReturn
gst_amc_audio_dec_change_state (GstElement * element, GstStateChange transition)
{
GstAmcAudioDec *self;
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GError *err = NULL;
g_return_val_if_fail (GST_IS_AMC_AUDIO_DEC (element),
GST_STATE_CHANGE_FAILURE);
self = GST_AMC_AUDIO_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
self->downstream_flow_ret = GST_FLOW_OK;
self->draining = FALSE;
self->started = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
self->flushing = TRUE;
gst_amc_codec_flush (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
break;
default:
break;
}
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static gboolean
gst_amc_audio_dec_set_src_caps (GstAmcAudioDec * self, GstAmcFormat * format)
{
gint rate, channels;
guint32 channel_mask = 0;
GstAudioChannelPosition to[64];
GError *err = NULL;
if (!gst_amc_format_get_int (format, "sample-rate", &rate, &err) ||
!gst_amc_format_get_int (format, "channel-count", &channels, &err)) {
GST_ERROR_OBJECT (self, "Failed to get output format metadata: %s",
err->message);
g_clear_error (&err);
return FALSE;
}
if (rate == 0 || channels == 0) {
GST_ERROR_OBJECT (self, "Rate or channels not set");
return FALSE;
}
/* Not always present */
gst_amc_format_get_int (format, "channel-mask", (gint *) & channel_mask,
NULL);
gst_amc_audio_channel_mask_to_positions (channel_mask, channels,
self->positions);
memcpy (to, self->positions, sizeof (to));
gst_audio_channel_positions_to_valid_order (to, channels);
self->needs_reorder =
(memcmp (self->positions, to,
sizeof (GstAudioChannelPosition) * channels) != 0);
if (self->needs_reorder)
gst_audio_get_channel_reorder_map (channels, self->positions, to,
self->reorder_map);
gst_audio_info_init (&self->info);
gst_audio_info_set_format (&self->info, GST_AUDIO_FORMAT_S16, rate, channels,
to);
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self),
&self->info))
return FALSE;
self->input_caps_changed = FALSE;
return TRUE;
}
static void
gst_amc_audio_dec_loop (GstAmcAudioDec * self)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
gboolean is_eos;
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
gint idx;
GError *err = NULL;
GST_AUDIO_DECODER_STREAM_LOCK (self);
retry:
/*if (self->input_caps_changed) {
idx = INFO_OUTPUT_FORMAT_CHANGED;
} else { */
GST_DEBUG_OBJECT (self, "Waiting for available output buffer");
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx =
gst_amc_codec_dequeue_output_buffer (self->codec, &buffer_info, 100000,
&err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
/*} */
if (idx < 0) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
switch (idx) {
case INFO_OUTPUT_BUFFERS_CHANGED:
/* Handled internally */
g_assert_not_reached ();
break;
case INFO_OUTPUT_FORMAT_CHANGED:{
GstAmcFormat *format;
gchar *format_string;
GST_DEBUG_OBJECT (self, "Output format has changed");
format = gst_amc_codec_get_output_format (self->codec, &err);
if (!format)
goto format_error;
format_string = gst_amc_format_to_string (format, &err);
if (err) {
gst_amc_format_free (format);
goto format_error;
}
GST_DEBUG_OBJECT (self, "Got new output format: %s", format_string);
g_free (format_string);
if (!gst_amc_audio_dec_set_src_caps (self, format)) {
gst_amc_format_free (format);
goto format_error;
}
gst_amc_format_free (format);
goto retry;
}
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing output buffer timed out");
goto retry;
case G_MININT:
GST_ERROR_OBJECT (self, "Failure dequeueing output buffer");
goto dequeue_error;
default:
g_assert_not_reached ();
break;
}
goto retry;
}
GST_DEBUG_OBJECT (self,
"Got output buffer at index %d: offset %d size %d time %" G_GINT64_FORMAT
" flags 0x%08x", idx, buffer_info.offset, buffer_info.size,
buffer_info.presentation_time_us, buffer_info.flags);
is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM);
buf = gst_amc_codec_get_output_buffer (self->codec, idx, &err);
if (err) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto failed_to_get_output_buffer;
} else if (!buf) {
goto got_null_output_buffer;
}
if (buffer_info.size > 0) {
GstBuffer *outbuf;
GstMapInfo minfo;
/* This sometimes happens at EOS or if the input is not properly framed,
* let's handle it gracefully by allocating a new buffer for the current
* caps and filling it
*/
if (buffer_info.size % self->info.bpf != 0)
goto invalid_buffer_size;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
buffer_info.size);
if (!outbuf)
goto failed_allocate;
gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE);
if (self->needs_reorder) {
gint i, n_samples, c, n_channels;
gint *reorder_map = self->reorder_map;
gint16 *dest, *source;
dest = (gint16 *) minfo.data;
source = (gint16 *) (buf->data + buffer_info.offset);
n_samples = buffer_info.size / self->info.bpf;
n_channels = self->info.channels;
for (i = 0; i < n_samples; i++) {
for (c = 0; c < n_channels; c++) {
dest[i * n_channels + reorder_map[c]] = source[i * n_channels + c];
}
}
} else {
orc_memcpy (minfo.data, buf->data + buffer_info.offset, buffer_info.size);
}
gst_buffer_unmap (outbuf, &minfo);
if (self->spf != -1) {
gst_adapter_push (self->output_adapter, outbuf);
} else {
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1);
}
}
gst_amc_buffer_free (buf);
buf = NULL;
if (self->spf != -1) {
GstBuffer *outbuf;
guint avail = gst_adapter_available (self->output_adapter);
guint nframes;
/* On EOS we take the complete adapter content, no matter
* if it is a multiple of the codec frame size or not.
* Otherwise we take a multiple of codec frames and push
* them downstream
*/
avail /= self->info.bpf;
if (!is_eos) {
nframes = avail / self->spf;
avail = nframes * self->spf;
} else {
nframes = (avail + self->spf - 1) / self->spf;
}
avail *= self->info.bpf;
if (avail > 0) {
outbuf = gst_adapter_take_buffer (self->output_adapter, avail);
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
nframes);
}
}
if (!gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err)) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto failed_release;
}
if (is_eos || flow_ret == GST_FLOW_EOS) {
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
} else if (flow_ret == GST_FLOW_OK) {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
} else {
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
}
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
dequeue_error:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
format_error:
{
if (err)
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
else
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Failed to handle format"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_release:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err);
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_FLUSHING;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
} else if (flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_FLOW_ERROR (self, flow_ret);
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
} else if (flow_ret == GST_FLOW_FLUSHING) {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_to_get_output_buffer:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err);
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
got_null_output_buffer:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Got no output buffer"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
invalid_buffer_size:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Invalid buffer size %u (bfp %d)", buffer_info.size, self->info.bpf));
gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&err);
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
failed_allocate:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to allocate output buffer"));
gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&err);
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
return;
}
}
static gboolean
gst_amc_audio_dec_start (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self;
self = GST_AMC_AUDIO_DEC (decoder);
self->last_upstream_ts = 0;
self->drained = TRUE;
self->downstream_flow_ret = GST_FLOW_OK;
self->started = FALSE;
self->flushing = TRUE;
return TRUE;
}
static gboolean
gst_amc_audio_dec_stop (GstAudioDecoder * decoder)
{
GstAmcAudioDec *self;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Stopping decoder");
self->flushing = TRUE;
if (self->started) {
gst_amc_codec_flush (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_amc_codec_stop (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
self->started = FALSE;
}
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
memset (self->positions, 0, sizeof (self->positions));
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
g_list_foreach (self->codec_datas, (GFunc) g_free, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->drained = TRUE;
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
GST_DEBUG_OBJECT (self, "Stopped decoder");
return TRUE;
}
static gboolean
gst_amc_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstAmcAudioDec *self;
GstStructure *s;
GstAmcFormat *format;
const gchar *mime;
gboolean is_format_change = FALSE;
gboolean needs_disable = FALSE;
gchar *format_string;
gint rate, channels;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps);
/* Check if the caps change is a real format change or if only irrelevant
* parts of the caps have changed or nothing at all.
*/
is_format_change |= (!self->input_caps
|| !gst_caps_is_equal (self->input_caps, caps));
needs_disable = self->started;
/* If the component is not started and a real format change happens
* we have to restart the component. If no real format change
* happened we can just exit here.
*/
if (needs_disable && !is_format_change) {
/* Framerate or something minor changed */
self->input_caps_changed = TRUE;
GST_DEBUG_OBJECT (self,
"Already running and caps did not change the format");
return TRUE;
}
if (needs_disable && is_format_change) {
gst_amc_audio_dec_drain (self);
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_amc_audio_dec_stop (GST_AUDIO_DECODER (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_amc_audio_dec_close (GST_AUDIO_DECODER (self));
if (!gst_amc_audio_dec_open (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to open codec again");
return FALSE;
}
if (!gst_amc_audio_dec_start (GST_AUDIO_DECODER (self))) {
GST_ERROR_OBJECT (self, "Failed to start codec again");
}
}
/* srcpad task is not running at this point */
mime = caps_to_mime (caps);
if (!mime) {
GST_ERROR_OBJECT (self, "Failed to convert caps to mime");
return FALSE;
}
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "rate", &rate) ||
!gst_structure_get_int (s, "channels", &channels)) {
GST_ERROR_OBJECT (self, "Failed to get rate/channels");
return FALSE;
}
format = gst_amc_format_new_audio (mime, rate, channels, &err);
if (!format) {
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion;
const gchar *stream_format;
if (!gst_structure_get_int (s, "mpegversion", &mpegversion))
mpegversion = -1;
stream_format = gst_structure_get_string (s, "stream-format");
if (mpegversion == 4 && g_strcmp0 (stream_format, "adts") == 0) {
gst_amc_format_set_int (format, "is-adts", 1, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
}
}
/* FIXME: These buffers needs to be valid until the codec is stopped again */
g_list_foreach (self->codec_datas, (GFunc) gst_buffer_unref, NULL);
g_list_free (self->codec_datas);
self->codec_datas = NULL;
if (gst_structure_has_field (s, "codec_data")) {
const GValue *h = gst_structure_get_value (s, "codec_data");
GstBuffer *codec_data = gst_value_get_buffer (h);
GstMapInfo minfo;
guint8 *data;
gst_buffer_map (codec_data, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, "csd-0", data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (codec_data, &minfo);
} else if (gst_structure_has_field (s, "streamheader")) {
const GValue *sh = gst_structure_get_value (s, "streamheader");
gint nsheaders = gst_value_array_get_size (sh);
GstBuffer *buf;
const GValue *h;
gint i, j;
gchar *fname;
GstMapInfo minfo;
guint8 *data;
for (i = 0, j = 0; i < nsheaders; i++) {
h = gst_value_array_get_value (sh, i);
buf = gst_value_get_buffer (h);
if (strcmp (mime, "audio/vorbis") == 0) {
guint8 header_type;
gst_buffer_extract (buf, 0, &header_type, 1);
/* Only use the identification and setup packets */
if (header_type != 0x01 && header_type != 0x05)
continue;
}
fname = g_strdup_printf ("csd-%d", j);
gst_buffer_map (buf, &minfo, GST_MAP_READ);
data = g_memdup (minfo.data, minfo.size);
self->codec_datas = g_list_prepend (self->codec_datas, data);
gst_amc_format_set_buffer (format, fname, data, minfo.size, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_buffer_unmap (buf, &minfo);
g_free (fname);
j++;
}
}
format_string = gst_amc_format_to_string (format, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
GST_DEBUG_OBJECT (self, "Configuring codec with format: %s",
GST_STR_NULL (format_string));
g_free (format_string);
if (!gst_amc_codec_configure (self->codec, format, NULL, &err)) {
GST_ERROR_OBJECT (self, "Failed to configure codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
gst_amc_format_free (format);
if (!gst_amc_codec_start (self->codec, &err)) {
GST_ERROR_OBJECT (self, "Failed to start codec");
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
return FALSE;
}
self->spf = -1;
/* TODO: Implement for other codecs too */
if (gst_structure_has_name (s, "audio/mpeg")) {
gint mpegversion = -1;
gst_structure_get_int (s, "mpegversion", &mpegversion);
if (mpegversion == 1) {
gint layer = -1, mpegaudioversion = -1;
gst_structure_get_int (s, "layer", &layer);
gst_structure_get_int (s, "mpegaudioversion", &mpegaudioversion);
if (layer == 1)
self->spf = 384;
else if (layer == 2)
self->spf = 1152;
else if (layer == 3 && mpegaudioversion != -1)
self->spf = (mpegaudioversion == 1 ? 1152 : 576);
}
}
self->started = TRUE;
self->input_caps_changed = TRUE;
/* Start the srcpad loop again */
self->flushing = FALSE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
return TRUE;
}
static void
gst_amc_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstAmcAudioDec *self;
GError *err = NULL;
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Resetting decoder");
if (!self->started) {
GST_DEBUG_OBJECT (self, "Codec not started yet");
return;
}
self->flushing = TRUE;
/* Wait until the srcpad loop is finished,
* unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
GST_PAD_STREAM_LOCK (GST_AUDIO_DECODER_SRC_PAD (self));
GST_PAD_STREAM_UNLOCK (GST_AUDIO_DECODER_SRC_PAD (self));
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_amc_codec_flush (self->codec, &err);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
self->flushing = FALSE;
/* Start the srcpad loop again */
self->last_upstream_ts = 0;
self->drained = TRUE;
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);
GST_DEBUG_OBJECT (self, "Reset decoder");
}
static GstFlowReturn
gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
{
GstAmcAudioDec *self;
gint idx;
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
guint offset = 0;
GstClockTime timestamp, duration, timestamp_offset = 0;
GstMapInfo minfo;
GError *err = NULL;
memset (&minfo, 0, sizeof (minfo));
self = GST_AMC_AUDIO_DEC (decoder);
GST_DEBUG_OBJECT (self, "Handling frame");
/* Make sure to keep a reference to the input here,
* it can be unreffed from the other thread if
* finish_frame() is called */
if (inbuf)
inbuf = gst_buffer_ref (inbuf);
if (!self->started) {
GST_ERROR_OBJECT (self, "Codec not started yet");
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_NOT_NEGOTIATED;
}
if (self->flushing)
goto flushing;
if (self->downstream_flow_ret != GST_FLOW_OK)
goto downstream_error;
if (!inbuf)
return gst_amc_audio_dec_drain (self);
timestamp = GST_BUFFER_PTS (inbuf);
duration = GST_BUFFER_DURATION (inbuf);
gst_buffer_map (inbuf, &minfo, GST_MAP_READ);
while (offset < minfo.size) {
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Wait at most 100ms here, some codecs don't fail dequeueing if
* the codec is flushing, causing deadlocks during shutdown */
idx = gst_amc_codec_dequeue_input_buffer (self->codec, 100000, &err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (idx < 0) {
if (self->flushing || self->downstream_flow_ret == GST_FLOW_FLUSHING) {
g_clear_error (&err);
goto flushing;
}
switch (idx) {
case INFO_TRY_AGAIN_LATER:
GST_DEBUG_OBJECT (self, "Dequeueing input buffer timed out");
continue; /* next try */
break;
case G_MININT:
GST_ERROR_OBJECT (self, "Failed to dequeue input buffer");
goto dequeue_error;
default:
g_assert_not_reached ();
break;
}
continue;
}
if (self->flushing) {
memset (&buffer_info, 0, sizeof (buffer_info));
gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, NULL);
goto flushing;
}
if (self->downstream_flow_ret != GST_FLOW_OK) {
memset (&buffer_info, 0, sizeof (buffer_info));
gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, &err);
if (err && !self->flushing)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
g_clear_error (&err);
goto downstream_error;
}
/* Now handle the frame */
/* Copy the buffer content in chunks of size as requested
* by the port */
buf = gst_amc_codec_get_input_buffer (self->codec, idx, &err);
if (err)
goto failed_to_get_input_buffer;
else if (!buf)
goto got_null_input_buffer;
memset (&buffer_info, 0, sizeof (buffer_info));
buffer_info.offset = 0;
buffer_info.size = MIN (minfo.size - offset, buf->size);
gst_amc_buffer_set_position_and_limit (buf, NULL, buffer_info.offset,
buffer_info.size);
orc_memcpy (buf->data, minfo.data + offset, buffer_info.size);
gst_amc_buffer_free (buf);
buf = NULL;
/* Interpolate timestamps if we're passing the buffer
* in multiple chunks */
if (offset != 0 && duration != GST_CLOCK_TIME_NONE) {
timestamp_offset = gst_util_uint64_scale (offset, duration, minfo.size);
}
if (timestamp != GST_CLOCK_TIME_NONE) {
buffer_info.presentation_time_us =
gst_util_uint64_scale (timestamp + timestamp_offset, 1, GST_USECOND);
self->last_upstream_ts = timestamp + timestamp_offset;
}
if (duration != GST_CLOCK_TIME_NONE)
self->last_upstream_ts += duration;
if (offset == 0) {
if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DELTA_UNIT))
buffer_info.flags |= BUFFER_FLAG_SYNC_FRAME;
}
offset += buffer_info.size;
GST_DEBUG_OBJECT (self,
"Queueing buffer %d: size %d time %" G_GINT64_FORMAT " flags 0x%08x",
idx, buffer_info.size, buffer_info.presentation_time_us,
buffer_info.flags);
if (!gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info,
&err)) {
if (self->flushing) {
g_clear_error (&err);
goto flushing;
}
goto queue_error;
}
self->drained = FALSE;
}
gst_buffer_unmap (inbuf, &minfo);
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
downstream_error:
{
GST_ERROR_OBJECT (self, "Downstream returned %s",
gst_flow_get_name (self->downstream_flow_ret));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return self->downstream_flow_ret;
}
failed_to_get_input_buffer:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
got_null_input_buffer:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Got no input buffer"));
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
dequeue_error:
{
GST_ELEMENT_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
queue_error:
{
GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err);
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
if (minfo.data)
gst_buffer_unmap (inbuf, &minfo);
if (inbuf)
gst_buffer_unref (inbuf);
return GST_FLOW_FLUSHING;
}
}
static GstFlowReturn
gst_amc_audio_dec_drain (GstAmcAudioDec * self)
{
GstFlowReturn ret;
gint idx;
GError *err = NULL;
GST_DEBUG_OBJECT (self, "Draining codec");
if (!self->started) {
GST_DEBUG_OBJECT (self, "Codec not started yet");
return GST_FLOW_OK;
}
/* Don't send drain buffer twice, this doesn't work */
if (self->drained) {
GST_DEBUG_OBJECT (self, "Codec is drained already");
return GST_FLOW_OK;
}
/* Make sure to release the base class stream lock, otherwise
* _loop() can't call _finish_frame() and we might block forever
* because no input buffers are released */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
/* Send an EOS buffer to the component and let the base
* class drop the EOS event. We will send it later when
* the EOS buffer arrives on the output port.
* Wait at most 0.5s here. */
idx = gst_amc_codec_dequeue_input_buffer (self->codec, 500000, &err);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (idx >= 0) {
GstAmcBuffer *buf;
GstAmcBufferInfo buffer_info;
buf = gst_amc_codec_get_input_buffer (self->codec, idx, &err);
if (buf) {
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
g_mutex_lock (&self->drain_lock);
self->draining = TRUE;
memset (&buffer_info, 0, sizeof (buffer_info));
buffer_info.size = 0;
buffer_info.presentation_time_us =
gst_util_uint64_scale (self->last_upstream_ts, 1, GST_USECOND);
buffer_info.flags |= BUFFER_FLAG_END_OF_STREAM;
gst_amc_buffer_set_position_and_limit (buf, NULL, 0, 0);
gst_amc_buffer_free (buf);
buf = NULL;
if (gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info,
&err)) {
GST_DEBUG_OBJECT (self, "Waiting until codec is drained");
g_cond_wait (&self->drain_cond, &self->drain_lock);
GST_DEBUG_OBJECT (self, "Drained codec");
ret = GST_FLOW_OK;
} else {
GST_ERROR_OBJECT (self, "Failed to queue input buffer");
if (self->flushing) {
g_clear_error (&err);
ret = GST_FLOW_FLUSHING;
} else {
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
ret = GST_FLOW_ERROR;
}
}
self->drained = TRUE;
self->draining = FALSE;
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
} else {
GST_ERROR_OBJECT (self, "Failed to get buffer for EOS: %d", idx);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
ret = GST_FLOW_ERROR;
}
} else {
GST_ERROR_OBJECT (self, "Failed to acquire buffer for EOS: %d", idx);
if (err)
GST_ELEMENT_WARNING_FROM_ERROR (self, err);
ret = GST_FLOW_ERROR;
}
gst_adapter_flush (self->output_adapter,
gst_adapter_available (self->output_adapter));
return ret;
}
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