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https://github.com/FFmpeg/FFmpeg.git
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avcodec/dca: move some enumeration typedefs into headers
These values will be used by the parser. Prefix them with DCA_ appropriately. Signed-off-by: James Almer <jamrial@gmail.com>
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@ -35,35 +35,7 @@ enum HeaderType {
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HEADER_XXCH
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};
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enum AudioMode {
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AMODE_MONO, // Mode 0: A (mono)
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AMODE_MONO_DUAL, // Mode 1: A + B (dual mono)
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AMODE_STEREO, // Mode 2: L + R (stereo)
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AMODE_STEREO_SUMDIFF, // Mode 3: (L+R) + (L-R) (sum-diff)
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AMODE_STEREO_TOTAL, // Mode 4: LT + RT (left and right total)
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AMODE_3F, // Mode 5: C + L + R
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AMODE_2F1R, // Mode 6: L + R + S
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AMODE_3F1R, // Mode 7: C + L + R + S
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AMODE_2F2R, // Mode 8: L + R + SL + SR
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AMODE_3F2R, // Mode 9: C + L + R + SL + SR
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AMODE_COUNT
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};
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enum ExtAudioType {
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EXT_AUDIO_XCH = 0,
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EXT_AUDIO_X96 = 2,
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EXT_AUDIO_XXCH = 6
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};
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enum LFEFlag {
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LFE_FLAG_NONE,
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LFE_FLAG_128,
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LFE_FLAG_64,
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LFE_FLAG_INVALID
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};
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static const int8_t prm_ch_to_spkr_map[AMODE_COUNT][5] = {
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static const int8_t prm_ch_to_spkr_map[DCA_AMODE_COUNT][5] = {
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{ DCA_SPEAKER_C, -1, -1, -1, -1 },
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{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
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{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
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@ -76,7 +48,7 @@ static const int8_t prm_ch_to_spkr_map[AMODE_COUNT][5] = {
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{ DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs }
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};
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static const uint8_t audio_mode_ch_mask[AMODE_COUNT] = {
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static const uint8_t audio_mode_ch_mask[DCA_AMODE_COUNT] = {
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DCA_SPEAKER_LAYOUT_MONO,
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DCA_SPEAKER_LAYOUT_STEREO,
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DCA_SPEAKER_LAYOUT_STEREO,
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@ -139,7 +111,7 @@ static int parse_frame_header(DCACoreDecoder *s)
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// Audio channel arrangement
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s->audio_mode = get_bits(&s->gb, 6);
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if (s->audio_mode >= AMODE_COUNT) {
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if (s->audio_mode >= DCA_AMODE_COUNT) {
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av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", s->audio_mode);
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return AVERROR_PATCHWELCOME;
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}
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@ -180,7 +152,7 @@ static int parse_frame_header(DCACoreDecoder *s)
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// Low frequency effects flag
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s->lfe_present = get_bits(&s->gb, 2);
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if (s->lfe_present == LFE_FLAG_INVALID) {
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if (s->lfe_present == DCA_LFE_FLAG_INVALID) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid low frequency effects flag\n");
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return AVERROR_INVALIDDATA;
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}
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@ -1783,7 +1755,7 @@ static int parse_optional_info(DCACoreDecoder *s)
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// must be done backwards from the end of core frame to work around
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// sync word aliasing issues.
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switch (s->ext_audio_type) {
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case EXT_AUDIO_XCH:
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case DCA_EXT_AUDIO_XCH:
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if (dca->request_channel_layout)
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break;
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@ -1813,7 +1785,7 @@ static int parse_optional_info(DCACoreDecoder *s)
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}
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break;
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case EXT_AUDIO_X96:
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case DCA_EXT_AUDIO_X96:
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// The distance between X96 sync word and end of the core frame
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// must be equal to X96 frame size. Minimum X96 frame size is 96
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// bytes.
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@ -1836,7 +1808,7 @@ static int parse_optional_info(DCACoreDecoder *s)
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}
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break;
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case EXT_AUDIO_XXCH:
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case DCA_EXT_AUDIO_XXCH:
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if (dca->request_channel_layout)
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break;
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@ -2102,7 +2074,7 @@ int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth)
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int nlfesamples = s->npcmblocks >> 1;
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// Check LFF
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if (s->lfe_present == LFE_FLAG_128) {
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if (s->lfe_present == DCA_LFE_FLAG_128) {
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av_log(s->avctx, AV_LOG_ERROR, "Fixed point mode doesn't support LFF=1\n");
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return AVERROR(EINVAL);
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}
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@ -2152,7 +2124,7 @@ static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame)
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// Undo embedded XCH downmix
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if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
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&& s->audio_mode >= AMODE_2F2R) {
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&& s->audio_mode >= DCA_AMODE_2F2R) {
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s->dcadsp->dmix_sub_xch(s->output_samples[DCA_SPEAKER_Ls],
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s->output_samples[DCA_SPEAKER_Rs],
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s->output_samples[DCA_SPEAKER_Cs],
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@ -2196,15 +2168,15 @@ static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame)
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if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
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// Front sum/difference decoding
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if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
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|| s->audio_mode == AMODE_STEREO_SUMDIFF) {
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if ((s->sumdiff_front && s->audio_mode > DCA_AMODE_MONO)
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|| s->audio_mode == DCA_AMODE_STEREO_SUMDIFF) {
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s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_L],
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s->output_samples[DCA_SPEAKER_R],
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nsamples);
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}
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// Surround sum/difference decoding
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if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
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if (s->sumdiff_surround && s->audio_mode >= DCA_AMODE_2F2R) {
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s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_Ls],
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s->output_samples[DCA_SPEAKER_Rs],
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nsamples);
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@ -2308,7 +2280,7 @@ static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame)
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// Filter LFE channel
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if (s->lfe_present) {
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int dec_select = (s->lfe_present == LFE_FLAG_128);
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int dec_select = (s->lfe_present == DCA_LFE_FLAG_128);
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float *samples = output_samples[DCA_SPEAKER_LFE1];
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int nlfesamples = s->npcmblocks >> (dec_select + 1);
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@ -2342,7 +2314,7 @@ static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame)
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// Undo embedded XCH downmix
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if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
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&& s->audio_mode >= AMODE_2F2R) {
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&& s->audio_mode >= DCA_AMODE_2F2R) {
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s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Ls],
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output_samples[DCA_SPEAKER_Cs],
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-M_SQRT1_2, nsamples);
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@ -2389,15 +2361,15 @@ static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame)
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if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
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// Front sum/difference decoding
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if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
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|| s->audio_mode == AMODE_STEREO_SUMDIFF) {
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if ((s->sumdiff_front && s->audio_mode > DCA_AMODE_MONO)
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|| s->audio_mode == DCA_AMODE_STEREO_SUMDIFF) {
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s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_L],
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output_samples[DCA_SPEAKER_R],
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nsamples);
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}
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// Surround sum/difference decoding
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if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
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if (s->sumdiff_surround && s->audio_mode >= DCA_AMODE_2F2R) {
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s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_Ls],
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output_samples[DCA_SPEAKER_Rs],
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nsamples);
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@ -2424,7 +2396,7 @@ int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame)
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// Handle downmixing to stereo request
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if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO
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&& s->audio_mode > AMODE_MONO && s->prim_dmix_embedded
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&& s->audio_mode > DCA_AMODE_MONO && s->prim_dmix_embedded
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&& (s->prim_dmix_type == DCA_DMIX_TYPE_LoRo ||
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s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
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s->request_mask = DCA_SPEAKER_LAYOUT_STEREO;
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@ -2457,8 +2429,8 @@ int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame)
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else
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avctx->bit_rate = 0;
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if (s->audio_mode == AMODE_STEREO_TOTAL || (s->request_mask != s->ch_mask &&
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s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
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if (s->audio_mode == DCA_AMODE_STEREO_TOTAL || (s->request_mask != s->ch_mask &&
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s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
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matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
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else
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matrix_encoding = AV_MATRIX_ENCODING_NONE;
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@ -56,6 +56,34 @@
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#define DCA_FILTER_MODE_X96 0x01
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#define DCA_FILTER_MODE_FIXED 0x02
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enum DCACoreAudioMode {
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DCA_AMODE_MONO, // Mode 0: A (mono)
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DCA_AMODE_MONO_DUAL, // Mode 1: A + B (dual mono)
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DCA_AMODE_STEREO, // Mode 2: L + R (stereo)
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DCA_AMODE_STEREO_SUMDIFF, // Mode 3: (L+R) + (L-R) (sum-diff)
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DCA_AMODE_STEREO_TOTAL, // Mode 4: LT + RT (left and right total)
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DCA_AMODE_3F, // Mode 5: C + L + R
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DCA_AMODE_2F1R, // Mode 6: L + R + S
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DCA_AMODE_3F1R, // Mode 7: C + L + R + S
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DCA_AMODE_2F2R, // Mode 8: L + R + SL + SR
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DCA_AMODE_3F2R, // Mode 9: C + L + R + SL + SR
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DCA_AMODE_COUNT
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};
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enum DCACoreExtAudioType {
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DCA_EXT_AUDIO_XCH = 0,
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DCA_EXT_AUDIO_X96 = 2,
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DCA_EXT_AUDIO_XXCH = 6
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};
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enum DCACoreLFEFlag {
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DCA_LFE_FLAG_NONE,
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DCA_LFE_FLAG_128,
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DCA_LFE_FLAG_64,
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DCA_LFE_FLAG_INVALID
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};
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typedef struct DCADSPData {
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union {
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struct {
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@ -30,11 +30,6 @@
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#define AMP_MAX 56
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enum LBRHeader {
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LBR_HEADER_SYNC_ONLY = 1,
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LBR_HEADER_DECODER_INIT = 2
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};
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enum LBRFlags {
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LBR_FLAG_24_BIT = 0x01,
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LBR_FLAG_LFE_PRESENT = 0x02,
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@ -1171,13 +1166,13 @@ int ff_dca_lbr_parse(DCALbrDecoder *s, uint8_t *data, DCAExssAsset *asset)
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// LBR header type
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switch (bytestream2_get_byte(&gb)) {
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case LBR_HEADER_SYNC_ONLY:
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case DCA_LBR_HEADER_SYNC_ONLY:
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if (!s->sample_rate) {
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av_log(s->avctx, AV_LOG_ERROR, "LBR decoder not initialized\n");
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return AVERROR_INVALIDDATA;
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}
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break;
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case LBR_HEADER_DECODER_INIT:
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case DCA_LBR_HEADER_DECODER_INIT:
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if ((ret = parse_decoder_init(s, &gb)) < 0) {
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s->sample_rate = 0;
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return ret;
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@ -41,6 +41,11 @@
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#define DCA_LBR_TIME_SAMPLES 128
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#define DCA_LBR_TIME_HISTORY 8
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enum DCALBRHeader {
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DCA_LBR_HEADER_SYNC_ONLY = 1,
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DCA_LBR_HEADER_DECODER_INIT = 2
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};
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typedef struct DCALbrTone {
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uint8_t x_freq; ///< Spectral line offset
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uint8_t f_delt; ///< Difference between original and center frequency
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