FreeRDP/libfreerdp/codec/dsp_ffmpeg.c
2024-06-03 17:55:22 +02:00

857 lines
22 KiB
C

/**
* FreeRDP: A Remote Desktop Protocol Implementation
* Digital Sound Processing - FFMPEG backend
*
* Copyright 2018 Armin Novak <armin.novak@thincast.com>
* Copyright 2018 Thincast Technologies GmbH
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#include <freerdp/config.h>
#include <freerdp/log.h>
#include <libavcodec/avcodec.h>
#include <libavutil/avutil.h>
#include <libavutil/opt.h>
#if defined(SWRESAMPLE_FOUND)
#include <libswresample/swresample.h>
#elif defined(AVRESAMPLE_FOUND)
#include <libavresample/avresample.h>
#else
#error "libswresample or libavresample required"
#endif
#include "dsp.h"
#include "dsp_ffmpeg.h"
#define TAG FREERDP_TAG("dsp.ffmpeg")
struct S_FREERDP_DSP_CONTEXT
{
AUDIO_FORMAT format;
BOOL isOpen;
BOOL encoder;
UINT32 bufferedSamples;
enum AVCodecID id;
AVCodec* codec;
AVCodecContext* context;
AVFrame* frame;
AVFrame* resampled;
AVFrame* buffered;
AVPacket* packet;
#if defined(SWRESAMPLE_FOUND)
SwrContext* rcontext;
#else
AVAudioResampleContext* rcontext;
#endif
wStream* channelmix;
};
static BOOL ffmpeg_codec_is_filtered(enum AVCodecID id, BOOL encoder)
{
switch (id)
{
#if !defined(WITH_DSP_EXPERIMENTAL)
case AV_CODEC_ID_ADPCM_IMA_OKI:
case AV_CODEC_ID_MP3:
case AV_CODEC_ID_ADPCM_MS:
case AV_CODEC_ID_G723_1:
return TRUE;
#endif
case AV_CODEC_ID_NONE:
return TRUE;
case AV_CODEC_ID_GSM_MS:
case AV_CODEC_ID_AAC:
case AV_CODEC_ID_AAC_LATM:
#if !defined(WITH_DSP_EXPERIMENTAL)
if (encoder)
return TRUE;
#endif
return FALSE;
default:
return FALSE;
}
}
static enum AVCodecID ffmpeg_get_avcodec(const AUDIO_FORMAT* WINPR_RESTRICT format)
{
if (!format)
return AV_CODEC_ID_NONE;
switch (format->wFormatTag)
{
case WAVE_FORMAT_UNKNOWN:
return AV_CODEC_ID_NONE;
case WAVE_FORMAT_PCM:
switch (format->wBitsPerSample)
{
case 16:
return AV_CODEC_ID_PCM_U16LE;
case 8:
return AV_CODEC_ID_PCM_U8;
default:
return AV_CODEC_ID_NONE;
}
case WAVE_FORMAT_DVI_ADPCM:
return AV_CODEC_ID_ADPCM_IMA_OKI;
case WAVE_FORMAT_ADPCM:
return AV_CODEC_ID_ADPCM_MS;
case WAVE_FORMAT_ALAW:
return AV_CODEC_ID_PCM_ALAW;
case WAVE_FORMAT_MULAW:
return AV_CODEC_ID_PCM_MULAW;
case WAVE_FORMAT_GSM610:
return AV_CODEC_ID_GSM_MS;
case WAVE_FORMAT_MSG723:
return AV_CODEC_ID_G723_1;
case WAVE_FORMAT_AAC_MS:
return AV_CODEC_ID_AAC;
case WAVE_FORMAT_OPUS:
return AV_CODEC_ID_OPUS;
default:
return AV_CODEC_ID_NONE;
}
}
static int ffmpeg_sample_format(const AUDIO_FORMAT* WINPR_RESTRICT format)
{
switch (format->wFormatTag)
{
case WAVE_FORMAT_PCM:
switch (format->wBitsPerSample)
{
case 8:
return AV_SAMPLE_FMT_U8;
case 16:
return AV_SAMPLE_FMT_S16;
default:
return FALSE;
}
case WAVE_FORMAT_DVI_ADPCM:
case WAVE_FORMAT_ADPCM:
return AV_SAMPLE_FMT_S16P;
case WAVE_FORMAT_MPEGLAYER3:
case WAVE_FORMAT_AAC_MS:
return AV_SAMPLE_FMT_FLTP;
case WAVE_FORMAT_OPUS:
return AV_SAMPLE_FMT_S16;
case WAVE_FORMAT_MSG723:
case WAVE_FORMAT_GSM610:
return AV_SAMPLE_FMT_S16P;
case WAVE_FORMAT_ALAW:
return AV_SAMPLE_FMT_S16;
default:
return FALSE;
}
}
static void ffmpeg_close_context(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context)
{
if (context)
{
if (context->context)
avcodec_free_context(&context->context);
if (context->frame)
av_frame_free(&context->frame);
if (context->resampled)
av_frame_free(&context->resampled);
if (context->buffered)
av_frame_free(&context->buffered);
if (context->packet)
av_packet_free(&context->packet);
if (context->rcontext)
{
#if defined(SWRESAMPLE_FOUND)
swr_free(&context->rcontext);
#else
avresample_free(&context->rcontext);
#endif
}
context->id = AV_CODEC_ID_NONE;
context->codec = NULL;
context->isOpen = FALSE;
context->context = NULL;
context->frame = NULL;
context->resampled = NULL;
context->packet = NULL;
context->rcontext = NULL;
}
}
static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context)
{
int ret = 0;
if (!context || context->isOpen)
return FALSE;
const AUDIO_FORMAT* format = &context->format;
if (!format)
return FALSE;
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
const int layout = av_get_default_channel_layout(format->nChannels);
#endif
context->id = ffmpeg_get_avcodec(format);
if (ffmpeg_codec_is_filtered(context->id, context->encoder))
goto fail;
if (context->encoder)
context->codec = avcodec_find_encoder(context->id);
else
context->codec = avcodec_find_decoder(context->id);
if (!context->codec)
goto fail;
context->context = avcodec_alloc_context3(context->codec);
if (!context->context)
goto fail;
switch (context->id)
{
/* We need support for multichannel and sample rates != 8000 */
case AV_CODEC_ID_GSM_MS:
context->context->strict_std_compliance = FF_COMPLIANCE_UNOFFICIAL;
break;
case AV_CODEC_ID_AAC:
context->context->profile = FF_PROFILE_AAC_MAIN;
break;
default:
break;
}
context->context->max_b_frames = 1;
context->context->delay = 0;
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
context->context->channels = format->nChannels;
context->context->channel_layout = layout;
#else
av_channel_layout_default(&context->context->ch_layout, format->nChannels);
#endif
context->context->sample_rate = format->nSamplesPerSec;
context->context->block_align = format->nBlockAlign;
context->context->bit_rate = format->nAvgBytesPerSec * 8;
context->context->sample_fmt = ffmpeg_sample_format(format);
context->context->time_base = av_make_q(1, context->context->sample_rate);
if ((ret = avcodec_open2(context->context, context->codec, NULL)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error avcodec_open2 %s [%d]", err, ret);
goto fail;
}
context->packet = av_packet_alloc();
if (!context->packet)
goto fail;
context->frame = av_frame_alloc();
if (!context->frame)
goto fail;
context->resampled = av_frame_alloc();
if (!context->resampled)
goto fail;
context->buffered = av_frame_alloc();
if (!context->buffered)
goto fail;
#if defined(SWRESAMPLE_FOUND)
context->rcontext = swr_alloc();
#else
context->rcontext = avresample_alloc_context();
#endif
if (!context->rcontext)
goto fail;
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
context->frame->channel_layout = layout;
context->frame->channels = format->nChannels;
#else
av_channel_layout_default(&context->frame->ch_layout, format->nChannels);
#endif
context->frame->sample_rate = format->nSamplesPerSec;
context->frame->format = AV_SAMPLE_FMT_S16;
if (context->encoder)
{
context->resampled->format = context->context->sample_fmt;
context->resampled->sample_rate = context->context->sample_rate;
}
else
{
context->resampled->format = AV_SAMPLE_FMT_S16;
context->resampled->sample_rate = format->nSamplesPerSec;
}
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
context->resampled->channel_layout = layout;
context->resampled->channels = format->nChannels;
#else
av_channel_layout_default(&context->resampled->ch_layout, format->nChannels);
#endif
if (context->context->frame_size > 0)
{
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
context->buffered->channel_layout = context->resampled->channel_layout;
context->buffered->channels = context->resampled->channels;
#else
av_channel_layout_copy(&context->buffered->ch_layout, &context->resampled->ch_layout);
#endif
context->buffered->format = context->resampled->format;
context->buffered->nb_samples = context->context->frame_size;
if ((ret = av_frame_get_buffer(context->buffered, 1)) < 0)
goto fail;
}
context->isOpen = TRUE;
return TRUE;
fail:
ffmpeg_close_context(context);
return FALSE;
}
#if defined(SWRESAMPLE_FOUND)
static BOOL ffmpeg_resample_frame(SwrContext* WINPR_RESTRICT context, AVFrame* WINPR_RESTRICT in,
AVFrame* WINPR_RESTRICT out)
{
int ret = 0;
if (!swr_is_initialized(context))
{
if ((ret = swr_config_frame(context, out, in)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
if ((ret = (swr_init(context))) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
}
if ((ret = swr_convert_frame(context, out, in)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
return TRUE;
}
#else
static BOOL ffmpeg_resample_frame(AVAudioResampleContext* WINPR_RESTRICT context,
AVFrame* WINPR_RESTRICT in, AVFrame* WINPR_RESTRICT out)
{
int ret;
if (!avresample_is_open(context))
{
if ((ret = avresample_config(context, out, in)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
if ((ret = (avresample_open(context))) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
}
if ((ret = avresample_convert_frame(context, out, in)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
return TRUE;
}
#endif
static BOOL ffmpeg_encode_frame(AVCodecContext* WINPR_RESTRICT context, AVFrame* WINPR_RESTRICT in,
AVPacket* WINPR_RESTRICT packet, wStream* WINPR_RESTRICT out)
{
if (in->format == AV_SAMPLE_FMT_FLTP)
{
uint8_t** pp = in->extended_data;
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
const int nr_channels = in->channels;
#else
const int nr_channels = in->ch_layout.nb_channels;
#endif
for (int y = 0; y < nr_channels; y++)
{
float* data = (float*)pp[y];
for (int x = 0; x < in->nb_samples; x++)
{
const float val1 = data[x];
if (isnan(val1))
data[x] = 0.0f;
else if (isinf(val1))
{
if (val1 < 0.0f)
data[x] = -1.0f;
else
data[x] = 1.0f;
}
}
}
}
/* send the packet with the compressed data to the encoder */
int ret = avcodec_send_frame(context, in);
if (ret < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error submitting the packet to the encoder %s [%d]", err, ret);
return FALSE;
}
/* read all the output frames (in general there may be any number of them */
while (ret >= 0)
{
ret = avcodec_receive_packet(context, packet);
if ((ret == AVERROR(EAGAIN)) || (ret == AVERROR_EOF))
return TRUE;
else if (ret < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during encoding %s [%d]", err, ret);
return FALSE;
}
if (!Stream_EnsureRemainingCapacity(out, packet->size))
return FALSE;
Stream_Write(out, packet->data, packet->size);
av_packet_unref(packet);
}
return TRUE;
}
static BOOL ffmpeg_fill_frame(AVFrame* WINPR_RESTRICT frame,
const AUDIO_FORMAT* WINPR_RESTRICT inputFormat,
const BYTE* WINPR_RESTRICT data, size_t size)
{
int ret = 0;
int bpp = 0;
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
frame->channels = inputFormat->nChannels;
frame->channel_layout = av_get_default_channel_layout(frame->channels);
#else
av_channel_layout_default(&frame->ch_layout, inputFormat->nChannels);
#endif
frame->sample_rate = inputFormat->nSamplesPerSec;
frame->format = ffmpeg_sample_format(inputFormat);
bpp = av_get_bytes_per_sample(frame->format);
frame->nb_samples = size / inputFormat->nChannels / bpp;
if ((ret = avcodec_fill_audio_frame(frame, inputFormat->nChannels, frame->format, data, size,
1)) < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during audio frame fill %s [%d]", err, ret);
return FALSE;
}
return TRUE;
}
#if defined(SWRESAMPLE_FOUND)
static BOOL ffmpeg_decode(AVCodecContext* WINPR_RESTRICT dec_ctx, AVPacket* WINPR_RESTRICT pkt,
AVFrame* WINPR_RESTRICT frame, SwrContext* WINPR_RESTRICT resampleContext,
AVFrame* WINPR_RESTRICT resampled, wStream* WINPR_RESTRICT out)
#else
static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt, AVFrame* frame,
AVAudioResampleContext* resampleContext, AVFrame* resampled, wStream* out)
#endif
{
int ret = 0;
/* send the packet with the compressed data to the decoder */
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error submitting the packet to the decoder %s [%d]", err, ret);
return FALSE;
}
/* read all the output frames (in general there may be any number of them */
while (ret >= 0)
{
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return TRUE;
else if (ret < 0)
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during decoding %s [%d]", err, ret);
return FALSE;
}
#if defined(SWRESAMPLE_FOUND)
if (!swr_is_initialized(resampleContext))
{
if ((ret = swr_config_frame(resampleContext, resampled, frame)) < 0)
{
#else
if (!avresample_is_open(resampleContext))
{
if ((ret = avresample_config(resampleContext, resampled, frame)) < 0)
{
#endif
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
#if defined(SWRESAMPLE_FOUND)
if ((ret = (swr_init(resampleContext))) < 0)
#else
if ((ret = (avresample_open(resampleContext))) < 0)
#endif
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
}
#if defined(SWRESAMPLE_FOUND)
if ((ret = swr_convert_frame(resampleContext, resampled, frame)) < 0)
#else
if ((ret = avresample_convert_frame(resampleContext, resampled, frame)) < 0)
#endif
{
const char* err = av_err2str(ret);
WLog_ERR(TAG, "Error during resampling %s [%d]", err, ret);
return FALSE;
}
{
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
const size_t channels = resampled->channels;
#else
const size_t channels = resampled->ch_layout.nb_channels;
#endif
const size_t data_size = channels * resampled->nb_samples * 2;
Stream_EnsureRemainingCapacity(out, data_size);
Stream_Write(out, resampled->data[0], data_size);
}
}
return TRUE;
}
BOOL freerdp_dsp_ffmpeg_supports_format(const AUDIO_FORMAT* WINPR_RESTRICT format, BOOL encode)
{
enum AVCodecID id = ffmpeg_get_avcodec(format);
if (ffmpeg_codec_is_filtered(id, encode))
return FALSE;
if (encode)
return avcodec_find_encoder(id) != NULL;
else
return avcodec_find_decoder(id) != NULL;
}
FREERDP_DSP_CONTEXT* freerdp_dsp_ffmpeg_context_new(BOOL encode)
{
FREERDP_DSP_CONTEXT* context = NULL;
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58, 10, 100)
avcodec_register_all();
#endif
context = calloc(1, sizeof(FREERDP_DSP_CONTEXT));
if (!context)
return NULL;
context->channelmix = Stream_New(NULL, 1024);
if (!context->channelmix)
{
WINPR_PRAGMA_DIAG_PUSH
WINPR_PRAGMA_DIAG_IGNORED_MISMATCHED_DEALLOC
freerdp_dsp_ffmpeg_context_free(context);
WINPR_PRAGMA_DIAG_POP
return NULL;
}
context->encoder = encode;
return context;
}
void freerdp_dsp_ffmpeg_context_free(FREERDP_DSP_CONTEXT* context)
{
if (context)
{
ffmpeg_close_context(context);
Stream_Free(context->channelmix, TRUE);
free(context);
}
}
BOOL freerdp_dsp_ffmpeg_context_reset(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context,
const AUDIO_FORMAT* WINPR_RESTRICT targetFormat)
{
if (!context || !targetFormat)
return FALSE;
ffmpeg_close_context(context);
context->format = *targetFormat;
return ffmpeg_open_context(context);
}
static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context,
const BYTE* WINPR_RESTRICT src, size_t size,
const AUDIO_FORMAT* WINPR_RESTRICT srcFormat,
const BYTE** WINPR_RESTRICT data, size_t* WINPR_RESTRICT length,
AUDIO_FORMAT* WINPR_RESTRICT dstFormat)
{
UINT32 bpp = 0;
size_t samples = 0;
if (!context || !data || !length || !dstFormat)
return FALSE;
if (srcFormat->wFormatTag != WAVE_FORMAT_PCM)
return FALSE;
bpp = srcFormat->wBitsPerSample > 8 ? 2 : 1;
samples = size / bpp / srcFormat->nChannels;
*dstFormat = *srcFormat;
if (context->format.nChannels == srcFormat->nChannels)
{
*data = src;
*length = size;
return TRUE;
}
Stream_SetPosition(context->channelmix, 0);
/* Destination has more channels than source */
if (context->format.nChannels > srcFormat->nChannels)
{
switch (srcFormat->nChannels)
{
case 1:
if (!Stream_EnsureCapacity(context->channelmix, size * 2))
return FALSE;
for (UINT32 x = 0; x < samples; x++)
{
for (UINT32 y = 0; y < bpp; y++)
Stream_Write_UINT8(context->channelmix, src[x * bpp + y]);
for (UINT32 y = 0; y < bpp; y++)
Stream_Write_UINT8(context->channelmix, src[x * bpp + y]);
}
Stream_SealLength(context->channelmix);
*data = Stream_Buffer(context->channelmix);
*length = Stream_Length(context->channelmix);
dstFormat->nChannels = 2;
return TRUE;
case 2: /* We only support stereo, so we can not handle this case. */
default: /* Unsupported number of channels */
WLog_WARN(TAG, "unsupported source channel count %" PRIu16, srcFormat->nChannels);
return FALSE;
}
}
/* Destination has less channels than source */
switch (srcFormat->nChannels)
{
case 2:
if (!Stream_EnsureCapacity(context->channelmix, size / 2))
return FALSE;
/* Simply drop second channel.
* TODO: Calculate average */
for (UINT32 x = 0; x < samples; x++)
{
for (UINT32 y = 0; y < bpp; y++)
Stream_Write_UINT8(context->channelmix, src[2 * x * bpp + y]);
}
Stream_SealLength(context->channelmix);
*data = Stream_Buffer(context->channelmix);
*length = Stream_Length(context->channelmix);
dstFormat->nChannels = 1;
return TRUE;
case 1: /* Invalid, do we want to use a 0 channel sound? */
default: /* Unsupported number of channels */
WLog_WARN(TAG, "unsupported channel count %" PRIu16, srcFormat->nChannels);
return FALSE;
}
return FALSE;
}
BOOL freerdp_dsp_ffmpeg_encode(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context,
const AUDIO_FORMAT* WINPR_RESTRICT format,
const BYTE* WINPR_RESTRICT data, size_t length,
wStream* WINPR_RESTRICT out)
{
AUDIO_FORMAT fmt = { 0 };
if (!context || !format || !data || !out || !context->encoder)
return FALSE;
if (!context || !data || !out)
return FALSE;
/* https://github.com/FreeRDP/FreeRDP/issues/7607
*
* we get noisy data with channel transformation, so do it ourselves.
*/
if (!freerdp_dsp_channel_mix(context, data, length, format, &data, &length, &fmt))
return FALSE;
/* Create input frame */
if (!ffmpeg_fill_frame(context->frame, format, data, length))
return FALSE;
/* Resample to desired format. */
if (!ffmpeg_resample_frame(context->rcontext, context->frame, context->resampled))
return FALSE;
if (context->context->frame_size <= 0)
{
return ffmpeg_encode_frame(context->context, context->resampled, context->packet, out);
}
else
{
int copied = 0;
int rest = context->resampled->nb_samples;
do
{
int inSamples = rest;
if ((inSamples < 0) || (context->bufferedSamples > (UINT32)(INT_MAX - inSamples)))
return FALSE;
if (inSamples + (int)context->bufferedSamples > context->context->frame_size)
inSamples = context->context->frame_size - (int)context->bufferedSamples;
#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
const int channels = context->context->channels;
#else
const int channels = context->context->ch_layout.nb_channels;
#endif
const int rc =
av_samples_copy(context->buffered->extended_data, context->resampled->extended_data,
(int)context->bufferedSamples, copied, inSamples, channels,
context->context->sample_fmt);
if (rc < 0)
return FALSE;
rest -= inSamples;
copied += inSamples;
context->bufferedSamples += (UINT32)inSamples;
if (context->context->frame_size <= (int)context->bufferedSamples)
{
/* Encode in desired format. */
if (!ffmpeg_encode_frame(context->context, context->buffered, context->packet, out))
return FALSE;
context->bufferedSamples = 0;
}
} while (rest > 0);
return TRUE;
}
}
BOOL freerdp_dsp_ffmpeg_decode(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context,
const AUDIO_FORMAT* WINPR_RESTRICT srcFormat,
const BYTE* WINPR_RESTRICT data, size_t length,
wStream* WINPR_RESTRICT out)
{
if (!context || !srcFormat || !data || !out || context->encoder)
return FALSE;
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58, 133, 100)
av_init_packet(context->packet);
#endif
context->packet->data = (uint8_t*)data;
context->packet->size = length;
return ffmpeg_decode(context->context, context->packet, context->frame, context->rcontext,
context->resampled, out);
}