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[codec,dsp] fix ffmpeg deprecations

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Armin Novak 2023-06-07 11:46:07 +02:00 committed by David Fort
parent 81e95e51ca
commit d0c5b1ae42
1 changed files with 42 additions and 12 deletions
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54
libfreerdp/codec/dsp_ffmpeg.c
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@ -224,18 +224,17 @@ static void ffmpeg_close_context(FREERDP_DSP_CONTEXT* context)
static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
{
int ret;
int layout;
const AUDIO_FORMAT* format;
if (!context || context->isOpen)
return FALSE;
format = &context->format;
const AUDIO_FORMAT* format = &context->format;
if (!format)
return FALSE;
layout = av_get_default_channel_layout(format->nChannels);
#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))
@ -271,8 +270,12 @@ static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
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;
@ -315,8 +318,12 @@ static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
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;
@ -331,13 +338,21 @@ static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
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;
@ -458,14 +473,20 @@ static BOOL ffmpeg_fill_frame(AVFrame* frame, const AUDIO_FORMAT* inputFormat, c
size_t size)
{
int ret, bpp;
#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);
frame->channel_layout = av_get_default_channel_layout(frame->channels);
bpp = av_get_bytes_per_sample(frame->format);
frame->nb_samples = size / inputFormat->nChannels / bpp;
if ((ret = avcodec_fill_audio_frame(frame, frame->channels, frame->format, data, size, 1)) < 0)
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);
@ -547,7 +568,12 @@ static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt, AVFrame* frame
}
{
const size_t data_size = resampled->channels * resampled->nb_samples * 2;
#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);
}
@ -745,10 +771,14 @@ BOOL freerdp_dsp_ffmpeg_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT*
if (inSamples + (int)context->bufferedSamples > context->context->frame_size)
inSamples = context->context->frame_size - (int)context->bufferedSamples;
rc =
av_samples_copy(context->buffered->extended_data, context->resampled->extended_data,
(int)context->bufferedSamples, copied, inSamples,
context->context->channels, context->context->sample_fmt);
#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
rc = av_samples_copy(context->buffered->extended_data,
context->resampled->extended_data, (int)context->bufferedSamples,
copied, inSamples, channels, context->context->sample_fmt);
rest -= inSamples;
copied += inSamples;
context->bufferedSamples += (UINT32)inSamples;