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Message-ID: <20230201134947.1638197-14-quic_mohs@quicinc.com>
Date: Wed, 1 Feb 2023 19:19:46 +0530
From: Mohammad Rafi Shaik <quic_mohs@...cinc.com>
To: <krzysztof.kozlowski+dt@...aro.org>, <agross@...nel.org>,
<andersson@...nel.org>, <lgirdwood@...il.com>,
<broonie@...nel.org>, <robh+dt@...nel.org>,
<quic_plai@...cinc.com>, <bgoswami@...cinc.com>,
<srinivas.kandagatla@...aro.org>, <quic_rohkumar@...cinc.com>,
<linux-arm-msm@...r.kernel.org>, <alsa-devel@...a-project.org>,
<linux-kernel@...r.kernel.org>, <swboyd@...omium.org>,
<judyhsiao@...omium.org>, <devicetree@...r.kernel.org>,
<konrad.dybcio@...aro.org>
CC: Mohammad Rafi Shaik <quic_mohs@...cinc.com>
Subject: [PATCH 13/14] ASoC: qdsp6: audioreach: Add MP3, AAC and FLAC compress format support
Add support for handling compressed formats such as MP3, AAC and FLAC.
Signed-off-by: Mohammad Rafi Shaik <quic_mohs@...cinc.com>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
---
sound/soc/qcom/qdsp6/audioreach.c | 106 ++++++++++++++++++++++++------
1 file changed, 86 insertions(+), 20 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 7c45c36e9156..250ed828c7d3 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -852,6 +852,68 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
return rc;
}
+static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
+ void *p, struct audioreach_module_config *mcfg)
+{
+ struct payload_media_fmt_aac_t *aac_cfg;
+ struct payload_media_fmt_pcm *mp3_cfg;
+ struct payload_media_fmt_flac_t *flac_cfg;
+ int ret = 0;
+
+ switch (mcfg->fmt) {
+ case SND_AUDIOCODEC_MP3:
+ media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+ media_fmt_hdr->fmt_id = MEDIA_FMT_ID_MP3;
+ media_fmt_hdr->payload_size = 0;
+ p = p + sizeof(*media_fmt_hdr);
+ mp3_cfg = p;
+ mp3_cfg->sample_rate = mcfg->sample_rate;
+ mp3_cfg->bit_width = mcfg->bit_width;
+ mp3_cfg->alignment = PCM_LSB_ALIGNED;
+ mp3_cfg->bits_per_sample = mcfg->bit_width;
+ mp3_cfg->q_factor = mcfg->bit_width - 1;
+ mp3_cfg->endianness = PCM_LITTLE_ENDIAN;
+ mp3_cfg->num_channels = mcfg->num_channels;
+
+ if (mcfg->num_channels == 1) {
+ mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L;
+ } else if (mcfg->num_channels == 2) {
+ mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L;
+ mp3_cfg->channel_mapping[1] = PCM_CHANNEL_R;
+ }
+ break;
+ case SND_AUDIOCODEC_AAC:
+ media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+ media_fmt_hdr->fmt_id = MEDIA_FMT_ID_AAC;
+ media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_aac_t);
+ p = p + sizeof(*media_fmt_hdr);
+ aac_cfg = p;
+ aac_cfg->aac_fmt_flag = 0;
+ aac_cfg->audio_obj_type = 5;
+ aac_cfg->num_channels = mcfg->num_channels;
+ aac_cfg->total_size_of_PCE_bits = 0;
+ aac_cfg->sample_rate = mcfg->sample_rate;
+ break;
+ case SND_AUDIOCODEC_FLAC:
+ media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+ media_fmt_hdr->fmt_id = MEDIA_FMT_ID_FLAC;
+ media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_flac_t);
+ p = p + sizeof(*media_fmt_hdr);
+ flac_cfg = p;
+ flac_cfg->sample_size = mcfg->codec.options.flac_d.sample_size;
+ flac_cfg->num_channels = mcfg->num_channels;
+ flac_cfg->min_blk_size = mcfg->codec.options.flac_d.min_blk_size;
+ flac_cfg->max_blk_size = mcfg->codec.options.flac_d.max_blk_size;
+ flac_cfg->sample_rate = mcfg->sample_rate;
+ flac_cfg->min_frame_size = mcfg->codec.options.flac_d.min_frame_size;
+ flac_cfg->max_frame_size = mcfg->codec.options.flac_d.max_frame_size;
+ break;
+ default:
+ return -EINVAL;
+ }
+ return ret;
+}
+
static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
struct audioreach_module *module,
struct audioreach_module_config *cfg)
@@ -1055,26 +1117,29 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph,
p = p + APM_MODULE_PARAM_DATA_SIZE;
header = p;
- header->data_format = DATA_FORMAT_FIXED_POINT;
- header->fmt_id = MEDIA_FMT_ID_PCM;
- header->payload_size = payload_size - sizeof(*header);
+ if (mcfg->fmt == SND_AUDIOCODEC_PCM) {
+ header->data_format = DATA_FORMAT_FIXED_POINT;
+ header->fmt_id = MEDIA_FMT_ID_PCM;
+ header->payload_size = payload_size - sizeof(*header);
- p = p + sizeof(*header);
- cfg = p;
- cfg->sample_rate = mcfg->sample_rate;
- cfg->bit_width = mcfg->bit_width;
- cfg->alignment = PCM_LSB_ALIGNED;
- cfg->bits_per_sample = mcfg->bit_width;
- cfg->q_factor = mcfg->bit_width - 1;
- cfg->endianness = PCM_LITTLE_ENDIAN;
- cfg->num_channels = mcfg->num_channels;
-
- if (mcfg->num_channels == 1) {
- cfg->channel_mapping[0] = PCM_CHANNEL_L;
- } else if (num_channels == 2) {
- cfg->channel_mapping[0] = PCM_CHANNEL_L;
- cfg->channel_mapping[1] = PCM_CHANNEL_R;
- }
+ p = p + sizeof(*header);
+ cfg = p;
+ cfg->sample_rate = mcfg->sample_rate;
+ cfg->bit_width = mcfg->bit_width;
+ cfg->alignment = PCM_LSB_ALIGNED;
+ cfg->bits_per_sample = mcfg->bit_width;
+ cfg->q_factor = mcfg->bit_width - 1;
+ cfg->endianness = PCM_LITTLE_ENDIAN;
+ cfg->num_channels = mcfg->num_channels;
+
+ if (mcfg->num_channels == 1)
+ cfg->channel_mapping[0] = PCM_CHANNEL_L;
+ else if (num_channels == 2) {
+ cfg->channel_mapping[0] = PCM_CHANNEL_L;
+ cfg->channel_mapping[1] = PCM_CHANNEL_R;
+ }
+ } else
+ audioreach_set_compr_media_format(header, p, mcfg);
rc = audioreach_graph_send_cmd_sync(graph, pkt, 0);
@@ -1401,7 +1466,8 @@ int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_modu
cfg->channel_mapping[0] = PCM_CHANNEL_L;
cfg->channel_mapping[1] = PCM_CHANNEL_R;
}
- }
+ } else
+ audioreach_set_compr_media_format(header, p, mcfg);
rc = gpr_send_port_pkt(graph->port, pkt);
kfree(pkt);
--
2.25.1
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