mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-11 11:51:34 +00:00
be957f6c61
See also https://docs.microsoft.com/en-us/windows/win32/medfound/aac-decoder Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1596>
285 lines
7.7 KiB
C++
285 lines
7.7 KiB
C++
/* GStreamer
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* Copyright (C) 2022 Seungha Yang <seungha@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include "gstmfaudiodecoder.h"
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#include <wrl.h>
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#include <string.h>
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/* *INDENT-OFF* */
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using namespace Microsoft::WRL;
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/* *INDENT-ON* */
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GST_DEBUG_CATEGORY (gst_mf_audio_decoder_debug);
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#define GST_CAT_DEFAULT gst_mf_audio_decoder_debug
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#define gst_mf_audio_decoder_parent_class parent_class
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G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstMFAudioDecoder, gst_mf_audio_decoder,
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GST_TYPE_AUDIO_DECODER,
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GST_DEBUG_CATEGORY_INIT (gst_mf_audio_decoder_debug, "mfaudiodecoder", 0,
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"mfaudiodecoder"));
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static gboolean gst_mf_audio_decoder_open (GstAudioDecoder * dec);
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static gboolean gst_mf_audio_decoder_close (GstAudioDecoder * dec);
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static gboolean gst_mf_audio_decoder_set_format (GstAudioDecoder * dec,
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GstCaps * caps);
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static GstFlowReturn gst_mf_audio_decoder_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static GstFlowReturn gst_mf_audio_decoder_drain (GstAudioDecoder * dec);
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static void gst_mf_audio_decoder_flush (GstAudioDecoder * dec, gboolean hard);
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static void
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gst_mf_audio_decoder_class_init (GstMFAudioDecoderClass * klass)
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{
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GstAudioDecoderClass *audiodec_class = GST_AUDIO_DECODER_CLASS (klass);
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audiodec_class->open = GST_DEBUG_FUNCPTR (gst_mf_audio_decoder_open);
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audiodec_class->close = GST_DEBUG_FUNCPTR (gst_mf_audio_decoder_close);
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audiodec_class->set_format =
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GST_DEBUG_FUNCPTR (gst_mf_audio_decoder_set_format);
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audiodec_class->handle_frame =
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GST_DEBUG_FUNCPTR (gst_mf_audio_decoder_handle_frame);
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audiodec_class->flush = GST_DEBUG_FUNCPTR (gst_mf_audio_decoder_flush);
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gst_type_mark_as_plugin_api (GST_TYPE_MF_AUDIO_DECODER,
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(GstPluginAPIFlags) 0);
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}
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static void
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gst_mf_audio_decoder_init (GstMFAudioDecoder * self)
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{
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gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE);
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}
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static gboolean
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gst_mf_audio_decoder_open (GstAudioDecoder * dec)
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{
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GstMFAudioDecoder *self = GST_MF_AUDIO_DECODER (dec);
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GstMFAudioDecoderClass *klass = GST_MF_AUDIO_DECODER_GET_CLASS (dec);
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GstMFTransformEnumParams enum_params = { 0, };
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MFT_REGISTER_TYPE_INFO input_type;
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input_type.guidMajorType = MFMediaType_Audio;
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input_type.guidSubtype = klass->codec_id;
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enum_params.category = MFT_CATEGORY_AUDIO_DECODER;
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enum_params.enum_flags = klass->enum_flags;
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enum_params.input_typeinfo = &input_type;
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enum_params.device_index = klass->device_index;
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GST_DEBUG_OBJECT (self, "Create MFT with enum flags 0x%x, device index %d",
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klass->enum_flags, klass->device_index);
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self->transform = gst_mf_transform_new (&enum_params);
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if (!self->transform) {
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GST_ERROR_OBJECT (self, "Cannot create MFT object");
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_mf_audio_decoder_close (GstAudioDecoder * dec)
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{
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GstMFAudioDecoder *self = GST_MF_AUDIO_DECODER (dec);
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gst_clear_object (&self->transform);
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return TRUE;
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}
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static gboolean
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gst_mf_audio_decoder_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstMFAudioDecoder *self = GST_MF_AUDIO_DECODER (dec);
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GstMFAudioDecoderClass *klass = GST_MF_AUDIO_DECODER_GET_CLASS (dec);
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g_assert (klass->set_format != nullptr);
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GST_DEBUG_OBJECT (self, "Set format");
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gst_mf_audio_decoder_drain (dec);
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if (!gst_mf_transform_open (self->transform)) {
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GST_ERROR_OBJECT (self, "Failed to open MFT");
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return FALSE;
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}
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if (!klass->set_format (self, self->transform, caps)) {
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GST_ERROR_OBJECT (self, "Failed to set format");
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_mf_audio_decoder_process_input (GstMFAudioDecoder * self,
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GstBuffer * buffer)
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{
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HRESULT hr;
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ComPtr < IMFSample > sample;
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ComPtr < IMFMediaBuffer > media_buffer;
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BYTE *data;
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gboolean res = FALSE;
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GstMapInfo info;
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if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
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GST_ELEMENT_ERROR (self,
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RESOURCE, READ, ("Couldn't map input buffer"), (nullptr));
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return FALSE;
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}
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GST_TRACE_OBJECT (self, "Process buffer %" GST_PTR_FORMAT, buffer);
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hr = MFCreateSample (&sample);
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if (!gst_mf_result (hr))
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goto done;
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hr = MFCreateMemoryBuffer (info.size, &media_buffer);
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if (!gst_mf_result (hr))
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goto done;
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hr = media_buffer->Lock (&data, nullptr, nullptr);
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if (!gst_mf_result (hr))
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goto done;
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memcpy (data, info.data, info.size);
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media_buffer->Unlock ();
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hr = media_buffer->SetCurrentLength (info.size);
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if (!gst_mf_result (hr))
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goto done;
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hr = sample->AddBuffer (media_buffer.Get ());
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if (!gst_mf_result (hr))
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goto done;
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if (!gst_mf_transform_process_input (self->transform, sample.Get ())) {
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GST_ERROR_OBJECT (self, "Failed to process input");
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goto done;
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}
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res = TRUE;
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done:
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gst_buffer_unmap (buffer, &info);
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return res;
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}
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static GstFlowReturn
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gst_mf_audio_decoder_process_output (GstMFAudioDecoder * self)
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{
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HRESULT hr;
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BYTE *data = nullptr;
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ComPtr < IMFMediaBuffer > media_buffer;
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ComPtr < IMFSample > sample;
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GstBuffer *buffer;
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GstFlowReturn res = GST_FLOW_ERROR;
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DWORD buffer_len = 0;
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res = gst_mf_transform_get_output (self->transform, &sample);
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if (res != GST_FLOW_OK)
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return res;
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hr = sample->GetBufferByIndex (0, &media_buffer);
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if (!gst_mf_result (hr))
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return GST_FLOW_ERROR;
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hr = media_buffer->Lock (&data, nullptr, &buffer_len);
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if (!gst_mf_result (hr))
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return GST_FLOW_ERROR;
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/* Can happen while draining */
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if (buffer_len == 0 || !data) {
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GST_DEBUG_OBJECT (self, "Empty media buffer");
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media_buffer->Unlock ();
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return GST_FLOW_OK;
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}
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buffer = gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
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buffer_len);
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gst_buffer_fill (buffer, 0, data, buffer_len);
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media_buffer->Unlock ();
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return gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), buffer, 1);
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}
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static GstFlowReturn
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gst_mf_audio_decoder_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
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{
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GstMFAudioDecoder *self = GST_MF_AUDIO_DECODER (dec);
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GstFlowReturn ret;
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if (!buffer)
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return gst_mf_audio_decoder_drain (dec);
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if (!gst_mf_audio_decoder_process_input (self, buffer)) {
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GST_ERROR_OBJECT (self, "Failed to process input");
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return GST_FLOW_ERROR;
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}
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do {
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ret = gst_mf_audio_decoder_process_output (self);
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} while (ret == GST_FLOW_OK);
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if (ret == GST_MF_TRANSFORM_FLOW_NEED_DATA)
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ret = GST_FLOW_OK;
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return ret;
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}
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static GstFlowReturn
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gst_mf_audio_decoder_drain (GstAudioDecoder * dec)
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{
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GstMFAudioDecoder *self = GST_MF_AUDIO_DECODER (dec);
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GstFlowReturn ret = GST_FLOW_OK;
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if (!self->transform)
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return GST_FLOW_OK;
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gst_mf_transform_drain (self->transform);
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do {
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ret = gst_mf_audio_decoder_process_output (self);
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} while (ret == GST_FLOW_OK);
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if (ret == GST_MF_TRANSFORM_FLOW_NEED_DATA)
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ret = GST_FLOW_OK;
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return ret;
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}
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static void
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gst_mf_audio_decoder_flush (GstAudioDecoder * dec, gboolean hard)
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{
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GstMFAudioDecoder *self = GST_MF_AUDIO_DECODER (dec);
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if (!self->transform)
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return;
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gst_mf_transform_flush (self->transform);
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}
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