| Message ID | 20220318130417.47935-1-onemda@gmail.com |
|---|---|
| State | New |
| Headers | show |
| Series | [FFmpeg-devel] avcodec/binkaudio: add support for >2 channels dct codec | expand |
| Context | Check | Description |
|---|---|---|
| yinshiyou/make_loongarch64 | success | Make finished |
| yinshiyou/make_fate_loongarch64 | success | Make fate finished |
| andriy/make_armv7_RPi4 | success | Make finished |
| andriy/make_fate_armv7_RPi4 | success | Make fate finished |
| andriy/make_aarch64_jetson | success | Make finished |
| andriy/make_fate_aarch64_jetson | success | Make fate finished |
Paul B Mahol: > As presented in .binka files. > > Signed-off-by: Paul B Mahol <onemda@gmail.com> > --- > libavcodec/binkaudio.c | 50 +++++++++++++++++++++++++++--------------- > 1 file changed, 32 insertions(+), 18 deletions(-) > > diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c > index b4ff15beeb..54b7e22854 100644 > --- a/libavcodec/binkaudio.c > +++ b/libavcodec/binkaudio.c > @@ -51,13 +51,14 @@ typedef struct BinkAudioContext { > int version_b; ///< Bink version 'b' > int first; > int channels; > + int ch_offset; > int frame_len; ///< transform size (samples) > int overlap_len; ///< overlap size (samples) > int block_size; > int num_bands; > float root; > unsigned int bands[26]; > - float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block > + float previous[6][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block > float quant_table[96]; > AVPacket *pkt; > union { > @@ -74,6 +75,7 @@ static av_cold int decode_init(AVCodecContext *avctx) > int sample_rate_half; > int i, ret; > int frame_len_bits; > + int max_channels = avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT ? MAX_CHANNELS : 6; If you allow up to six channels, then MAX_CHANNELS (i.e. two) needs to be renamed. > int channels = avctx->ch_layout.nb_channels; > > /* determine frame length */ > @@ -85,7 +87,7 @@ static av_cold int decode_init(AVCodecContext *avctx) > frame_len_bits = 11; > } > > - if (channels < 1 || channels > MAX_CHANNELS) { > + if (channels < 1 || channels > max_channels) { > av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", channels); > return AVERROR_INVALIDDATA; > } > @@ -110,7 +112,7 @@ static av_cold int decode_init(AVCodecContext *avctx) > > s->frame_len = 1 << frame_len_bits; > s->overlap_len = s->frame_len / 16; > - s->block_size = (s->frame_len - s->overlap_len) * s->channels; > + s->block_size = (s->frame_len - s->overlap_len) * FFMIN(MAX_CHANNELS, s->channels); > sample_rate_half = (sample_rate + 1LL) / 2; > if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) > s->root = 2.0 / (sqrt(s->frame_len) * 32768.0); > @@ -166,7 +168,8 @@ static const uint8_t rle_length_tab[16] = { > * @param[out] out Output buffer (must contain s->block_size elements) > * @return 0 on success, negative error code on failure > */ > -static int decode_block(BinkAudioContext *s, float **out, int use_dct) > +static int decode_block(BinkAudioContext *s, float **out, int use_dct, > + int channels, int ch_offset) > { > int ch, i, j, k; > float q, quant[25]; > @@ -176,8 +179,8 @@ static int decode_block(BinkAudioContext *s, float **out, int use_dct) > if (use_dct) > skip_bits(gb, 2); > > - for (ch = 0; ch < s->channels; ch++) { > - FFTSample *coeffs = out[ch]; > + for (ch = 0; ch < channels; ch++) { > + FFTSample *coeffs = out[ch + ch_offset]; > > if (s->version_b) { > if (get_bits_left(gb) < 64) > @@ -252,17 +255,17 @@ static int decode_block(BinkAudioContext *s, float **out, int use_dct) > s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs); > } > > - for (ch = 0; ch < s->channels; ch++) { > + for (ch = 0; ch < channels; ch++) { > int j; > - int count = s->overlap_len * s->channels; > + int count = s->overlap_len * channels; > if (!s->first) { > j = ch; > - for (i = 0; i < s->overlap_len; i++, j += s->channels) > - out[ch][i] = (s->previous[ch][i] * (count - j) + > - out[ch][i] * j) / count; > + for (i = 0; i < s->overlap_len; i++, j += channels) > + out[ch + ch_offset][i] = (s->previous[ch + ch_offset][i] * (count - j) + > + out[ch + ch_offset][i] * j) / count; > } > - memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len], > - s->overlap_len * sizeof(*s->previous[ch])); > + memcpy(s->previous[ch + ch_offset], &out[ch + ch_offset][s->frame_len - s->overlap_len], > + s->overlap_len * sizeof(*s->previous[ch + ch_offset])); > } > > s->first = 0; > @@ -293,6 +296,7 @@ static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame) > GetBitContext *gb = &s->gb; > int ret; > > +again: > if (!s->pkt->data) { > ret = ff_decode_get_packet(avctx, s->pkt); > if (ret < 0) > @@ -313,22 +317,31 @@ static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame) > } > > /* get output buffer */ > - frame->nb_samples = s->frame_len; > - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) > - return ret; > + if (s->ch_offset == 0) { > + frame->nb_samples = s->frame_len; > + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) > + return ret; > + } > > if (decode_block(s, (float **)frame->extended_data, > - avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) { > + avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT, > + FFMIN(MAX_CHANNELS, s->channels), s->ch_offset)) { > av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n"); > return AVERROR_INVALIDDATA; > } > + s->ch_offset += MAX_CHANNELS; > get_bits_align32(gb); > if (!get_bits_left(gb)) { > memset(gb, 0, sizeof(*gb)); > av_packet_unref(s->pkt); > } > + if (s->ch_offset >= s->channels) { > + s->ch_offset = 0; > + } else { > + goto again; > + } Is it really intended that the data for one multi-channel frame is divided into several input packets? > > - frame->nb_samples = s->block_size / avctx->ch_layout.nb_channels; > + frame->nb_samples = s->block_size / FFMIN(avctx->ch_layout.nb_channels, MAX_CHANNELS); > > return 0; > fail: > @@ -343,6 +356,7 @@ static void decode_flush(AVCodecContext *avctx) > /* s->pkt coincides with avctx->internal->in_pkt > * and is unreferenced generically when flushing. */ > s->first = 1; > + s->ch_offset = 0; > } > > const AVCodec ff_binkaudio_rdft_decoder = {
On 3/18/22, Andreas Rheinhardt <andreas.rheinhardt@outlook.com> wrote: > Paul B Mahol: >> As presented in .binka files. >> >> Signed-off-by: Paul B Mahol <onemda@gmail.com> >> --- >> libavcodec/binkaudio.c | 50 +++++++++++++++++++++++++++--------------- >> 1 file changed, 32 insertions(+), 18 deletions(-) >> >> diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c >> index b4ff15beeb..54b7e22854 100644 >> --- a/libavcodec/binkaudio.c >> +++ b/libavcodec/binkaudio.c >> @@ -51,13 +51,14 @@ typedef struct BinkAudioContext { >> int version_b; ///< Bink version 'b' >> int first; >> int channels; >> + int ch_offset; >> int frame_len; ///< transform size (samples) >> int overlap_len; ///< overlap size (samples) >> int block_size; >> int num_bands; >> float root; >> unsigned int bands[26]; >> - float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs >> from previous audio block >> + float previous[6][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from >> previous audio block >> float quant_table[96]; >> AVPacket *pkt; >> union { >> @@ -74,6 +75,7 @@ static av_cold int decode_init(AVCodecContext *avctx) >> int sample_rate_half; >> int i, ret; >> int frame_len_bits; >> + int max_channels = avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT ? >> MAX_CHANNELS : 6; > > If you allow up to six channels, then MAX_CHANNELS (i.e. two) needs to > be renamed. > >> int channels = avctx->ch_layout.nb_channels; >> >> /* determine frame length */ >> @@ -85,7 +87,7 @@ static av_cold int decode_init(AVCodecContext *avctx) >> frame_len_bits = 11; >> } >> >> - if (channels < 1 || channels > MAX_CHANNELS) { >> + if (channels < 1 || channels > max_channels) { >> av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", >> channels); >> return AVERROR_INVALIDDATA; >> } >> @@ -110,7 +112,7 @@ static av_cold int decode_init(AVCodecContext *avctx) >> >> s->frame_len = 1 << frame_len_bits; >> s->overlap_len = s->frame_len / 16; >> - s->block_size = (s->frame_len - s->overlap_len) * s->channels; >> + s->block_size = (s->frame_len - s->overlap_len) * >> FFMIN(MAX_CHANNELS, s->channels); >> sample_rate_half = (sample_rate + 1LL) / 2; >> if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) >> s->root = 2.0 / (sqrt(s->frame_len) * 32768.0); >> @@ -166,7 +168,8 @@ static const uint8_t rle_length_tab[16] = { >> * @param[out] out Output buffer (must contain s->block_size elements) >> * @return 0 on success, negative error code on failure >> */ >> -static int decode_block(BinkAudioContext *s, float **out, int use_dct) >> +static int decode_block(BinkAudioContext *s, float **out, int use_dct, >> + int channels, int ch_offset) >> { >> int ch, i, j, k; >> float q, quant[25]; >> @@ -176,8 +179,8 @@ static int decode_block(BinkAudioContext *s, float >> **out, int use_dct) >> if (use_dct) >> skip_bits(gb, 2); >> >> - for (ch = 0; ch < s->channels; ch++) { >> - FFTSample *coeffs = out[ch]; >> + for (ch = 0; ch < channels; ch++) { >> + FFTSample *coeffs = out[ch + ch_offset]; >> >> if (s->version_b) { >> if (get_bits_left(gb) < 64) >> @@ -252,17 +255,17 @@ static int decode_block(BinkAudioContext *s, float >> **out, int use_dct) >> s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs); >> } >> >> - for (ch = 0; ch < s->channels; ch++) { >> + for (ch = 0; ch < channels; ch++) { >> int j; >> - int count = s->overlap_len * s->channels; >> + int count = s->overlap_len * channels; >> if (!s->first) { >> j = ch; >> - for (i = 0; i < s->overlap_len; i++, j += s->channels) >> - out[ch][i] = (s->previous[ch][i] * (count - j) + >> - out[ch][i] * j) / count; >> + for (i = 0; i < s->overlap_len; i++, j += channels) >> + out[ch + ch_offset][i] = (s->previous[ch + ch_offset][i] >> * (count - j) + >> + out[ch + ch_offset][i] * >> j) / count; >> } >> - memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len], >> - s->overlap_len * sizeof(*s->previous[ch])); >> + memcpy(s->previous[ch + ch_offset], &out[ch + >> ch_offset][s->frame_len - s->overlap_len], >> + s->overlap_len * sizeof(*s->previous[ch + ch_offset])); >> } >> >> s->first = 0; >> @@ -293,6 +296,7 @@ static int binkaudio_receive_frame(AVCodecContext >> *avctx, AVFrame *frame) >> GetBitContext *gb = &s->gb; >> int ret; >> >> +again: >> if (!s->pkt->data) { >> ret = ff_decode_get_packet(avctx, s->pkt); >> if (ret < 0) >> @@ -313,22 +317,31 @@ static int binkaudio_receive_frame(AVCodecContext >> *avctx, AVFrame *frame) >> } >> >> /* get output buffer */ >> - frame->nb_samples = s->frame_len; >> - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) >> - return ret; >> + if (s->ch_offset == 0) { >> + frame->nb_samples = s->frame_len; >> + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) >> + return ret; >> + } >> >> if (decode_block(s, (float **)frame->extended_data, >> - avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) { >> + avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT, >> + FFMIN(MAX_CHANNELS, s->channels), s->ch_offset)) { >> av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n"); >> return AVERROR_INVALIDDATA; >> } >> + s->ch_offset += MAX_CHANNELS; >> get_bits_align32(gb); >> if (!get_bits_left(gb)) { >> memset(gb, 0, sizeof(*gb)); >> av_packet_unref(s->pkt); >> } >> + if (s->ch_offset >= s->channels) { >> + s->ch_offset = 0; >> + } else { >> + goto again; >> + } > > Is it really intended that the data for one multi-channel frame is > divided into several input packets? You are missing big picture here, >2 files have channels in different packets interleaved. Something like in XMA. (And nothing signals how are they interleaved. so its worse than in XMA) So it is working fine. I just need another look for possible regressions and security implications. Renaming MAX_CHANNELS is not useful as that is not property of both codecs. > >> >> - frame->nb_samples = s->block_size / avctx->ch_layout.nb_channels; >> + frame->nb_samples = s->block_size / >> FFMIN(avctx->ch_layout.nb_channels, MAX_CHANNELS); >> >> return 0; >> fail: >> @@ -343,6 +356,7 @@ static void decode_flush(AVCodecContext *avctx) >> /* s->pkt coincides with avctx->internal->in_pkt >> * and is unreferenced generically when flushing. */ >> s->first = 1; >> + s->ch_offset = 0; >> } >> >> const AVCodec ff_binkaudio_rdft_decoder = { > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". >
On Fri, Mar 18, 2022 at 04:21:44PM +0100, Paul B Mahol wrote: > On 3/18/22, Andreas Rheinhardt <andreas.rheinhardt@outlook.com> wrote: > > Paul B Mahol: > >> As presented in .binka files. > >> > >> Signed-off-by: Paul B Mahol <onemda@gmail.com> > >> --- > >> libavcodec/binkaudio.c | 50 +++++++++++++++++++++++++++--------------- > >> 1 file changed, 32 insertions(+), 18 deletions(-) > >> > >> diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c > >> index b4ff15beeb..54b7e22854 100644 > >> --- a/libavcodec/binkaudio.c > >> +++ b/libavcodec/binkaudio.c > >> @@ -51,13 +51,14 @@ typedef struct BinkAudioContext { > >> int version_b; ///< Bink version 'b' > >> int first; > >> int channels; > >> + int ch_offset; > >> int frame_len; ///< transform size (samples) > >> int overlap_len; ///< overlap size (samples) > >> int block_size; > >> int num_bands; > >> float root; > >> unsigned int bands[26]; > >> - float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs > >> from previous audio block > >> + float previous[6][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from > >> previous audio block > >> float quant_table[96]; > >> AVPacket *pkt; > >> union { > >> @@ -74,6 +75,7 @@ static av_cold int decode_init(AVCodecContext *avctx) > >> int sample_rate_half; > >> int i, ret; > >> int frame_len_bits; > >> + int max_channels = avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT ? > >> MAX_CHANNELS : 6; > > > > If you allow up to six channels, then MAX_CHANNELS (i.e. two) needs to > > be renamed. > > > >> int channels = avctx->ch_layout.nb_channels; > >> > >> /* determine frame length */ > >> @@ -85,7 +87,7 @@ static av_cold int decode_init(AVCodecContext *avctx) > >> frame_len_bits = 11; > >> } > >> > >> - if (channels < 1 || channels > MAX_CHANNELS) { > >> + if (channels < 1 || channels > max_channels) { > >> av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", > >> channels); > >> return AVERROR_INVALIDDATA; > >> } > >> @@ -110,7 +112,7 @@ static av_cold int decode_init(AVCodecContext *avctx) > >> > >> s->frame_len = 1 << frame_len_bits; > >> s->overlap_len = s->frame_len / 16; > >> - s->block_size = (s->frame_len - s->overlap_len) * s->channels; > >> + s->block_size = (s->frame_len - s->overlap_len) * > >> FFMIN(MAX_CHANNELS, s->channels); > >> sample_rate_half = (sample_rate + 1LL) / 2; > >> if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) > >> s->root = 2.0 / (sqrt(s->frame_len) * 32768.0); > >> @@ -166,7 +168,8 @@ static const uint8_t rle_length_tab[16] = { > >> * @param[out] out Output buffer (must contain s->block_size elements) > >> * @return 0 on success, negative error code on failure > >> */ > >> -static int decode_block(BinkAudioContext *s, float **out, int use_dct) > >> +static int decode_block(BinkAudioContext *s, float **out, int use_dct, > >> + int channels, int ch_offset) > >> { > >> int ch, i, j, k; > >> float q, quant[25]; > >> @@ -176,8 +179,8 @@ static int decode_block(BinkAudioContext *s, float > >> **out, int use_dct) > >> if (use_dct) > >> skip_bits(gb, 2); > >> > >> - for (ch = 0; ch < s->channels; ch++) { > >> - FFTSample *coeffs = out[ch]; > >> + for (ch = 0; ch < channels; ch++) { > >> + FFTSample *coeffs = out[ch + ch_offset]; > >> > >> if (s->version_b) { > >> if (get_bits_left(gb) < 64) > >> @@ -252,17 +255,17 @@ static int decode_block(BinkAudioContext *s, float > >> **out, int use_dct) > >> s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs); > >> } > >> > >> - for (ch = 0; ch < s->channels; ch++) { > >> + for (ch = 0; ch < channels; ch++) { > >> int j; > >> - int count = s->overlap_len * s->channels; > >> + int count = s->overlap_len * channels; > >> if (!s->first) { > >> j = ch; > >> - for (i = 0; i < s->overlap_len; i++, j += s->channels) > >> - out[ch][i] = (s->previous[ch][i] * (count - j) + > >> - out[ch][i] * j) / count; > >> + for (i = 0; i < s->overlap_len; i++, j += channels) > >> + out[ch + ch_offset][i] = (s->previous[ch + ch_offset][i] > >> * (count - j) + > >> + out[ch + ch_offset][i] * > >> j) / count; ^^^ This line needs to be indented some more, to match the previous line. > >> } > >> - memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len], > >> - s->overlap_len * sizeof(*s->previous[ch])); > >> + memcpy(s->previous[ch + ch_offset], &out[ch + > >> ch_offset][s->frame_len - s->overlap_len], > >> + s->overlap_len * sizeof(*s->previous[ch + ch_offset])); > >> } > >> > >> s->first = 0; > >> @@ -293,6 +296,7 @@ static int binkaudio_receive_frame(AVCodecContext > >> *avctx, AVFrame *frame) > >> GetBitContext *gb = &s->gb; > >> int ret; > >> > >> +again: > >> if (!s->pkt->data) { > >> ret = ff_decode_get_packet(avctx, s->pkt); > >> if (ret < 0) > >> @@ -313,22 +317,31 @@ static int binkaudio_receive_frame(AVCodecContext > >> *avctx, AVFrame *frame) > >> } > >> > >> /* get output buffer */ > >> - frame->nb_samples = s->frame_len; > >> - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) > >> - return ret; > >> + if (s->ch_offset == 0) { > >> + frame->nb_samples = s->frame_len; > >> + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) > >> + return ret; > >> + } > >> > >> if (decode_block(s, (float **)frame->extended_data, > >> - avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) { > >> + avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT, > >> + FFMIN(MAX_CHANNELS, s->channels), s->ch_offset)) { > >> av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n"); > >> return AVERROR_INVALIDDATA; > >> } > >> + s->ch_offset += MAX_CHANNELS; > >> get_bits_align32(gb); > >> if (!get_bits_left(gb)) { > >> memset(gb, 0, sizeof(*gb)); > >> av_packet_unref(s->pkt); > >> } > >> + if (s->ch_offset >= s->channels) { > >> + s->ch_offset = 0; > >> + } else { > >> + goto again; > >> + } > > > > Is it really intended that the data for one multi-channel frame is > > divided into several input packets? > > You are missing big picture here, >2 files have channels in different > packets interleaved. > Something like in XMA. (And nothing signals how are they interleaved. > so its worse than in XMA) So it is working fine. I just need another > look for possible regressions and security implications. Renaming > MAX_CHANNELS is not useful as that is not property of both codecs. MAX_CHANNELS (2) *is* a property of both codecs, and should be left alone. I would prefer the '6' magic number be put into a descriptive macro. LGTM. -- Peter (A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c index b4ff15beeb..54b7e22854 100644 --- a/libavcodec/binkaudio.c +++ b/libavcodec/binkaudio.c @@ -51,13 +51,14 @@ typedef struct BinkAudioContext { int version_b; ///< Bink version 'b' int first; int channels; + int ch_offset; int frame_len; ///< transform size (samples) int overlap_len; ///< overlap size (samples) int block_size; int num_bands; float root; unsigned int bands[26]; - float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block + float previous[6][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block float quant_table[96]; AVPacket *pkt; union { @@ -74,6 +75,7 @@ static av_cold int decode_init(AVCodecContext *avctx) int sample_rate_half; int i, ret; int frame_len_bits; + int max_channels = avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT ? MAX_CHANNELS : 6; int channels = avctx->ch_layout.nb_channels; /* determine frame length */ @@ -85,7 +87,7 @@ static av_cold int decode_init(AVCodecContext *avctx) frame_len_bits = 11; } - if (channels < 1 || channels > MAX_CHANNELS) { + if (channels < 1 || channels > max_channels) { av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", channels); return AVERROR_INVALIDDATA; } @@ -110,7 +112,7 @@ static av_cold int decode_init(AVCodecContext *avctx) s->frame_len = 1 << frame_len_bits; s->overlap_len = s->frame_len / 16; - s->block_size = (s->frame_len - s->overlap_len) * s->channels; + s->block_size = (s->frame_len - s->overlap_len) * FFMIN(MAX_CHANNELS, s->channels); sample_rate_half = (sample_rate + 1LL) / 2; if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) s->root = 2.0 / (sqrt(s->frame_len) * 32768.0); @@ -166,7 +168,8 @@ static const uint8_t rle_length_tab[16] = { * @param[out] out Output buffer (must contain s->block_size elements) * @return 0 on success, negative error code on failure */ -static int decode_block(BinkAudioContext *s, float **out, int use_dct) +static int decode_block(BinkAudioContext *s, float **out, int use_dct, + int channels, int ch_offset) { int ch, i, j, k; float q, quant[25]; @@ -176,8 +179,8 @@ static int decode_block(BinkAudioContext *s, float **out, int use_dct) if (use_dct) skip_bits(gb, 2); - for (ch = 0; ch < s->channels; ch++) { - FFTSample *coeffs = out[ch]; + for (ch = 0; ch < channels; ch++) { + FFTSample *coeffs = out[ch + ch_offset]; if (s->version_b) { if (get_bits_left(gb) < 64) @@ -252,17 +255,17 @@ static int decode_block(BinkAudioContext *s, float **out, int use_dct) s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs); } - for (ch = 0; ch < s->channels; ch++) { + for (ch = 0; ch < channels; ch++) { int j; - int count = s->overlap_len * s->channels; + int count = s->overlap_len * channels; if (!s->first) { j = ch; - for (i = 0; i < s->overlap_len; i++, j += s->channels) - out[ch][i] = (s->previous[ch][i] * (count - j) + - out[ch][i] * j) / count; + for (i = 0; i < s->overlap_len; i++, j += channels) + out[ch + ch_offset][i] = (s->previous[ch + ch_offset][i] * (count - j) + + out[ch + ch_offset][i] * j) / count; } - memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len], - s->overlap_len * sizeof(*s->previous[ch])); + memcpy(s->previous[ch + ch_offset], &out[ch + ch_offset][s->frame_len - s->overlap_len], + s->overlap_len * sizeof(*s->previous[ch + ch_offset])); } s->first = 0; @@ -293,6 +296,7 @@ static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame) GetBitContext *gb = &s->gb; int ret; +again: if (!s->pkt->data) { ret = ff_decode_get_packet(avctx, s->pkt); if (ret < 0) @@ -313,22 +317,31 @@ static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame) } /* get output buffer */ - frame->nb_samples = s->frame_len; - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) - return ret; + if (s->ch_offset == 0) { + frame->nb_samples = s->frame_len; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + } if (decode_block(s, (float **)frame->extended_data, - avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) { + avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT, + FFMIN(MAX_CHANNELS, s->channels), s->ch_offset)) { av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n"); return AVERROR_INVALIDDATA; } + s->ch_offset += MAX_CHANNELS; get_bits_align32(gb); if (!get_bits_left(gb)) { memset(gb, 0, sizeof(*gb)); av_packet_unref(s->pkt); } + if (s->ch_offset >= s->channels) { + s->ch_offset = 0; + } else { + goto again; + } - frame->nb_samples = s->block_size / avctx->ch_layout.nb_channels; + frame->nb_samples = s->block_size / FFMIN(avctx->ch_layout.nb_channels, MAX_CHANNELS); return 0; fail: @@ -343,6 +356,7 @@ static void decode_flush(AVCodecContext *avctx) /* s->pkt coincides with avctx->internal->in_pkt * and is unreferenced generically when flushing. */ s->first = 1; + s->ch_offset = 0; } const AVCodec ff_binkaudio_rdft_decoder = {
As presented in .binka files. Signed-off-by: Paul B Mahol <onemda@gmail.com> --- libavcodec/binkaudio.c | 50 +++++++++++++++++++++++++++--------------- 1 file changed, 32 insertions(+), 18 deletions(-)