[codec,dsp] revert FFMPEG related changes
some changes were not intended, revert these.
This commit is contained in:
Armin Novak
parent
a632e4e9db
commit
3ca781075f
@ -90,15 +90,11 @@ static BOOL ffmpeg_codec_is_filtered(enum AVCodecID id, BOOL encoder)
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}
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}
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static enum AVCodecID ffmpeg_get_avcodec(const AUDIO_FORMAT* format)
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static enum AVCodecID ffmpeg_get_avcodec(const AUDIO_FORMAT* WINPR_RESTRICT format)
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{
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const char* id;
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if (!format)
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return AV_CODEC_ID_NONE;
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id = audio_format_get_tag_string(format->wFormatTag);
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switch (format->wFormatTag)
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{
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case WAVE_FORMAT_UNKNOWN:
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@ -138,12 +134,15 @@ static enum AVCodecID ffmpeg_get_avcodec(const AUDIO_FORMAT* format)
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case WAVE_FORMAT_AAC_MS:
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return AV_CODEC_ID_AAC;
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case WAVE_FORMAT_OPUS:
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return AV_CODEC_ID_OPUS;
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default:
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return AV_CODEC_ID_NONE;
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}
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}
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static int ffmpeg_sample_format(const AUDIO_FORMAT* format)
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static int ffmpeg_sample_format(const AUDIO_FORMAT* WINPR_RESTRICT format)
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{
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switch (format->wFormatTag)
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{
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@ -168,6 +167,9 @@ static int ffmpeg_sample_format(const AUDIO_FORMAT* format)
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case WAVE_FORMAT_AAC_MS:
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return AV_SAMPLE_FMT_FLTP;
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case WAVE_FORMAT_OPUS:
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return AV_SAMPLE_FMT_S16;
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case WAVE_FORMAT_MSG723:
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case WAVE_FORMAT_GSM610:
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return AV_SAMPLE_FMT_S16P;
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@ -180,7 +182,7 @@ static int ffmpeg_sample_format(const AUDIO_FORMAT* format)
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}
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}
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static void ffmpeg_close_context(FREERDP_DSP_CONTEXT* context)
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static void ffmpeg_close_context(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context)
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{
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if (context)
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{
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@ -219,19 +221,17 @@ static void ffmpeg_close_context(FREERDP_DSP_CONTEXT* context)
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}
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}
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static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* context)
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static BOOL ffmpeg_open_context(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context)
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{
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int ret;
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const AUDIO_FORMAT* format;
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int ret = 0;
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if (!context || context->isOpen)
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return FALSE;
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format = &context->common.format;
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const AUDIO_FORMAT* format = &context->common.format;
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if (!format)
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return FALSE;
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context->id = ffmpeg_get_avcodec(format);
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if (ffmpeg_codec_is_filtered(context->id, context->common.encoder))
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@ -369,9 +369,10 @@ fail:
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}
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#if defined(SWRESAMPLE_FOUND)
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static BOOL ffmpeg_resample_frame(SwrContext* context, AVFrame* in, AVFrame* out)
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static BOOL ffmpeg_resample_frame(SwrContext* WINPR_RESTRICT context, AVFrame* WINPR_RESTRICT in,
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AVFrame* WINPR_RESTRICT out)
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{
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int ret;
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int ret = 0;
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if (!swr_is_initialized(context))
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{
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@ -400,7 +401,8 @@ static BOOL ffmpeg_resample_frame(SwrContext* context, AVFrame* in, AVFrame* out
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return TRUE;
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}
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#else
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static BOOL ffmpeg_resample_frame(AVAudioResampleContext* context, AVFrame* in, AVFrame* out)
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static BOOL ffmpeg_resample_frame(AVAudioResampleContext* WINPR_RESTRICT context,
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AVFrame* WINPR_RESTRICT in, AVFrame* WINPR_RESTRICT out)
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{
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int ret;
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@ -432,12 +434,38 @@ static BOOL ffmpeg_resample_frame(AVAudioResampleContext* context, AVFrame* in,
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}
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#endif
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static BOOL ffmpeg_encode_frame(AVCodecContext* context, AVFrame* in, AVPacket* packet,
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wStream* out)
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static BOOL ffmpeg_encode_frame(AVCodecContext* WINPR_RESTRICT context, AVFrame* WINPR_RESTRICT in,
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AVPacket* WINPR_RESTRICT packet, wStream* WINPR_RESTRICT out)
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{
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int ret;
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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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#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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for (int y = 0; y < nr_channels; y++)
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{
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float* data = (float*)pp[y];
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for (int x = 0; x < in->nb_samples; x++)
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{
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const float val1 = data[x];
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if (isnan(val1))
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data[x] = 0.0f;
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else if (isinf(val1))
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{
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if (val1 < 0.0f)
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data[x] = -1.0f;
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else
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data[x] = 1.0f;
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}
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}
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}
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}
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/* send the packet with the compressed data to the encoder */
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ret = avcodec_send_frame(context, in);
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int ret = avcodec_send_frame(context, in);
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if (ret < 0)
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{
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@ -470,16 +498,17 @@ static BOOL ffmpeg_encode_frame(AVCodecContext* context, AVFrame* in, AVPacket*
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return TRUE;
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}
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static BOOL ffmpeg_fill_frame(AVFrame* frame, const AUDIO_FORMAT* inputFormat, const BYTE* data,
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size_t size)
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static BOOL ffmpeg_fill_frame(AVFrame* WINPR_RESTRICT frame,
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const AUDIO_FORMAT* WINPR_RESTRICT inputFormat,
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const BYTE* WINPR_RESTRICT data, 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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av_channel_layout_default(&frame->ch_layout, inputFormat->nChannels);
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#else
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int ret = 0;
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int bpp = 0;
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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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@ -498,14 +527,15 @@ static BOOL ffmpeg_fill_frame(AVFrame* frame, const AUDIO_FORMAT* inputFormat, c
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return TRUE;
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}
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#if defined(SWRESAMPLE_FOUND)
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static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt, AVFrame* frame,
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SwrContext* resampleContext, AVFrame* resampled, wStream* out)
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static BOOL ffmpeg_decode(AVCodecContext* WINPR_RESTRICT dec_ctx, AVPacket* WINPR_RESTRICT pkt,
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AVFrame* WINPR_RESTRICT frame, SwrContext* WINPR_RESTRICT resampleContext,
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AVFrame* WINPR_RESTRICT resampled, wStream* WINPR_RESTRICT out)
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#else
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static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt, AVFrame* frame,
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AVAudioResampleContext* resampleContext, AVFrame* resampled, wStream* out)
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#endif
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{
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int ret;
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int ret = 0;
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/* send the packet with the compressed data to the decoder */
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ret = avcodec_send_packet(dec_ctx, pkt);
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@ -586,7 +616,7 @@ static BOOL ffmpeg_decode(AVCodecContext* dec_ctx, AVPacket* pkt, AVFrame* frame
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return TRUE;
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}
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BOOL freerdp_dsp_ffmpeg_supports_format(const AUDIO_FORMAT* format, BOOL encode)
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BOOL freerdp_dsp_ffmpeg_supports_format(const AUDIO_FORMAT* WINPR_RESTRICT format, BOOL encode)
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{
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enum AVCodecID id = ffmpeg_get_avcodec(format);
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@ -601,7 +631,7 @@ BOOL freerdp_dsp_ffmpeg_supports_format(const AUDIO_FORMAT* format, BOOL encode)
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FREERDP_DSP_CONTEXT* freerdp_dsp_ffmpeg_context_new(BOOL encode)
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{
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FREERDP_DSP_CONTEXT* context;
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FREERDP_DSP_CONTEXT* context = NULL;
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58, 10, 100)
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avcodec_register_all();
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#endif
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@ -630,8 +660,8 @@ void freerdp_dsp_ffmpeg_context_free(FREERDP_DSP_CONTEXT* context)
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}
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}
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BOOL freerdp_dsp_ffmpeg_context_reset(FREERDP_DSP_CONTEXT* context,
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const AUDIO_FORMAT* targetFormat)
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BOOL freerdp_dsp_ffmpeg_context_reset(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context,
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const AUDIO_FORMAT* WINPR_RESTRICT targetFormat)
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{
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if (!context || !targetFormat)
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return FALSE;
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@ -641,13 +671,14 @@ BOOL freerdp_dsp_ffmpeg_context_reset(FREERDP_DSP_CONTEXT* context,
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return ffmpeg_open_context(context);
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}
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static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* src, size_t size,
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const AUDIO_FORMAT* srcFormat, const BYTE** data,
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size_t* length, AUDIO_FORMAT* dstFormat)
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static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context,
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const BYTE* WINPR_RESTRICT src, size_t size,
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const AUDIO_FORMAT* WINPR_RESTRICT srcFormat,
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const BYTE** WINPR_RESTRICT data, size_t* WINPR_RESTRICT length,
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AUDIO_FORMAT* WINPR_RESTRICT dstFormat)
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{
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UINT32 bpp;
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size_t samples;
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size_t x, y;
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UINT32 bpp = 0;
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size_t samples = 0;
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if (!context || !data || !length || !dstFormat)
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return FALSE;
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@ -677,12 +708,12 @@ static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* sr
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if (!Stream_EnsureCapacity(context->common.channelmix, size * 2))
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return FALSE;
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for (x = 0; x < samples; x++)
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for (size_t x = 0; x < samples; x++)
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{
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for (y = 0; y < bpp; y++)
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for (size_t y = 0; y < bpp; y++)
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Stream_Write_UINT8(context->common.channelmix, src[x * bpp + y]);
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for (y = 0; y < bpp; y++)
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for (size_t y = 0; y < bpp; y++)
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Stream_Write_UINT8(context->common.channelmix, src[x * bpp + y]);
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}
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@ -709,9 +740,9 @@ static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* sr
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/* Simply drop second channel.
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* TODO: Calculate average */
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for (x = 0; x < samples; x++)
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for (size_t x = 0; x < samples; x++)
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{
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for (y = 0; y < bpp; y++)
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for (size_t y = 0; y < bpp; y++)
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Stream_Write_UINT8(context->common.channelmix, src[2 * x * bpp + y]);
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}
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@ -731,10 +762,11 @@ static BOOL freerdp_dsp_channel_mix(FREERDP_DSP_CONTEXT* context, const BYTE* sr
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return FALSE;
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}
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BOOL freerdp_dsp_ffmpeg_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* format,
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const BYTE* data, size_t length, wStream* out)
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BOOL freerdp_dsp_ffmpeg_encode(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context,
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const AUDIO_FORMAT* WINPR_RESTRICT format,
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const BYTE* WINPR_RESTRICT data, size_t length,
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wStream* WINPR_RESTRICT out)
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{
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int rc;
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AUDIO_FORMAT fmt = { 0 };
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if (!context || !format || !data || !out || !context->common.encoder)
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@ -782,10 +814,12 @@ BOOL freerdp_dsp_ffmpeg_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT*
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#else
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const size_t nrchannels = context->context->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, nrchannels, context->context->sample_fmt);
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const int 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, nrchannels,
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context->context->sample_fmt);
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if (rc < 0)
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return FALSE;
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rest -= inSamples;
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copied += inSamples;
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context->bufferedSamples += (UINT32)inSamples;
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@ -804,8 +838,10 @@ BOOL freerdp_dsp_ffmpeg_encode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT*
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}
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}
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BOOL freerdp_dsp_ffmpeg_decode(FREERDP_DSP_CONTEXT* context, const AUDIO_FORMAT* srcFormat,
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const BYTE* data, size_t length, wStream* out)
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BOOL freerdp_dsp_ffmpeg_decode(FREERDP_DSP_CONTEXT* WINPR_RESTRICT context,
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const AUDIO_FORMAT* WINPR_RESTRICT srcFormat,
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const BYTE* WINPR_RESTRICT data, size_t length,
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wStream* WINPR_RESTRICT out)
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{
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if (!context || !srcFormat || !data || !out || context->common.encoder)
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return FALSE;
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