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4778
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1 mails by A'rpi and Marcus Blomenkamp <Marcus.Blomenkamp@epost.de>
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2 describing how this ac3-passtrough hack work under linux and mplayer...
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3 -----------------------------------------------------------------------
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4 Hi,
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5
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6 > I received the following patch from Steven Brookes <stevenjb@mda.co.uk>.
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7 > He is working on fixing the digital audio output of the dxr3 driver and
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8 > told me he fixed some bugs in mplayer along the way. I don't know shit
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9 > about hwac3 output so all I did was to make sure the patch applied
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10 > against latest cvs.
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11 > This is from his e-mail to me:
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12 >
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13 > "Secondly there is a patch to dec_audio.c and
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14 > ac3-iec958 to fix the -ac hwac3 codec stuff and to use liba52 to sync it.
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15
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16 > Seems to work for everything I've thrown at and maintains sync for all audio
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17 > types through the DXR3."
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18
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19 patch applied (with some comments added and an unwanted change (in software
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20 a52 decoder) removed)
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21
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22 now i understand how this whole hwac3 mess work.
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23 it's very very tricky. it virtually decodes ac3 to LPCM packets, but really
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24 it keeps the original compressed data padded by zeros. this way it's
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25 constant bitrate, and sync is calculated just like for stereo PCM.
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26 (so it bypass LPCM-capable media converters...)
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27
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28 so, every ac3 frame is translated to 6144 byte long tricky LPCM packet.
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29 6144 = 4*(6*256) = 4 * samples_per_ac3_frame = LPCM size of uncompressed ac3
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30 frame.
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31
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32 i wanna know if it works for sblive and other ac3-capable cards too?
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33 (i can't test it, lack of ac3 decoder)
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34
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35 A'rpi / Astral & ESP-team
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36
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37 -----------------------------------------------------------------------
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38 Hi folks.
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39 I spend some time fiddling with ac3 passthrough in mplayer. The
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40 traditional way of setting the output format to AFMT_AC3 was no ideal
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41 solution since not all digital io cards/drivers supported this format or
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42 honoured it to set the spdif non-audio bit. To make it short, it only
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43 worked with oss sblive driver IIRC.
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44
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45 Inspired by alsa's ac3dec program I found an alternative way by
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46 inspecting to which format the alsa device had been set. Suprise: it was
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47 simple 16bit_le 2_channel pcm. So setting the non-audio bit doesn't
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48 necessarily mean the point. The only important thing seems to be
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49 bit-identical output at the correct samplerate. Modern AV-Receivers seem
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50 to be quite tolerant/compatible.
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51
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52 So I changed the output format of hwac3 from
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53
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54 AFMT_AC3 channels=1
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55 to
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56 AFMT_S16_LE channels=2
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57
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58 and corrected the absolute time calculation. That was all to get it
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59 running for me.
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60
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61 -----------------------------------------------------------------------
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62 Hi there.
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63
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64 Perhaps I can clear up some mystification about AC3 passthrough in
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65 general and mplayer in special:
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66
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67 To get the external decoder solution working, it must be fed with data
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68 which is bitidentical to the chunks in the source ac3 file (compressed
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69 data is very picky about bit errors). Additionally - or better to say
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70 'historically' - the non-audio bit should be set in the spdif status
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71 fields to prevent old spdif hardware from reproducing ugly scratchy
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72 noise. Note: for current decoders (probably those with DTS capability)
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73 this safety bit isn't needed anymore. At least I can state that for my
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74 Sherwood RVD-6095RDS. I think it is due to DTS because DTS sound can
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75 reside on a ordinary AudioCD and an ordinary AudioCD-Player will always
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76 have it's audio-bit set.
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77
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78 The sample format of the data must be 2channel 16bit (little endian
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79 IIRC). Samplerates are 48kHz - although my receiver also accepts
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80 44100Hz. I do not know if this is due to an over-compatability of my
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81 receiver or if 44100 is also possible in the ac3 specs. For safety's
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82 sake lets keep this at 48000Hz. AC3 data chunks are inserted into the
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83 stream every 0x1600 bytes (don't bite me on that, look into
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84 'ac3-iec958.c': 'ac3_iec958_build_burst').
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85
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86 To come back to the problem: data must be played bit-identically through
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87 the soundcard at the correct samplerate and should optionally have it's
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88 non-audio bit set. There are two ways to accomplish this:
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89
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90 1) Some OSS guy invented the format AFMT_AC3. Soundcard drivers
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91 implementing this format should therefore adjust it's mixers and
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92 switches to produce the desired output. Unfortunately some soundcard
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93 drivers do not support this format correctly and most do not even
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94 support it at all (including ALSA).
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95
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96 2) The alternative approach currently in mplayer CVS is to simply set
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97 the output format to 48kHz16bitLE and rely on the user to have the
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98 soundcard mixers adjusted properly.
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99
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100 I do have two soundcards with digital IO facilities (CMI8738 and
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101 Trident4DWaveNX based) plus the mentioned decoder. I'm currently running
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102 Linux-2.4.17. Following configurations are happily running here:
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103
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104 1. Trident with ALSA drivers (OSS does not support Hoontech's dig. IO)
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105 2. CMI with ALSA drivers
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106 3. CMI with OSS drivers
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107
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108 For Linux I'd suggest using ALSA because of it's cleaner architecture
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109 and more consitent user interface. Not to mention that it'll be the
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110 standard sound support in Linux soon.
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111
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112 For those who want to stick to OSS drivers: The CMI8738 drivers works
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113 out-of-the-box, if the PCM/Wave mixer is set to 100%.
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114
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115 For ALSA I'd suggest using its OSS emulation. More on that later.
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116 ALSA-0.9 invented the idea of cards, devices and dubdevices. You can
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117 reach the digital interface of all supported cards consitently by using
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118 the device 'hw:x,2' (x counting from 0 is the number of your soundcard).
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119 So most people would end up at 'hw:0,2'. This device can only be opened
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120 in sample formats and rates which are directly supported in hardware
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121 hence no samplerate conversion is done keeping the stream as-is. However
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122 most consumer soundcards do not support 44kHz so it would definitively
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123 be a bad idea to use this as your standard device if you wanted to
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124 listen to some mp3s (most of them are 44kHz due to CD source). Here the
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125 OSS comes to play again. You can configure which OSS device (/dev/dsp
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126 and /dev/adsp) uses which ALSA device. So I'd suggest pointing the
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127 standard '/dev/dsp' to standard 'hw:0,0' which suports mixing and
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128 samplerate conversion. No further reconfiguration would be needed for
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129 your sound apps. For movies I'd point '/dev/adsp' to 'hw:0,2' and
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130 configure mplayer to use adsp instead of dsp. The samplerate constrain
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131 is no big deal here since movies usually are in 48Khz anyway. The
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132 configuration in '/etc/modules.conf' is no big deal also:
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133
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134 alias snd-card-0 snd-card-cmipci # insert your card here
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135 alias snd-card-1 snd-pcm-oss # load OSS emulation
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136 options snd-pcm-oss snd_dsp_map=0 snd_adsp_map=2 # do the mapping
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137
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138 This works flawlessly in combination with alsa's native
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139 SysVrc-init-script 'alsasound'. Be sure to disable any distribution
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140 dependant script (e.g. Mandrake-8.1 has an 'alsa' script which depends
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141 on ALSA-0.5).
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142
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143 Sorry for you *BSD'lers out there. I have no grasp on sound support there.
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144
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145 HTH Marcus
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