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Message-Id: <20230609145407.18774-9-srinivas.kandagatla@linaro.org>
Date: Fri, 9 Jun 2023 15:54:04 +0100
From: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
To: broonie@...nel.org
Cc: perex@...ex.cz, tiwai@...e.com, lgirdwood@...il.com,
ckeepax@...nsource.cirrus.com, kuninori.morimoto.gx@...esas.com,
linux-kernel@...r.kernel.org, pierre-louis.bossart@...ux.intel.com,
alsa-devel@...a-project.org,
Srinivas Kandagatla <srinivas.kandagatla@...aro.org>,
Mohammad Rafi Shaik <quic_mohs@...cinc.com>
Subject: [PATCH v2 08/11] ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks
Add q6apm get compress DAI capabilities and codec capabilities callbacks
to support compress offload playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Co-developed-by: Mohammad Rafi Shaik <quic_mohs@...cinc.com>
Signed-off-by: Mohammad Rafi Shaik <quic_mohs@...cinc.com>
---
sound/soc/qcom/qdsp6/q6apm-dai.c | 53 ++++++++++++++++++++++++++++++++
1 file changed, 53 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 32df5db014d3..d43705bf523a 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -30,8 +30,25 @@
#define BUFFER_BYTES_MIN (PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE)
#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
+#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
+#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
#define SID_MASK_DEFAULT 0xF
+static const struct snd_compr_codec_caps q6apm_compr_caps = {
+ .num_descriptors = 1,
+ .descriptor[0].max_ch = 2,
+ .descriptor[0].sample_rates = { 8000, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000, 88200,
+ 96000, 176400, 192000 },
+ .descriptor[0].num_sample_rates = 13,
+ .descriptor[0].bit_rate[0] = 320,
+ .descriptor[0].bit_rate[1] = 128,
+ .descriptor[0].num_bitrates = 2,
+ .descriptor[0].profiles = 0,
+ .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
+ .descriptor[0].formats = 0,
+};
+
enum stream_state {
Q6APM_STREAM_IDLE = 0,
Q6APM_STREAM_STOPPED,
@@ -41,6 +58,7 @@ enum stream_state {
struct q6apm_dai_rtd {
struct snd_pcm_substream *substream;
struct snd_compr_stream *cstream;
+ struct snd_codec codec;
struct snd_compr_params codec_param;
struct snd_dma_buffer dma_buffer;
phys_addr_t phys;
@@ -54,6 +72,7 @@ struct q6apm_dai_rtd {
uint16_t bits_per_sample;
uint16_t source; /* Encoding source bit mask */
uint16_t session_id;
+ bool next_track;
enum stream_state state;
struct q6apm_graph *graph;
spinlock_t lock;
@@ -517,9 +536,43 @@ static int q6apm_dai_compr_free(struct snd_soc_component *component,
return 0;
}
+
+static int q6apm_dai_compr_get_caps(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_caps *caps)
+{
+ caps->direction = SND_COMPRESS_PLAYBACK;
+ caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
+ caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
+ caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
+ caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+ caps->num_codecs = 3;
+ caps->codecs[0] = SND_AUDIOCODEC_MP3;
+ caps->codecs[1] = SND_AUDIOCODEC_AAC;
+ caps->codecs[2] = SND_AUDIOCODEC_FLAC;
+
+ return 0;
+}
+
+static int q6apm_dai_compr_get_codec_caps(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_codec_caps *codec)
+{
+ switch (codec->codec) {
+ case SND_AUDIOCODEC_MP3:
+ *codec = q6apm_compr_caps;
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
static const struct snd_compress_ops q6apm_dai_compress_ops = {
.open = q6apm_dai_compr_open,
.free = q6apm_dai_compr_free,
+ .get_caps = q6apm_dai_compr_get_caps,
+ .get_codec_caps = q6apm_dai_compr_get_codec_caps,
};
static const struct snd_soc_component_driver q6apm_fe_dai_component = {
--
2.21.0
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