gstreamer/sys/mediafoundation/gstmfaudioenc.cpp

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/* GStreamer
* Copyright (C) 2020 Seungha Yang <seungha@centricular.com>
*
* 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 <gst/gst.h>
#include "gstmfaudioenc.h"
#include <wrl.h>
#include <string.h>
using namespace Microsoft::WRL;
GST_DEBUG_CATEGORY (gst_mf_audio_enc_debug);
#define GST_CAT_DEFAULT gst_mf_audio_enc_debug
#define gst_mf_audio_enc_parent_class parent_class
G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstMFAudioEnc, gst_mf_audio_enc,
GST_TYPE_AUDIO_ENCODER,
GST_DEBUG_CATEGORY_INIT (gst_mf_audio_enc_debug, "mfaudioenc", 0,
"mfaudioenc"));
static gboolean gst_mf_audio_enc_open (GstAudioEncoder * enc);
static gboolean gst_mf_audio_enc_close (GstAudioEncoder * enc);
static gboolean gst_mf_audio_enc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_mf_audio_enc_handle_frame (GstAudioEncoder * enc,
GstBuffer *buffer);
static GstFlowReturn gst_mf_audio_enc_drain (GstAudioEncoder * enc);
static void gst_mf_audio_enc_flush (GstAudioEncoder * enc);
static void
gst_mf_audio_enc_class_init (GstMFAudioEncClass * klass)
{
GstAudioEncoderClass *audioenc_class = GST_AUDIO_ENCODER_CLASS (klass);
audioenc_class->open = GST_DEBUG_FUNCPTR (gst_mf_audio_enc_open);
audioenc_class->close = GST_DEBUG_FUNCPTR (gst_mf_audio_enc_close);
audioenc_class->set_format = GST_DEBUG_FUNCPTR (gst_mf_audio_enc_set_format);
audioenc_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_mf_audio_enc_handle_frame);
audioenc_class->flush =
GST_DEBUG_FUNCPTR (gst_mf_audio_enc_flush);
gst_type_mark_as_plugin_api (GST_TYPE_MF_AUDIO_ENC, (GstPluginAPIFlags) 0);
}
static void
gst_mf_audio_enc_init (GstMFAudioEnc * self)
{
gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (self), TRUE);
}
static gboolean
gst_mf_audio_enc_open (GstAudioEncoder * enc)
{
GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
GstMFAudioEncClass *klass = GST_MF_AUDIO_ENC_GET_CLASS (enc);
GstMFTransformEnumParams enum_params = { 0, };
MFT_REGISTER_TYPE_INFO output_type;
gboolean ret;
output_type.guidMajorType = MFMediaType_Audio;
output_type.guidSubtype = klass->codec_id;
enum_params.category = MFT_CATEGORY_AUDIO_ENCODER;
enum_params.enum_flags = klass->enum_flags;
enum_params.output_typeinfo = &output_type;
enum_params.device_index = klass->device_index;
GST_DEBUG_OBJECT (self, "Create MFT with enum flags 0x%x, device index %d",
klass->enum_flags, klass->device_index);
self->transform = gst_mf_transform_new (&enum_params);
ret = !!self->transform;
if (!ret)
GST_ERROR_OBJECT (self, "Cannot create MFT object");
return ret;
}
static gboolean
gst_mf_audio_enc_close (GstAudioEncoder * enc)
{
GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
gst_clear_object (&self->transform);
return TRUE;
}
static gboolean
gst_mf_audio_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
GstMFAudioEncClass *klass = GST_MF_AUDIO_ENC_GET_CLASS (enc);
ComPtr<IMFMediaType> in_type;
ComPtr<IMFMediaType> out_type;
GST_DEBUG_OBJECT (self, "Set format");
gst_mf_audio_enc_drain (enc);
if (!gst_mf_transform_open (self->transform)) {
GST_ERROR_OBJECT (self, "Failed to open MFT");
return FALSE;
}
g_assert (klass->get_output_type != NULL);
if (!klass->get_output_type (self, info, &out_type)) {
GST_ERROR_OBJECT (self, "subclass failed to set output type");
return FALSE;
}
gst_mf_dump_attributes (out_type.Get(), "Set output type", GST_LEVEL_DEBUG);
if (!gst_mf_transform_set_output_type (self->transform, out_type.Get ())) {
GST_ERROR_OBJECT (self, "Couldn't set output type");
return FALSE;
}
g_assert (klass->get_input_type != NULL);
if (!klass->get_input_type (self, info, &in_type)) {
GST_ERROR_OBJECT (self, "subclass didn't provide input type");
return FALSE;
}
gst_mf_dump_attributes (in_type.Get(), "Set input type", GST_LEVEL_DEBUG);
if (!gst_mf_transform_set_input_type (self->transform, in_type.Get ())) {
GST_ERROR_OBJECT (self, "Couldn't set input media type");
return FALSE;
}
g_assert (klass->set_src_caps != NULL);
if (!klass->set_src_caps (self, info))
return FALSE;
g_assert (klass->frame_samples > 0);
gst_audio_encoder_set_frame_samples_min (enc, klass->frame_samples);
gst_audio_encoder_set_frame_samples_max (enc, klass->frame_samples);
gst_audio_encoder_set_frame_max (enc, 1);
/* mediafoundation encoder needs timestamp and duration */
self->sample_count = 0;
self->sample_duration_in_mf = gst_util_uint64_scale (klass->frame_samples,
10000000, GST_AUDIO_INFO_RATE (info));
GST_DEBUG_OBJECT (self,
"Calculated sample duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (self->sample_duration_in_mf * 100));
return TRUE;
}
static gboolean
gst_mf_audio_enc_process_input (GstMFAudioEnc * self, GstBuffer * buffer)
{
HRESULT hr;
ComPtr<IMFSample> sample;
ComPtr<IMFMediaBuffer> media_buffer;
BYTE *data;
gboolean res = FALSE;
GstMapInfo info;
guint64 timestamp;
if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
GST_ELEMENT_ERROR (self,
RESOURCE, READ, ("Couldn't map input buffer"), (NULL));
return FALSE;
}
GST_TRACE_OBJECT (self, "Process buffer %" GST_PTR_FORMAT, buffer);
timestamp = self->sample_count * self->sample_duration_in_mf;
hr = MFCreateSample (sample.GetAddressOf ());
if (!gst_mf_result (hr))
goto done;
hr = MFCreateMemoryBuffer (info.size, media_buffer.GetAddressOf ());
if (!gst_mf_result (hr))
goto done;
hr = media_buffer->Lock (&data, NULL, NULL);
if (!gst_mf_result (hr))
goto done;
memcpy (data, info.data, info.size);
media_buffer->Unlock ();
hr = media_buffer->SetCurrentLength (info.size);
if (!gst_mf_result (hr))
goto done;
hr = sample->AddBuffer (media_buffer.Get ());
if (!gst_mf_result (hr))
goto done;
hr = sample->SetSampleTime (timestamp);
if (!gst_mf_result (hr))
goto done;
hr = sample->SetSampleDuration (self->sample_duration_in_mf);
if (!gst_mf_result (hr))
goto done;
if (!gst_mf_transform_process_input (self->transform, sample.Get ())) {
GST_ERROR_OBJECT (self, "Failed to process input");
goto done;
}
self->sample_count++;
res = TRUE;
done:
gst_buffer_unmap (buffer, &info);
return res;
}
static GstFlowReturn
gst_mf_audio_enc_process_output (GstMFAudioEnc * self)
{
GstMFAudioEncClass *klass = GST_MF_AUDIO_ENC_GET_CLASS (self);
HRESULT hr;
BYTE *data;
ComPtr<IMFMediaBuffer> media_buffer;
ComPtr<IMFSample> sample;
GstBuffer *buffer;
GstFlowReturn res = GST_FLOW_ERROR;
DWORD buffer_len;
res = gst_mf_transform_get_output (self->transform, sample.GetAddressOf ());
if (res != GST_FLOW_OK)
return res;
hr = sample->GetBufferByIndex (0, media_buffer.GetAddressOf ());
if (!gst_mf_result (hr))
return GST_FLOW_ERROR;
hr = media_buffer->Lock (&data, NULL, &buffer_len);
if (!gst_mf_result (hr))
return GST_FLOW_ERROR;
buffer = gst_audio_encoder_allocate_output_buffer (GST_AUDIO_ENCODER (self),
buffer_len);
gst_buffer_fill (buffer, 0, data, buffer_len);
media_buffer->Unlock ();
return gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self), buffer,
klass->frame_samples);
}
static GstFlowReturn
gst_mf_audio_enc_handle_frame (GstAudioEncoder * enc,
GstBuffer *buffer)
{
GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
GstFlowReturn ret;
if (!buffer)
return gst_mf_audio_enc_drain (enc);
if (!gst_mf_audio_enc_process_input (self, buffer)) {
GST_ERROR_OBJECT (self, "Failed to process input");
return GST_FLOW_ERROR;
}
do {
ret = gst_mf_audio_enc_process_output (self);
} while (ret == GST_FLOW_OK);
if (ret == GST_MF_TRANSFORM_FLOW_NEED_DATA)
ret = GST_FLOW_OK;
return ret;
}
static GstFlowReturn
gst_mf_audio_enc_drain (GstAudioEncoder * enc)
{
GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
GstFlowReturn ret = GST_FLOW_OK;
if (!self->transform)
return GST_FLOW_OK;
gst_mf_transform_drain (self->transform);
do {
ret = gst_mf_audio_enc_process_output (self);
} while (ret == GST_FLOW_OK);
if (ret == GST_MF_TRANSFORM_FLOW_NEED_DATA)
ret = GST_FLOW_OK;
return ret;
}
static void
gst_mf_audio_enc_flush (GstAudioEncoder * enc)
{
GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
if (!self->transform)
return;
gst_mf_transform_flush (self->transform);
}