| Message ID | CAPUDrwdFcSaxyGd_GgWsMSk7GOi_5f89z31hP8m_YBzUOGZL-g@mail.gmail.com |
|---|---|
| State | Superseded |
| Headers | show |
On Tue, Feb 20, 2018 at 3:37 PM, Dale Curtis <dalecurtis@chromium.org> wrote: > From 97380752ef12337d9b9a01801a9a84df3b4b9c0a Mon Sep 17 00:00:00 2001 > From: Dale Curtis <dalecurtis@chromium.org> > Date: Thu, 15 Feb 2018 16:22:55 -0800 > Subject: [PATCH] [aacdec] Parse and drop gain control data, so that SSR > packets decode. > > This will result in poor quality audio for SSR streams, but they > will at least demux and decode without error; partially fixing > ticket #1693. > > This just pulls in the decode_gain_control() function from the > ffmpeg summer-of-code repo (original author Robert Swain) at > svn://svn.mplayerhq.hu/soc/aac/aac.c and adds AOT_AAC_SSR to > decode_audio_specific_config_gb(). > > Signed-off-by: Dale Curtis <dalecurtis@chromium.org> > Co-authored-by: Robert Swain <robert.swain@gmail.com> > --- > libavcodec/aacdec_template.c | 49 +++++++++++++++++++++++++++++++++--- > 1 file changed, 46 insertions(+), 3 deletions(-) > > diff --git a/libavcodec/aacdec_template.c b/libavcodec/aacdec_template.c > index c2d9802023..c3ec3eefe8 100644 > --- a/libavcodec/aacdec_template.c > +++ b/libavcodec/aacdec_template.c > @@ -997,6 +997,7 @@ static int decode_audio_specific_config_gb(AACContext *ac, > switch (m4ac->object_type) { > case AOT_AAC_MAIN: > case AOT_AAC_LC: > + case AOT_AAC_SSR: > case AOT_AAC_LTP: > case AOT_ER_AAC_LC: > case AOT_ER_AAC_LD: > @@ -1967,6 +1968,44 @@ static void apply_prediction(AACContext *ac, SingleChannelElement *sce) > reset_all_predictors(sce->predictor_state); > } > > +static int decode_gain_control(SingleChannelElement * sce, GetBitContext * gb) > +{ > + // wd_num, wd_test, aloc_size > + static const int gain_mode[4][3] = { I would make this uint8_t > + {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0, > + {2, 1, 2}, // LONG_START_SEQUENCE, > + {8, 0, 2}, // EIGHT_SHORT_SEQUENCE, > + {2, 1, 5}, // LONG_STOP_SEQUENCE > + }; > + > + // per-element gain control for SSR > + struct { > + int max_band; > + int adjust_num[4][8]; > + int alev[4][8][8]; > + int aloc[4][8][8]; > + } ssr; I would drop this stack struck and replace the leaf get_bits() with skip bits. It makes the code that much harder to exploit and there is no point in storing data we don't plan on decoding > + > + const int mode = sce->ics.window_sequence[0]; > + int bd, wd, ad; > + > + // FIXME: Store the gain control data on |sce| and do something with it. > + ssr.max_band = get_bits(gb, 2); > + for (bd = 0; bd < ssr.max_band; bd++) { > + for (wd = 0; wd < gain_mode[mode][0]; wd++) { > + ssr.adjust_num[bd][wd] = get_bits(gb, 3); > + for (ad = 0; ad < ssr.adjust_num[bd][wd]; ad++) { > + ssr.alev[bd][wd][ad] = get_bits(gb, 4); > + if (wd == 0 && gain_mode[mode][1]) > + ssr.aloc[bd][wd][ad] = get_bits(gb, 4); > + else > + ssr.aloc[bd][wd][ad] = get_bits(gb, gain_mode[mode][2]); > + } > + } > + } > + return 0; > +} > + > /** > * Decode an individual_channel_stream payload; reference: table 4.44. > * > @@ -2034,9 +2073,13 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce, > goto fail; > } > if (!eld_syntax && get_bits1(gb)) { > - avpriv_request_sample(ac->avctx, "SSR"); > - ret = AVERROR_PATCHWELCOME; > - goto fail; > + // FIXME: Do something with the gain control data... > + ret = decode_gain_control(sce, gb); > + if (ret < 0) { > + ret = AVERROR_PATCHWELCOME; > + avpriv_request_sample(ac->avctx, "SSR"); > + goto fail; > + } This block seems funny. decode_gain_control() always returns zero. Maybe make this warn once per stream when present like some of the other AAC warn cases. > } > // I see no textual basis in the spec for this occurring after SSR gain > // control, but this is what both reference and real implmentations do > -- > 2.16.1.291.g4437f3f132-goog >
From 97380752ef12337d9b9a01801a9a84df3b4b9c0a Mon Sep 17 00:00:00 2001 From: Dale Curtis <dalecurtis@chromium.org> Date: Thu, 15 Feb 2018 16:22:55 -0800 Subject: [PATCH] [aacdec] Parse and drop gain control data, so that SSR packets decode. This will result in poor quality audio for SSR streams, but they will at least demux and decode without error; partially fixing ticket #1693. This just pulls in the decode_gain_control() function from the ffmpeg summer-of-code repo (original author Robert Swain) at svn://svn.mplayerhq.hu/soc/aac/aac.c and adds AOT_AAC_SSR to decode_audio_specific_config_gb(). Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Co-authored-by: Robert Swain <robert.swain@gmail.com> --- libavcodec/aacdec_template.c | 49 +++++++++++++++++++++++++++++++++--- 1 file changed, 46 insertions(+), 3 deletions(-) diff --git a/libavcodec/aacdec_template.c b/libavcodec/aacdec_template.c index c2d9802023..c3ec3eefe8 100644 --- a/libavcodec/aacdec_template.c +++ b/libavcodec/aacdec_template.c @@ -997,6 +997,7 @@ static int decode_audio_specific_config_gb(AACContext *ac, switch (m4ac->object_type) { case AOT_AAC_MAIN: case AOT_AAC_LC: + case AOT_AAC_SSR: case AOT_AAC_LTP: case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: @@ -1967,6 +1968,44 @@ static void apply_prediction(AACContext *ac, SingleChannelElement *sce) reset_all_predictors(sce->predictor_state); } +static int decode_gain_control(SingleChannelElement * sce, GetBitContext * gb) +{ + // wd_num, wd_test, aloc_size + static const int gain_mode[4][3] = { + {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0, + {2, 1, 2}, // LONG_START_SEQUENCE, + {8, 0, 2}, // EIGHT_SHORT_SEQUENCE, + {2, 1, 5}, // LONG_STOP_SEQUENCE + }; + + // per-element gain control for SSR + struct { + int max_band; + int adjust_num[4][8]; + int alev[4][8][8]; + int aloc[4][8][8]; + } ssr; + + const int mode = sce->ics.window_sequence[0]; + int bd, wd, ad; + + // FIXME: Store the gain control data on |sce| and do something with it. + ssr.max_band = get_bits(gb, 2); + for (bd = 0; bd < ssr.max_band; bd++) { + for (wd = 0; wd < gain_mode[mode][0]; wd++) { + ssr.adjust_num[bd][wd] = get_bits(gb, 3); + for (ad = 0; ad < ssr.adjust_num[bd][wd]; ad++) { + ssr.alev[bd][wd][ad] = get_bits(gb, 4); + if (wd == 0 && gain_mode[mode][1]) + ssr.aloc[bd][wd][ad] = get_bits(gb, 4); + else + ssr.aloc[bd][wd][ad] = get_bits(gb, gain_mode[mode][2]); + } + } + } + return 0; +} + /** * Decode an individual_channel_stream payload; reference: table 4.44. * @@ -2034,9 +2073,13 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce, goto fail; } if (!eld_syntax && get_bits1(gb)) { - avpriv_request_sample(ac->avctx, "SSR"); - ret = AVERROR_PATCHWELCOME; - goto fail; + // FIXME: Do something with the gain control data... + ret = decode_gain_control(sce, gb); + if (ret < 0) { + ret = AVERROR_PATCHWELCOME; + avpriv_request_sample(ac->avctx, "SSR"); + goto fail; + } } // I see no textual basis in the spec for this occurring after SSR gain // control, but this is what both reference and real implmentations do -- 2.16.1.291.g4437f3f132-goog
This will result in poor quality audio for SSR streams, but they will at least demux and decode without error; partially fixing ticket #1693. This just pulls in the decode_gain_control() function from the ffmpeg summer-of-code repo (original author Robert Swain) at svn://svn.mplayerhq.hu/soc/aac/aac.c and adds AOT_AAC_SSR to decode_audio_specific_config_gb(). Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Co-authored-by: Robert Swain <robert.swain@gmail.com>