On Wed, 2018-07-11 at 17:52 -0500, Pierre-Louis Bossart wrote:
On 07/11/2018 04:43 PM, Ranjani Sridharan wrote:
On Wed, 2018-07-11 at 11:55 -0500, Pierre-Louis Bossart wrote:
On 07/11/2018 01:23 AM, Ranjani Sridharan wrote:
On Tue, 2018-07-10 at 18:15 -0500, Pierre-Louis Bossart wrote:
Add based on sof-glk-codec. In theory this is the same topology with only a swap of SSPs
FIXME: this is a mess between pipeline ID, PCM ID, DAI ID, DAI #. I could use additional eyes to proof-read this...
Looks good to me. I couldnt spot any mistakes. Keyon, Could you please also check?
FIXME: not sure why we would need PCM_CAPTURE_ADD/PCM_PLAYBACK_DAI, this seems to be already done by DAI_ADD?
DAI_ADD only adds the BE DAI's. PCM_CAPTURE_ADD/PLAYBACK_ADD are needed for adding the FE dai links.
ok, but then why do those macros make references to the DAIs?
The dai_id is used when creating the cpu_dai for the PCM elems and their ID is set to the dai_id specified here.
I wish I understood cpu dai's better but I couldnt wrap my head around why we do that.
Keyon, any hints?
So it's a different dai_id from what is used for back-ends? Looks like a complete mess to me...
It is the same ID that is used for the BE DAI, Pierre. Here's a quick look at the all the DAI's that are registered in the case of up2 with the hifiberry dac+ with the test-playback-ssp5*.tplg.
The "Passthrough 0" dai is the last one and has the dai_id of 0 which is the same as the SSP5 id in the topology.
[ 0.076806] NET: Registered protocol family 16 [ 0.176459] Registered efivars operations [ 0.319302] NET: Registered protocol family 2 [ 0.320794] NET: Registered protocol family 1 [ 1.467017] NET: Registered protocol family 10 [ 1.467826] NET: Registered protocol family 17 [ 3.038697] snd-soc-dummy snd-soc-dummy: ASoC: Registered DAI 'snd- soc-dummy-dai' with id 0 [ 3.227853] sof-audio sof-audio: ASoC: Registered platform 'sof- audio' [ 3.227862] sof-audio sof-audio: ASoC: Registered DAI 'SSP0 Pin' with id 0 [ 3.227866] sof-audio sof-audio: ASoC: Registered DAI 'SSP1 Pin' with id 1 [ 3.227869] sof-audio sof-audio: ASoC: Registered DAI 'SSP2 Pin' with id 2 [ 3.227873] sof-audio sof-audio: ASoC: Registered DAI 'SSP3 Pin' with id 3 [ 3.227876] sof-audio sof-audio: ASoC: Registered DAI 'SSP4 Pin' with id 4 [ 3.227879] sof-audio sof-audio: ASoC: Registered DAI 'SSP5 Pin' with id 5 [ 3.227883] sof-audio sof-audio: ASoC: Registered DAI 'DMIC01 Pin' with id 6 [ 3.227886] sof-audio sof-audio: ASoC: Registered DAI 'DMIC16k Pin' with id 7 [ 3.227889] sof-audio sof-audio: ASoC: Registered DAI 'iDisp1 Pin' with id 8 [ 3.227893] sof-audio sof-audio: ASoC: Registered DAI 'iDisp2 Pin' with id 9 [ 3.227896] sof-audio sof-audio: ASoC: Registered DAI 'iDisp3 Pin' with id 10 [ 3.227900] sof-audio sof-audio: ASoC: Registered DAI 'Analog Codec DAI' with id 11 [ 3.227903] sof-audio sof-audio: ASoC: Registered DAI 'Digital Codec DAI' with id 12 [ 3.227906] sof-audio sof-audio: ASoC: Registered DAI 'Alt Analog Codec DAI' with id 13 [ 3.578667] pcm512x i2c-104C5122:00: ASoC: Registered DAI 'pcm512x- hifi' with id 0 [ 4.141636] pstore: Registered efi as persistent store backend [ 4.203432] sof-audio sof-audio: ASoC: Registered DAI 'Passthrough 0' with id 0 [ 4.609670] NET: Registered protocol family 31
+# PCM (PCM_CAPTURE_ADD(name, pipeline, pcm_id, dai_id, capture)) +#PCM_CAPTURE_ADD(DMIC0, 1, 0, 0, PIPELINE_PCM_1) +#PCM_PLAYBACK_ADD(SSP5, 2, 0, 5, PIPELINE_PCM_2) +#PCM_PLAYBACK_ADD(SSP1, 3, 1, 1, PIPELINE_PCM_3)
In addition the information on the PCM # is already provided in PIPELINE_ADD, so this looks redundant on both the FE and BE sides?
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@lin ux.i ntel .com>
topology/Makefile.am | 1 + topology/sof-apl-da7219.m4 | 104 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 105 insertions(+) create mode 100644 topology/sof-apl-da7219.m4
diff --git a/topology/Makefile.am b/topology/Makefile.am index 33d0418..396b5b9 100644 --- a/topology/Makefile.am +++ b/topology/Makefile.am @@ -31,6 +31,7 @@ MACHINES = \ sof-hsw-rt5640.tplg \ sof-apl-tdf8532.tplg \ sof-apl-pcm512x.tplg \
sof-apl-da7219.tplg \ sof-glk-codec.tplg
# Uncomment the following line if you want to debug conf
files diff --git a/topology/sof-apl-da7219.m4 b/topology/sof-apl- da7219.m4 new file mode 100644 index 0000000..3934df5 --- /dev/null +++ b/topology/sof-apl-da7219.m4 @@ -0,0 +1,104 @@ +# +# Topology for AppoloLake with headset on SSP1, spk on SSP5 and DMIC+# PCM (PCM_CAPTURE_ADD(name, pipeline, pcm_id, dai_id, capture)) +#PCM_CAPTURE_ADD(DMIC0, 1, 0, 0, PIPELINE_PCM_1) +#PCM_PLAYBACK_ADD(SSP5, 2, 0, 5, PIPELINE_PCM_2) +#PCM_PLAYBACK_ADD(SSP1, 3, 1, 1, PIPELINE_PCM_3) capture +#
+# Include topology builder +include(`utils.m4') +include(`dai.m4') +include(`pipeline.m4') +include(`ssp.m4')
+# Include TLV library +include(`common/tlv.m4')
+# Include Token library +include(`sof/tokens.m4')
+# Include bxt DSP configuration +include(`platform/intel/bxt.m4') +include(`platform/intel/dmic.m4')
+# +# Define the pipelines +# +# PCM0 <---- volume <----- DMIC0 (dmic capture) +# PCM0 ----> volume -----> SSP5 (speaker - maxim98357a) +# PCM1 ----> volume -----> SSP1 (headset - da7219) +#
+# Low Latency capture pipeline 1 on PCM 0 using max 4 channels of s32le. +# Schedule 48 frames per 1000us deadline on core 0 with priority 0 +PIPELINE_PCM_ADD(sof/pipe-volume-capture.m4,
- 1, 0, 4, s32le,
- 48, 1000, 0, 0)
+# Low Latency playback pipeline 2 on PCM 0 using max 2 channels of s16le. +# Schedule 48 frames per 1000us deadline on core 0 with priority 0 +PIPELINE_PCM_ADD(sof/pipe-volume-playback.m4,
- 2, 0, 2, s16le,
- 48, 1000, 0, 0)
+# Low Latency playback pipeline 3 on PCM 1 using max 2 channels of s16le. +# Schedule 48 frames per 1000us deadline on core 0 with priority 0 +PIPELINE_PCM_ADD(sof/pipe-volume-playback.m4,
- 3, 1, 2, s16le,
- 48, 1000, 0, 0)
+# +# DAIs configuration +#
+# capture DAI is DMIC0 using 2 periods +# Buffers use s16le format, with 48 frame per 1000us on core 0 with priority 0 +DAI_ADD(sof/pipe-dai-capture.m4,
- 1, DMIC, 0, DMIC0,
- PIPELINE_SINK_1, 2, s32le,
- 48, 1000, 0, 0)
+# playback DAI is SSP5 using 2 periods +# Buffers use s16le format, with 48 frame per 1000us on core 0 with priority 0 +DAI_ADD(sof/pipe-dai-playback.m4,
- 2, SSP, 5, SSP5-Codec,
- PIPELINE_SOURCE_2, 2, s16le,
- 48, 1000, 0, 0)
+# playback DAI is SSP1 using 2 periods +# Buffers use s16le format, with 48 frame per 1000us on core 0 with priority 0 +DAI_ADD(sof/pipe-dai-playback.m4,
- 3, SSP, 1, SSP1-Codec,
- PIPELINE_SOURCE_3, 2, s16le,
- 48, 1000, 0, 0)
+# FIXME: Why is this needed? And is this correct? +# PCM (PCM_CAPTURE_ADD(name, pipeline, pcm_id, dai_id, capture)) +#PCM_CAPTURE_ADD(DMIC0, 1, 0, 0, PIPELINE_PCM_1) +#PCM_PLAYBACK_ADD(SSP5, 2, 0, 5, PIPELINE_PCM_2) +#PCM_PLAYBACK_ADD(SSP1, 3, 1, 1, PIPELINE_PCM_3)
+# +# BE configurations - overrides config in ACPI if present +# +# FIXME: the machine driver uses .name = "dmic01" and .id=2 +# DMIC0 (id: 2) +DAI_CONFIG(DMIC, 0, 2, dmic01,
- DMIC_CONFIG(1, 500000, 4800000, 40, 60, 48000,
DMIC_WORD_LENGTH(s32le), DMIC, 0,PDM_CONFIG(DMIC, 0, FOUR_CH_PDM0_PDM1)))+# FIXME: the machine driver uses .name = "SSP5-Codec", .id=0, .cpu_name="SSP5 Pin" +#SSP 5 (ID: 0) with 19.2MHz mclk with MCLK_ID 0 +DAI_CONFIG(SSP, 5, 0, SSP5-Codec,
- SSP_CONFIG(I2S, SSP_CLOCK(mclk, 19200000,
codec_mclk_in),
SSP_CLOCK(bclk, 1920000, codec_slave),SSP_CLOCK(fsync, 48000, codec_slave),SSP_TDM(2, 20, 3, 3),SSP_CONFIG_DATA(SSP, 5, 16, 0)))+# FIXME: the machine driver uses .name = "SSP1-Codec", .id=1, .cpu_name="SSP1 Pin" +#SSP 1 (ID: 1) with 19.2MHz mclk with MCLK_ID 0 +DAI_CONFIG(SSP, 1, 1, SSP1-Codec,
- SSP_CONFIG(I2S, SSP_CLOCK(mclk, 19200000,
codec_mclk_in),
SSP_CLOCK(bclk, 1920000, codec_slave),SSP_CLOCK(fsync, 48000, codec_slave),SSP_TDM(2, 20, 3, 3),SSP_CONFIG_DATA(SSP, 1, 16, 0)))
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