mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-29 19:50:40 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-omx/omx/gstomxaudioenc.c

1181 lines
37 KiB
C
Raw Blame History

/*
* Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
*
* 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 <string.h>
#include "gstomxaudioenc.h"
GST_DEBUG_CATEGORY_STATIC (gst_omx_audio_enc_debug_category);
#define GST_CAT_DEFAULT gst_omx_audio_enc_debug_category
/* prototypes */
static void gst_omx_audio_enc_finalize (GObject * object);
static GstStateChangeReturn
gst_omx_audio_enc_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_omx_audio_enc_open (GstAudioEncoder * encoder);
static gboolean gst_omx_audio_enc_close (GstAudioEncoder * encoder);
static gboolean gst_omx_audio_enc_start (GstAudioEncoder * encoder);
static gboolean gst_omx_audio_enc_stop (GstAudioEncoder * encoder);
static gboolean gst_omx_audio_enc_set_format (GstAudioEncoder * encoder,
GstAudioInfo * info);
static GstFlowReturn gst_omx_audio_enc_handle_frame (GstAudioEncoder *
encoder, GstBuffer * buffer);
static void gst_omx_audio_enc_flush (GstAudioEncoder * encoder);
static GstFlowReturn gst_omx_audio_enc_drain (GstOMXAudioEnc * self);
enum
{
PROP_0
};
/* class initialization */
#define do_init \
{ \
GST_DEBUG_CATEGORY_INIT (gst_omx_audio_enc_debug_category, "omxaudioenc", 0, \
"debug category for gst-omx audio encoder base class"); \
G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL); \
}
G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstOMXAudioEnc, gst_omx_audio_enc,
GST_TYPE_AUDIO_ENCODER, do_init);
static void
gst_omx_audio_enc_class_init (GstOMXAudioEncClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *audio_encoder_class = GST_AUDIO_ENCODER_CLASS (klass);
gobject_class->finalize = gst_omx_audio_enc_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_omx_audio_enc_change_state);
audio_encoder_class->open = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_open);
audio_encoder_class->close = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_close);
audio_encoder_class->start = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_start);
audio_encoder_class->stop = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_stop);
audio_encoder_class->flush = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_flush);
audio_encoder_class->set_format =
GST_DEBUG_FUNCPTR (gst_omx_audio_enc_set_format);
audio_encoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_omx_audio_enc_handle_frame);
klass->cdata.type = GST_OMX_COMPONENT_TYPE_FILTER;
klass->cdata.default_sink_template_caps = "audio/x-raw, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, " G_STRINGIFY (OMX_AUDIO_MAXCHANNELS) " ], "
"format = (string) { S8, U8, S16LE, S16BE, U16LE, U16BE, "
"S24LE, S24BE, U24LE, U24BE, S32LE, S32BE, U32LE, U32BE }";
}
static void
gst_omx_audio_enc_init (GstOMXAudioEnc * self)
{
g_mutex_init (&self->drain_lock);
g_cond_init (&self->drain_cond);
}
static gboolean
gst_omx_audio_enc_open (GstAudioEncoder * encoder)
{
GstOMXAudioEnc *self = GST_OMX_AUDIO_ENC (encoder);
GstOMXAudioEncClass *klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
gint in_port_index, out_port_index;
self->enc =
gst_omx_component_new (GST_OBJECT_CAST (self), klass->cdata.core_name,
klass->cdata.component_name, klass->cdata.component_role,
klass->cdata.hacks);
self->started = FALSE;
if (!self->enc)
return FALSE;
if (gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateLoaded)
return FALSE;
in_port_index = klass->cdata.in_port_index;
out_port_index = klass->cdata.out_port_index;
if (in_port_index == -1 || out_port_index == -1) {
OMX_PORT_PARAM_TYPE param;
OMX_ERRORTYPE err;
GST_OMX_INIT_STRUCT (&param);
err =
gst_omx_component_get_parameter (self->enc, OMX_IndexParamAudioInit,
&param);
if (err != OMX_ErrorNone) {
GST_WARNING_OBJECT (self, "Couldn't get port information: %s (0x%08x)",
gst_omx_error_to_string (err), err);
/* Fallback */
in_port_index = 0;
out_port_index = 1;
} else {
GST_DEBUG_OBJECT (self, "Detected %u ports, starting at %u",
(guint) param.nPorts, (guint) param.nStartPortNumber);
in_port_index = param.nStartPortNumber + 0;
out_port_index = param.nStartPortNumber + 1;
}
}
self->enc_in_port = gst_omx_component_add_port (self->enc, in_port_index);
self->enc_out_port = gst_omx_component_add_port (self->enc, out_port_index);
if (!self->enc_in_port || !self->enc_out_port)
return FALSE;
return TRUE;
}
static gboolean
gst_omx_audio_enc_shutdown (GstOMXAudioEnc * self)
{
OMX_STATETYPE state;
GST_DEBUG_OBJECT (self, "Shutting down encoder");
state = gst_omx_component_get_state (self->enc, 0);
if (state > OMX_StateLoaded || state == OMX_StateInvalid) {
if (state > OMX_StateIdle) {
gst_omx_component_set_state (self->enc, OMX_StateIdle);
gst_omx_component_get_state (self->enc, 5 * GST_SECOND);
}
gst_omx_component_set_state (self->enc, OMX_StateLoaded);
gst_omx_port_deallocate_buffers (self->enc_in_port);
gst_omx_port_deallocate_buffers (self->enc_out_port);
if (state > OMX_StateLoaded)
gst_omx_component_get_state (self->enc, 5 * GST_SECOND);
}
return TRUE;
}
static gboolean
gst_omx_audio_enc_close (GstAudioEncoder * encoder)
{
GstOMXAudioEnc *self = GST_OMX_AUDIO_ENC (encoder);
GST_DEBUG_OBJECT (self, "Closing encoder");
if (!gst_omx_audio_enc_shutdown (self))
return FALSE;
self->enc_in_port = NULL;
self->enc_out_port = NULL;
if (self->enc)
gst_omx_component_unref (self->enc);
self->enc = NULL;
return TRUE;
}
static void
gst_omx_audio_enc_finalize (GObject * object)
{
GstOMXAudioEnc *self = GST_OMX_AUDIO_ENC (object);
g_mutex_clear (&self->drain_lock);
g_cond_clear (&self->drain_cond);
G_OBJECT_CLASS (gst_omx_audio_enc_parent_class)->finalize (object);
}
static GstStateChangeReturn
gst_omx_audio_enc_change_state (GstElement * element, GstStateChange transition)
{
GstOMXAudioEnc *self;
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
g_return_val_if_fail (GST_IS_OMX_AUDIO_ENC (element),
GST_STATE_CHANGE_FAILURE);
self = GST_OMX_AUDIO_ENC (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:
if (self->enc_in_port)
gst_omx_port_set_flushing (self->enc_in_port, 5 * GST_SECOND, TRUE);
if (self->enc_out_port)
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, TRUE);
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
break;
default:
break;
}
ret =
GST_ELEMENT_CLASS (gst_omx_audio_enc_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;
if (!gst_omx_audio_enc_shutdown (self))
ret = GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
{
GstOMXAudioEncClass *klass;
GstOMXPort *port = self->enc_out_port;
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
OMX_ERRORTYPE err;
klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
acq_return = gst_omx_port_acquire_buffer (port, &buf, GST_OMX_WAIT);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) {
goto component_error;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
goto flushing;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
goto eos;
}
if (!gst_pad_has_current_caps (GST_AUDIO_ENCODER_SRC_PAD (self))
|| acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
GstAudioInfo *info =
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self));
GstCaps *caps;
GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
/* Reallocate all buffers */
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
err = gst_omx_port_set_enabled (port, FALSE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
caps = klass->get_caps (self, self->enc_out_port, info);
if (!caps) {
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
GST_DEBUG_OBJECT (self, "Setting output caps: %" GST_PTR_FORMAT, caps);
if (!gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (self), caps)) {
gst_caps_unref (caps);
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
gst_caps_unref (caps);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_populate (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
/* Now get a buffer */
if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) {
return;
}
}
g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
if (!buf) {
g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto eos;
}
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
(guint) buf->omx_buf->nFlags,
(guint64) GST_OMX_GET_TICKS (buf->omx_buf->nTimeStamp));
/* This prevents a deadlock between the srcpad stream
* lock and the videocodec stream lock, if ::reset()
* is called at the wrong time
*/
if (gst_omx_port_is_flushing (self->enc_out_port)) {
GST_DEBUG_OBJECT (self, "Flushing");
gst_omx_port_release_buffer (self->enc_out_port, buf);
goto flushing;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
&& buf->omx_buf->nFilledLen > 0) {
GstCaps *caps;
GstBuffer *codec_data;
GstMapInfo map = GST_MAP_INFO_INIT;
GST_DEBUG_OBJECT (self, "Handling codec data");
caps =
gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD
(self)));
codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
memcpy (map.data,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
gst_buffer_unmap (codec_data, &map);
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps);
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
gst_caps_unref (caps);
flow_ret = GST_FLOW_OK;
} else if (buf->omx_buf->nFilledLen > 0) {
GstBuffer *outbuf;
guint n_samples;
GST_DEBUG_OBJECT (self, "Handling output data");
n_samples =
klass->get_num_samples (self, self->enc_out_port,
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
if (buf->omx_buf->nFilledLen > 0) {
GstMapInfo map = GST_MAP_INFO_INIT;
outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
memcpy (map.data,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
gst_buffer_unmap (outbuf, &map);
} else {
outbuf = gst_buffer_new ();
}
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale (GST_OMX_GET_TICKS (buf->omx_buf->nTimeStamp),
GST_SECOND, OMX_TICKS_PER_SECOND);
if (buf->omx_buf->nTickCount != 0)
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
OMX_TICKS_PER_SECOND);
flow_ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
outbuf, n_samples);
}
GST_DEBUG_OBJECT (self, "Handled output data");
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
component_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->enc),
gst_omx_component_get_last_error (self->enc)));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
g_mutex_lock (&self->drain_lock);
if (self->draining) {
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
}
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
g_mutex_unlock (&self->drain_lock);
return;
}
eos:
{
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
flow_ret = GST_FLOW_OK;
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
} else {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_ENCODER_STREAM_LOCK (self);
self->downstream_flow_ret = flow_ret;
/* Here we fallback and pause the task for the EOS case */
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->started = FALSE;
} else if (flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Internal data stream error."),
("stream stopped, reason %s", gst_flow_get_name (flow_ret)));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->started = FALSE;
} else if (flow_ret == GST_FLOW_FLUSHING) {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
g_mutex_lock (&self->drain_lock);
if (self->draining) {
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
}
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->started = FALSE;
g_mutex_unlock (&self->drain_lock);
}
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure output port"));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
caps_failed:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
release_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to relase output buffer to component: %s (0x%08x)",
gst_omx_error_to_string (err), err));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
}
static gboolean
gst_omx_audio_enc_start (GstAudioEncoder * encoder)
{
GstOMXAudioEnc *self;
self = GST_OMX_AUDIO_ENC (encoder);
self->last_upstream_ts = 0;
self->downstream_flow_ret = GST_FLOW_OK;
return TRUE;
}
static gboolean
gst_omx_audio_enc_stop (GstAudioEncoder * encoder)
{
GstOMXAudioEnc *self;
self = GST_OMX_AUDIO_ENC (encoder);
GST_DEBUG_OBJECT (self, "Stopping encoder");
gst_omx_port_set_flushing (self->enc_in_port, 5 * GST_SECOND, TRUE);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, TRUE);
gst_pad_stop_task (GST_AUDIO_ENCODER_SRC_PAD (encoder));
if (gst_omx_component_get_state (self->enc, 0) > OMX_StateIdle)
gst_omx_component_set_state (self->enc, OMX_StateIdle);
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
g_mutex_lock (&self->drain_lock);
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
g_mutex_unlock (&self->drain_lock);
gst_omx_component_get_state (self->enc, 5 * GST_SECOND);
return TRUE;
}
static gboolean
gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
{
GstOMXAudioEnc *self;
GstOMXAudioEncClass *klass;
gboolean needs_disable = FALSE;
OMX_PARAM_PORTDEFINITIONTYPE port_def;
OMX_AUDIO_PARAM_PCMMODETYPE pcm_param;
gint i;
OMX_ERRORTYPE err;
self = GST_OMX_AUDIO_ENC (encoder);
klass = GST_OMX_AUDIO_ENC_GET_CLASS (encoder);
GST_DEBUG_OBJECT (self, "Setting new caps");
/* Set audio encoder base class properties */
gst_audio_encoder_set_frame_samples_min (encoder,
gst_util_uint64_scale_ceil (OMX_MIN_PCMPAYLOAD_MSEC,
GST_MSECOND * info->rate, GST_SECOND));
gst_audio_encoder_set_frame_samples_max (encoder, 0);
gst_omx_port_get_port_definition (self->enc_in_port, &port_def);
needs_disable =
gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateLoaded;
/* If the component is not in Loaded state and a real format change happens
* we have to disable the port and re-allocate all buffers. If no real
* format change happened we can just exit here.
*/
if (needs_disable) {
GST_DEBUG_OBJECT (self, "Need to disable and drain encoder");
gst_omx_audio_enc_drain (self);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, TRUE);
/* Wait until the srcpad loop is finished,
* unlock GST_AUDIO_ENCODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
gst_pad_stop_task (GST_AUDIO_ENCODER_SRC_PAD (encoder));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
if (klass->cdata.hacks & GST_OMX_HACK_NO_COMPONENT_RECONFIGURE) {
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
gst_omx_audio_enc_stop (GST_AUDIO_ENCODER (self));
gst_omx_audio_enc_close (GST_AUDIO_ENCODER (self));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
if (!gst_omx_audio_enc_open (GST_AUDIO_ENCODER (self)))
return FALSE;
needs_disable = FALSE;
/* The local port_def is now obsolete so get it again. */
gst_omx_port_get_port_definition (self->enc_in_port, &port_def);
} else {
/* Disabling at the same time input port and output port is only
* required when a buffer is shared between the ports. This cannot
* be the case for a encoder because its input and output buffers
* are of different nature. So let's disable ports sequencially.
* Starting from IL 1.2.0, this point has been clarified.
* OMX_SendCommand will return an error if the IL client attempts to
* call it when there is already an on-going command being processed.
* The exception is for buffer sharing above and the event
* OMX_EventPortNeedsDisable will be sent to request disabling the
* other port at the same time. */
if (gst_omx_port_set_enabled (self->enc_in_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_buffers_released (self->enc_in_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_in_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_set_enabled (self->enc_out_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_buffers_released (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->enc_out_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
}
GST_DEBUG_OBJECT (self, "Encoder drained and disabled");
}
port_def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
GST_DEBUG_OBJECT (self, "Setting inport port definition");
if (gst_omx_port_update_port_definition (self->enc_in_port,
&port_def) != OMX_ErrorNone)
return FALSE;
GST_OMX_INIT_STRUCT (&pcm_param);
pcm_param.nPortIndex = self->enc_in_port->index;
pcm_param.nChannels = info->channels;
pcm_param.eNumData =
((info->finfo->flags & GST_AUDIO_FORMAT_FLAG_SIGNED) ?
OMX_NumericalDataSigned : OMX_NumericalDataUnsigned);
pcm_param.eEndian =
((info->finfo->endianness == G_LITTLE_ENDIAN) ?
OMX_EndianLittle : OMX_EndianBig);
pcm_param.bInterleaved = OMX_TRUE;
pcm_param.nBitPerSample = info->finfo->width;
pcm_param.nSamplingRate = info->rate;
pcm_param.ePCMMode = OMX_AUDIO_PCMModeLinear;
for (i = 0; i < pcm_param.nChannels; i++) {
OMX_AUDIO_CHANNELTYPE pos;
switch (info->position[i]) {
case GST_AUDIO_CHANNEL_POSITION_MONO:
case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER:
pos = OMX_AUDIO_ChannelCF;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
pos = OMX_AUDIO_ChannelLF;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
pos = OMX_AUDIO_ChannelRF;
break;
case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT:
pos = OMX_AUDIO_ChannelLS;
break;
case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT:
pos = OMX_AUDIO_ChannelRS;
break;
case GST_AUDIO_CHANNEL_POSITION_LFE1:
pos = OMX_AUDIO_ChannelLFE;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER:
pos = OMX_AUDIO_ChannelCS;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT:
pos = OMX_AUDIO_ChannelLR;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT:
pos = OMX_AUDIO_ChannelRR;
break;
default:
pos = OMX_AUDIO_ChannelNone;
break;
}
pcm_param.eChannelMapping[i] = pos;
}
GST_DEBUG_OBJECT (self, "Setting PCM parameters");
err =
gst_omx_component_set_parameter (self->enc, OMX_IndexParamAudioPcm,
&pcm_param);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Failed to set PCM parameters: %s (0x%08x)",
gst_omx_error_to_string (err), err);
return FALSE;
}
if (klass->set_format) {
if (!klass->set_format (self, self->enc_in_port, info)) {
GST_ERROR_OBJECT (self, "Subclass failed to set the new format");
return FALSE;
}
}
GST_DEBUG_OBJECT (self, "Updating outport port definition");
if (gst_omx_port_update_port_definition (self->enc_out_port,
NULL) != OMX_ErrorNone)
return FALSE;
GST_DEBUG_OBJECT (self, "Enabling component");
if (needs_disable) {
if (gst_omx_port_set_enabled (self->enc_in_port, TRUE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_allocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if ((klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) {
if (gst_omx_port_set_enabled (self->enc_out_port, TRUE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_allocate_buffers (self->enc_out_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_out_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
}
if (gst_omx_port_wait_enabled (self->enc_in_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_mark_reconfigured (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
} else {
if (!(klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) {
/* Disable output port */
if (gst_omx_port_set_enabled (self->enc_out_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_set_state (self->enc,
OMX_StateIdle) != OMX_ErrorNone)
return FALSE;
/* Need to allocate buffers to reach Idle state */
if (gst_omx_port_allocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
} else {
if (gst_omx_component_set_state (self->enc,
OMX_StateIdle) != OMX_ErrorNone)
return FALSE;
/* Need to allocate buffers to reach Idle state */
if (gst_omx_port_allocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_allocate_buffers (self->enc_out_port) != OMX_ErrorNone)
return FALSE;
}
if (gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateIdle)
return FALSE;
if (gst_omx_component_set_state (self->enc,
OMX_StateExecuting) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateExecuting)
return FALSE;
}
/* Unset flushing to allow ports to accept data again */
gst_omx_port_set_flushing (self->enc_in_port, 5 * GST_SECOND, FALSE);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, FALSE);
if (gst_omx_component_get_last_error (self->enc) != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Component in error state: %s (0x%08x)",
gst_omx_component_get_last_error_string (self->enc),
gst_omx_component_get_last_error (self->enc));
return FALSE;
}
/* Start the srcpad loop again */
GST_DEBUG_OBJECT (self, "Starting task again");
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_enc_loop, encoder, NULL);
return TRUE;
}
static void
gst_omx_audio_enc_flush (GstAudioEncoder * encoder)
{
GstOMXAudioEnc *self;
self = GST_OMX_AUDIO_ENC (encoder);
GST_DEBUG_OBJECT (self, "Resetting encoder");
gst_omx_port_set_flushing (self->enc_in_port, 5 * GST_SECOND, TRUE);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, TRUE);
/* Wait until the srcpad loop is finished */
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
GST_PAD_STREAM_LOCK (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_PAD_STREAM_UNLOCK (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
gst_omx_port_set_flushing (self->enc_in_port, 5 * GST_SECOND, FALSE);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, FALSE);
gst_omx_port_populate (self->enc_out_port);
/* Start the srcpad loop again */
self->last_upstream_ts = 0;
self->downstream_flow_ret = GST_FLOW_OK;
self->started = FALSE;
gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_enc_loop, encoder, NULL);
}
static GstFlowReturn
gst_omx_audio_enc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
{
GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR;
GstOMXAudioEnc *self;
GstOMXPort *port;
GstOMXBuffer *buf;
gsize size;
guint offset = 0;
GstClockTime timestamp, duration, timestamp_offset = 0;
OMX_ERRORTYPE err;
self = GST_OMX_AUDIO_ENC (encoder);
if (self->downstream_flow_ret != GST_FLOW_OK) {
return self->downstream_flow_ret;
}
if (inbuf == NULL)
return gst_omx_audio_enc_drain (self);
GST_DEBUG_OBJECT (self, "Handling frame");
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
duration = GST_BUFFER_DURATION (inbuf);
port = self->enc_in_port;
size = gst_buffer_get_size (inbuf);
while (offset < 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_ENCODER_STREAM_UNLOCK (self);
acq_ret = gst_omx_port_acquire_buffer (port, &buf, GST_OMX_WAIT);
if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto component_error;
} else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto flushing;
} else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
/* Reallocate all buffers */
err = gst_omx_port_set_enabled (port, FALSE);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
goto reconfigure_error;
}
/* Now get a new buffer and fill it */
GST_AUDIO_ENCODER_STREAM_LOCK (self);
continue;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
g_assert (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL);
if (self->downstream_flow_ret != GST_FLOW_OK) {
gst_omx_port_release_buffer (port, buf);
return self->downstream_flow_ret;
}
if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <= 0) {
gst_omx_port_release_buffer (port, buf);
goto full_buffer;
}
GST_DEBUG_OBJECT (self, "Handling frame at offset %d", offset);
/* Copy the buffer content in chunks of size as requested
* by the port */
buf->omx_buf->nFilledLen =
MIN (size - offset, buf->omx_buf->nAllocLen - buf->omx_buf->nOffset);
gst_buffer_extract (inbuf, offset,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
/* 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, size);
}
if (timestamp != GST_CLOCK_TIME_NONE) {
GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp,
gst_util_uint64_scale (timestamp + timestamp_offset,
OMX_TICKS_PER_SECOND, GST_SECOND));
self->last_upstream_ts = timestamp + timestamp_offset;
}
if (duration != GST_CLOCK_TIME_NONE) {
buf->omx_buf->nTickCount =
gst_util_uint64_scale (buf->omx_buf->nFilledLen, duration, size);
buf->omx_buf->nTickCount =
gst_util_uint64_scale (buf->omx_buf->nTickCount,
OMX_TICKS_PER_SECOND, GST_SECOND);
self->last_upstream_ts += duration;
}
offset += buf->omx_buf->nFilledLen;
self->started = TRUE;
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
}
GST_DEBUG_OBJECT (self, "Passed frame to component");
return self->downstream_flow_ret;
full_buffer:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("Got OpenMAX buffer with no free space (%p, %u/%u)", buf,
(guint) buf->omx_buf->nOffset, (guint) buf->omx_buf->nAllocLen));
return GST_FLOW_ERROR;
}
component_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->enc),
gst_omx_component_get_last_error (self->enc)));
return GST_FLOW_ERROR;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING");
return GST_FLOW_FLUSHING;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure input port"));
return GST_FLOW_ERROR;
}
release_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to relase input buffer to component: %s (0x%08x)",
gst_omx_error_to_string (err), err));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_omx_audio_enc_drain (GstOMXAudioEnc * self)
{
GstOMXAudioEncClass *klass;
GstOMXBuffer *buf;
GstOMXAcquireBufferReturn acq_ret;
OMX_ERRORTYPE err;
GST_DEBUG_OBJECT (self, "Draining component");
klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
if (!self->started) {
GST_DEBUG_OBJECT (self, "Component not started yet");
return GST_FLOW_OK;
}
self->started = FALSE;
if ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) {
GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers");
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_ENCODER_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. */
acq_ret = gst_omx_port_acquire_buffer (self->enc_in_port, &buf, GST_OMX_WAIT);
if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK) {
GST_AUDIO_ENCODER_STREAM_LOCK (self);
GST_ERROR_OBJECT (self, "Failed to acquire buffer for draining: %d",
acq_ret);
return GST_FLOW_ERROR;
}
g_mutex_lock (&self->drain_lock);
self->draining = TRUE;
buf->omx_buf->nFilledLen = 0;
GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp,
gst_util_uint64_scale (self->last_upstream_ts, OMX_TICKS_PER_SECOND,
GST_SECOND));
buf->omx_buf->nTickCount = 0;
buf->omx_buf->nFlags |= OMX_BUFFERFLAG_EOS;
err = gst_omx_port_release_buffer (self->enc_in_port, buf);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Failed to drain component: %s (0x%08x)",
gst_omx_error_to_string (err), err);
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_ENCODER_STREAM_LOCK (self);
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (self, "Waiting until component is drained");
g_cond_wait (&self->drain_cond, &self->drain_lock);
GST_DEBUG_OBJECT (self, "Drained component");
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_ENCODER_STREAM_LOCK (self);
self->started = FALSE;
return GST_FLOW_OK;
}
Reference in a new issue View git blame Copy permalink