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[190.225.105.197]) by smtp.gmail.com with ESMTPSA id g26-20020a9d621a000000b006add66c7d73sm1267191otj.27.2023.06.20.12.50.26 for (version=TLS1_3 cipher=TLS_AES_128_GCM_SHA256 bits=128/128); Tue, 20 Jun 2023 12:50:26 -0700 (PDT) Message-ID: <0312c295-0fd7-29fe-556c-bd129b9c1e35@gmail.com> Date: Tue, 20 Jun 2023 16:50:38 -0300 MIME-Version: 1.0 User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:102.0) Gecko/20100101 Thunderbird/102.12.0 Content-Language: en-US To: ffmpeg-devel@ffmpeg.org References: <20230620185304.10056-1-anton@khirnov.net> <477c88ca-f9b8-aa95-a2e0-e920b1246d6f@gmail.com> <168729009130.9711.16248011863887429642@lain.khirnov.net> From: James Almer In-Reply-To: <168729009130.9711.16248011863887429642@lain.khirnov.net> Subject: Re: [FFmpeg-devel] [PATCH] tests/fate/ffmpeg: silence the audio for fate-ffmpeg-streamloop-transcode-av X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.29 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Archived-At: List-Archive: List-Post: On 6/20/2023 4:41 PM, Anton Khirnov wrote: > Quoting James Almer (2023-06-20 21:39:22) >> On 6/20/2023 3:53 PM, Anton Khirnov wrote: >>> Fixed-point AAC decoder currently does not produce the same output on >>> all platforms. Until that is fixed, silence the audio stream using the >>> volume filter. >>> >>> Also, actually use the aac_fixed decoder as was the original intent. >>> --- >>> tests/fate/ffmpeg.mak | 7 +- >>> tests/ref/fate/ffmpeg-streamloop-transcode-av | 144 +++++++++--------- >>> 2 files changed, 77 insertions(+), 74 deletions(-) >>> >>> diff --git a/tests/fate/ffmpeg.mak b/tests/fate/ffmpeg.mak >>> index 87cd0b46d0..763ed381ee 100644 >>> --- a/tests/fate/ffmpeg.mak >>> +++ b/tests/fate/ffmpeg.mak >>> @@ -132,9 +132,12 @@ fate-ffmpeg-fix_sub_duration_heartbeat: CMD = fmtstdout srt -fix_sub_duration \ >>> -c:s srt \ >>> -f null - >>> >>> -FATE_SAMPLES_FFMPEG-$(call FRAMECRC, MATROSKA, H264 AAC_FIXED, PCM_S32LE_ENCODER) += fate-ffmpeg-streamloop-transcode-av >>> +# FIXME: the integer AAC decoder does not produce the same output on all platforms >>> +# so until that is fixed we use the volume filter to silence the data >>> +FATE_SAMPLES_FFMPEG-$(call FRAMECRC, MATROSKA, H264 AAC_FIXED, PCM_S32LE_ENCODER VOLUME_FILTER) += fate-ffmpeg-streamloop-transcode-av >>> fate-ffmpeg-streamloop-transcode-av: CMD = \ >>> - framecrc -auto_conversion_filters -stream_loop 3 -i $(TARGET_SAMPLES)/mkv/1242-small.mkv -c:a pcm_s32le >>> + framecrc -auto_conversion_filters -stream_loop 3 -c:a aac_fixed -i $(TARGET_SAMPLES)/mkv/1242-small.mkv \ >>> + -af volume=0 -c:a pcm_s32le >> >> Maybe do volume=0:precision=fixed so you can remove the >> -auto_conversion_filters part. > > It's mainly there to interleave s32p to s32 for encoding Yeah, just checked and it fails without -auto_conversion_filters even if you force fixed point in the filter. Still, even if it's a temporal change, it's best to add precision=fixed since it will prevent conversion to float and back to fixed from auto inserting two aresample instances in the filterchain. With precision=fixed > [AVFilterGraph @ 000001f29a306e20] Setting 'volume' to value '0' > [AVFilterGraph @ 000001f29a306e20] Setting 'precision' to value 'fixed' > [graph_0_in_0_0 @ 000001f29a30eea0] Setting 'time_base' to value '1/48000' > [graph_0_in_0_0 @ 000001f29a30eea0] Setting 'sample_rate' to value '48000' > [graph_0_in_0_0 @ 000001f29a30eea0] Setting 'sample_fmt' to value 's32p' > [graph_0_in_0_0 @ 000001f29a30eea0] Setting 'channel_layout' to value 'stereo' > [graph_0_in_0_0 @ 000001f29a30eea0] tb:1/48000 samplefmt:s32p samplerate:48000 chlayout:stereo > [format_out_0_0 @ 000001f29a0defa0] Setting 'sample_fmts' to value 's32' > [format_out_0_0 @ 000001f29a0defa0] auto-inserting filter 'auto_aresample_0' between the filter 'Parsed_volume_0' and the filter 'format_out_0_0' > [AVFilterGraph @ 000001f29a306e20] query_formats: 4 queried, 4 merged, 6 already done, 0 delayed > [graph_0_in_0_0 @ 000001f29a30eea0] tb:0.000021 sample_rate:48000.000000 nb_channels:2.000000 > [Parsed_volume_0 @ 000001f29a026f60] n:nan t:nan pts:nan precision:fixed volume_i:0/255 volume:0.000000 volume_dB:-inf > [auto_aresample_0 @ 000001f29a0df8a0] [SWR @ 000001f29a1cbf60] Using s32p internally between filters > [auto_aresample_0 @ 000001f29a0df8a0] ch:2 chl:stereo fmt:s32p r:48000Hz -> ch:2 chl:stereo fmt:s32 r:48000Hz Without it > [AVFilterGraph @ 000001ccb6036aa0] Setting 'volume' to value '0' > [graph_0_in_0_0 @ 000001ccb603b6c0] Setting 'time_base' to value '1/48000' > [graph_0_in_0_0 @ 000001ccb603b6c0] Setting 'sample_rate' to value '48000' > [graph_0_in_0_0 @ 000001ccb603b6c0] Setting 'sample_fmt' to value 's32p' > [graph_0_in_0_0 @ 000001ccb603b6c0] Setting 'channel_layout' to value 'stereo' > [graph_0_in_0_0 @ 000001ccb603b6c0] tb:1/48000 samplefmt:s32p samplerate:48000 chlayout:stereo > [format_out_0_0 @ 000001ccb5e0eec0] Setting 'sample_fmts' to value 's32' > [Parsed_volume_0 @ 000001ccb5d56f80] auto-inserting filter 'auto_aresample_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_volume_0' > [format_out_0_0 @ 000001ccb5e0eec0] auto-inserting filter 'auto_aresample_1' between the filter 'Parsed_volume_0' and the filter 'format_out_0_0' > [AVFilterGraph @ 000001ccb6036aa0] query_formats: 4 queried, 2 merged, 9 already done, 0 delayed > [auto_aresample_0 @ 000001ccb5d59be0] picking fltp out of 2 ref:s32p > [auto_aresample_0 @ 000001ccb5d59be0] [SWR @ 000001ccb5efbf80] Using fltp internally between filters > [auto_aresample_0 @ 000001ccb5d59be0] ch:2 chl:stereo fmt:s32p r:48000Hz -> ch:2 chl:stereo fmt:fltp r:48000Hz > [auto_aresample_0 @ 000001ccb5d59be0] tb:0.000021 sample_rate:48000.000000 nb_channels:2.000000 > [Parsed_volume_0 @ 000001ccb5d56f80] n:nan t:nan pts:nan precision:float volume:0.000000 volume_dB:-inf > [auto_aresample_1 @ 000001ccb5d5ae80] [SWR @ 000001ccb5f10fc0] Using fltp internally between filters > [auto_aresample_1 @ 000001ccb5d5ae80] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s32 r:48000Hz _______________________________________________ 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".