* update freerdp2 to 2.11.5-3
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28
freerdp2/1ef7b9e3.patch
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28
freerdp2/1ef7b9e3.patch
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From 1ef7b9e3e06ef69e5145c6f11cc330078abaa9ea Mon Sep 17 00:00:00 2001
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From: Armin Novak <anovak@thincast.com>
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Date: Fri, 24 Nov 2023 19:40:52 +0100
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Subject: [PATCH] [codec,dsp] fix ffmpeg deprecation warning
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---
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libfreerdp/codec/dsp_ffmpeg.c | 8 +++++++-
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1 file changed, 7 insertions(+), 1 deletion(-)
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diff --git a/libfreerdp/codec/dsp_ffmpeg.c b/libfreerdp/codec/dsp_ffmpeg.c
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index c5c30fb89ed2..c929b3c553a2 100644
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--- a/libfreerdp/codec/dsp_ffmpeg.c
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+++ b/libfreerdp/codec/dsp_ffmpeg.c
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@@ -439,7 +439,13 @@ static BOOL ffmpeg_encode_frame(AVCodecContext* context, AVFrame* in, AVPacket*
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if (in->format == AV_SAMPLE_FMT_FLTP)
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{
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uint8_t** pp = in->extended_data;
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- for (int y = 0; y < in->channels; y++)
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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+ const int nr_channels = in->channels;
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+#else
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+ const int nr_channels = in->ch_layout.nb_channels;
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+#endif
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+
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+ for (int y = 0; y < nr_channels; y++)
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{
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float* data = pp[y];
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for (int x = 0; x < in->nb_samples; x++)
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140
freerdp2/d0c5b1ae.patch
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140
freerdp2/d0c5b1ae.patch
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From d0c5b1ae4289c7f3cde3fbc031cb4a3160df05ff Mon Sep 17 00:00:00 2001
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From: Armin Novak <armin.novak@thincast.com>
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Date: Wed, 7 Jun 2023 11:46:07 +0200
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Subject: [PATCH] [codec,dsp] fix ffmpeg deprecations
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---
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libfreerdp/codec/dsp_ffmpeg.c | 54 +++++++++++++++++++++++++++--------
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1 file changed, 42 insertions(+), 12 deletions(-)
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diff --git a/libfreerdp/codec/dsp_ffmpeg.c b/libfreerdp/codec/dsp_ffmpeg.c
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index 2c93a667750e..ebba52b147d2 100644
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--- a/libfreerdp/codec/dsp_ffmpeg.c
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+++ b/libfreerdp/codec/dsp_ffmpeg.c
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@@ -224,18 +224,17 @@ static void ffmpeg_close_context(FREERDP_DSP_CONTEXT* context)
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static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
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{
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int ret;
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- int layout;
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- const AUDIO_FORMAT* format;
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if (!context || context->isOpen)
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return FALSE;
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- format = &context->format;
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+ const AUDIO_FORMAT* format = &context->format;
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if (!format)
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return FALSE;
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-
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- layout = av_get_default_channel_layout(format->nChannels);
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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+ const int layout = av_get_default_channel_layout(format->nChannels);
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+#endif
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context->id = ffmpeg_get_avcodec(format);
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if (ffmpeg_codec_is_filtered(context->id, context->encoder))
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@@ -271,8 +270,12 @@ static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
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break;
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}
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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context->context->channels = format->nChannels;
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context->context->channel_layout = layout;
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+#else
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+ av_channel_layout_default(&context->context->ch_layout, format->nChannels);
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+#endif
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context->context->sample_rate = format->nSamplesPerSec;
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context->context->block_align = format->nBlockAlign;
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context->context->bit_rate = format->nAvgBytesPerSec * 8;
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@@ -315,8 +318,12 @@ static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
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if (!context->rcontext)
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goto fail;
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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context->frame->channel_layout = layout;
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context->frame->channels = format->nChannels;
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+#else
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+ av_channel_layout_default(&context->frame->ch_layout, format->nChannels);
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+#endif
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context->frame->sample_rate = format->nSamplesPerSec;
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context->frame->format = AV_SAMPLE_FMT_S16;
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@@ -331,13 +338,21 @@ static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
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context->resampled->sample_rate = format->nSamplesPerSec;
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}
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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context->resampled->channel_layout = layout;
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context->resampled->channels = format->nChannels;
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+#else
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+ av_channel_layout_default(&context->resampled->ch_layout, format->nChannels);
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+#endif
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if (context->context->frame_size > 0)
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{
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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context->buffered->channel_layout = context->resampled->channel_layout;
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context->buffered->channels = context->resampled->channels;
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+#else
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+ av_channel_layout_copy(&context->buffered->ch_layout, &context->resampled->ch_layout);
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+#endif
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context->buffered->format = context->resampled->format;
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context->buffered->nb_samples = context->context->frame_size;
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@@ -458,14 +473,20 @@ static BOOL ffmpeg_fill_frame(AVFrame* frame, const AUDIO_FORMAT* inputFormat, c
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size_t size)
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{
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int ret, bpp;
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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frame->channels = inputFormat->nChannels;
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+ frame->channel_layout = av_get_default_channel_layout(frame->channels);
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+#else
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+ av_channel_layout_default(&frame->ch_layout, inputFormat->nChannels);
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+#endif
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frame->sample_rate = inputFormat->nSamplesPerSec;
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frame->format = ffmpeg_sample_format(inputFormat);
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- frame->channel_layout = av_get_default_channel_layout(frame->channels);
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+
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bpp = av_get_bytes_per_sample(frame->format);
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frame->nb_samples = size / inputFormat->nChannels / bpp;
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- if ((ret = avcodec_fill_audio_frame(frame, frame->channels, frame->format, data, size, 1)) < 0)
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+ if ((ret = avcodec_fill_audio_frame(frame, inputFormat->nChannels, frame->format, data, size,
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+ 1)) < 0)
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{
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const char* err = av_err2str(ret);
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WLog_ERR(TAG, "Error during audio frame fill %s [%d]", err, ret);
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@@ -547,7 +568,12 @@ static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt, AVFrame* frame
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}
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{
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- const size_t data_size = resampled->channels * resampled->nb_samples * 2;
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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+ const size_t channels = resampled->channels;
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+#else
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+ const size_t channels = resampled->ch_layout.nb_channels;
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+#endif
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+ const size_t data_size = channels * resampled->nb_samples * 2;
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Stream_EnsureRemainingCapacity(out, data_size);
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Stream_Write(out, resampled->data[0], data_size);
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}
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@@ -745,10 +771,14 @@ BOOL freerdp_dsp_ffmpeg_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT*
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if (inSamples + (int)context->bufferedSamples > context->context->frame_size)
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inSamples = context->context->frame_size - (int)context->bufferedSamples;
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- rc =
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- av_samples_copy(context->buffered->extended_data, context->resampled->extended_data,
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- (int)context->bufferedSamples, copied, inSamples,
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- context->context->channels, context->context->sample_fmt);
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+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(57, 28, 100)
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+ const int channels = context->context->channels;
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+#else
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+ const int channels = context->context->ch_layout.nb_channels;
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+#endif
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+ rc = av_samples_copy(context->buffered->extended_data,
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+ context->resampled->extended_data, (int)context->bufferedSamples,
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+ copied, inSamples, channels, context->context->sample_fmt);
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rest -= inSamples;
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copied += inSamples;
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context->bufferedSamples += (UINT32)inSamples;
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