alsa: send the paramters to AoC for interrupt-drivenandroid-15-beta-2_r0.8android-15-beta-2_r0.7android-gs-shusky-5.15-android15-betaandroid-gs-akita-5.15-android15-beta

1. Sending the buffer size and period size of each entry point
   to AoC for interrupt-driven.
   AoC will set the thresold of the read/write pointer in the
   ring buffer according to the parameters.
   The detail information is in go/audio-int-driven-v2
2. Enlarge period_bytes_max of snd_aoc_playback_hw

Bug: 200718439
Bug: 249436330
Test: PCM playback, PCM recording, hifi, voip, incallscren
Change-Id: If7753e62104452d0eafc635db9b6c5721dfa6215
Signed-off-by: Miller Liang <millerliang@google.com>

5 files changed, 112 insertions, 6 deletions

diff --git a/alsa/aoc_alsa.h b/alsa/aoc_alsa.h
index 24a21ed..c7ea617 100644
--- a/alsa/aoc_alsa.h
+++ b/alsa/aoc_alsa.h
@@ -388,6 +388,11 @@ int aoc_audio_close(struct aoc_alsa_stream *alsa_stream);
 int aoc_audio_set_params(struct aoc_alsa_stream *alsa_stream, uint32_t channels,
 			 uint32_t samplerate, uint32_t bps, bool pcm_float_fmt, int source_mode);
 
+#if !IS_ENABLED(CONFIG_SOC_GS101)
+int aoc_voip_set_params(struct aoc_alsa_stream *alsa_stream);
+int aoc_hifi_incall_set_params(struct aoc_alsa_stream *alsa_stream);
+#endif
+
 int aoc_audio_start(struct aoc_alsa_stream *alsa_stream);
 int aoc_audio_stop(struct aoc_alsa_stream *alsa_stream);
 int aoc_audio_incall_start(struct aoc_alsa_stream *alsa_stream);
diff --git a/alsa/aoc_alsa_hw.c b/alsa/aoc_alsa_hw.c
index 1057349..4552841 100644
--- a/alsa/aoc_alsa_hw.c
+++ b/alsa/aoc_alsa_hw.c
@@ -1670,6 +1670,46 @@ int aoc_audio_path_close(struct aoc_chip *chip, int src, int dest, bool be_on)
 			 hw_id_to_sink(dest_idx), STOP, chip);
 }
 
+#if !IS_ENABLED(CONFIG_SOC_GS101)
+static int aoc_audio_playback_set_params2(struct aoc_alsa_stream *alsa_stream, int source_mode)
+{
+	struct CMD_AUDIO_OUTPUT_EP_SETUP2 cmd;
+
+	AocCmdHdrSet(&(cmd.parent), CMD_AUDIO_OUTPUT_EP_SETUP2_ID, sizeof(cmd));
+	cmd.channel = alsa_stream->entry_point_idx;
+
+	cmd.buffer_size = alsa_stream->buffer_size;
+	cmd.period_size = alsa_stream->period_size;
+
+	cmd.mode = source_mode;
+
+	pr_debug("audio set param2:idx %d, buffer_size=%d, period_size=%d\n", cmd.channel,
+				cmd.buffer_size, cmd.period_size);
+	return aoc_audio_control(CMD_OUTPUT_CHANNEL, (uint8_t *)&cmd,
+				sizeof(cmd), NULL, alsa_stream->chip);
+}
+
+int aoc_hifi_incall_set_params(struct aoc_alsa_stream *alsa_stream)
+{
+	int source_mode;
+
+	if (alsa_stream->stream_type == INCALL)
+		source_mode = ENTRYPOINT_MODE_INCALL_SCREEN;
+	else
+		source_mode = ENTRYPOINT_MODE_HIFI;
+
+	return aoc_audio_playback_set_params2(alsa_stream, source_mode);
+}
+
+int aoc_voip_set_params(struct aoc_alsa_stream *alsa_stream)
+{
+	int source_mode;
+
+	source_mode = ENTRYPOINT_MODE_VOIP;
+	return aoc_audio_playback_set_params2(alsa_stream, source_mode);
+}
+#endif
+
 static int aoc_audio_playback_set_params(struct aoc_alsa_stream *alsa_stream,
 					 uint32_t channels, uint32_t samplerate,
 					 uint32_t bps, bool pcm_float_fmt,
@@ -1743,12 +1783,43 @@ static int aoc_audio_playback_set_params(struct aoc_alsa_stream *alsa_stream,
 
 	err = aoc_audio_control(CMD_OUTPUT_CHANNEL, (uint8_t *)&cmd,
 				sizeof(cmd), NULL, alsa_stream->chip);
-	if (err < 0)
+	if (err < 0) {
 		pr_err("ERR:%d in playback set parameters\n", err);
+		goto exit;
+	}
+
+#if !IS_ENABLED(CONFIG_SOC_GS101)
+	err = aoc_audio_playback_set_params2(alsa_stream, cmd.mode);
+	if (err < 0)
+		pr_err("ERR:%d in playback set parameters2\n", err);
+#endif
 
+exit:
 	return err;
 }
 
+#if !IS_ENABLED(CONFIG_SOC_GS101)
+static int aoc_audio_capture_set_params2(struct aoc_alsa_stream *alsa_stream) {
+	struct CMD_AUDIO_INPUT_MIC_RECORD_AP_SET_PARAMS2 cmd;
+	int cmd_id;
+
+	cmd.channel = alsa_stream->idx;
+	cmd.buffer_size = alsa_stream->buffer_size;
+	cmd.period_size = alsa_stream->period_size;
+
+	if (alsa_stream->idx == UC_ULTRASONIC_RECORD)
+		cmd_id = CMD_AUDIO_INPUT_ULTRASONIC_AP_SET_PARAMS2_ID;
+	else
+		cmd_id = CMD_AUDIO_INPUT_MIC_RECORD_AP_SET_PARAMS2_ID;
+	AocCmdHdrSet(&(cmd.parent), cmd_id, sizeof(cmd));
+
+	pr_debug("audio set param2:idx %d, buffer_size=%d, period_size=%d\n", cmd.channel,
+				cmd.buffer_size, cmd.period_size);
+	return aoc_audio_control(CMD_INPUT_CHANNEL, (uint8_t *)&cmd,
+				sizeof(cmd), NULL, alsa_stream->chip);
+}
+#endif
+
 static int aoc_audio_capture_set_params(struct aoc_alsa_stream *alsa_stream, uint32_t channels,
 					uint32_t samplerate, uint32_t bps, bool pcm_float_fmt)
 {
@@ -1867,6 +1938,14 @@ static int aoc_audio_capture_set_params(struct aoc_alsa_stream *alsa_stream, uin
 		goto exit;
 	}
 
+#if !IS_ENABLED(CONFIG_SOC_GS101)
+	err = aoc_audio_capture_set_params2(alsa_stream);
+	if (err < 0) {
+		pr_err("ERR:%d in capture parameter2 setup\n", err);
+		goto exit;
+	}
+#endif
+
 	chip->capture_param_set |= (1 << alsa_stream->idx);
 
 	if (alsa_stream->idx == UC_ULTRASONIC_RECORD) {
diff --git a/alsa/aoc_alsa_incall.c b/alsa/aoc_alsa_incall.c
index 29e1bbd..7aa3404 100644
--- a/alsa/aoc_alsa_incall.c
+++ b/alsa/aoc_alsa_incall.c
@@ -397,12 +397,25 @@ static int snd_aoc_pcm_prepare(struct snd_soc_component *component,
 	struct aoc_service_dev *dev = alsa_stream->dev;
 	struct aoc_chip *chip = alsa_stream->chip;
 	int channels;
+#if !IS_ENABLED(CONFIG_SOC_GS101)
+	int err;
+#endif
 
 	aoc_timer_stop_sync(alsa_stream);
 
 	if (mutex_lock_interruptible(&chip->audio_mutex))
 		return -EINTR;
 
+	alsa_stream->buffer_size = snd_pcm_lib_buffer_bytes(substream);
+	alsa_stream->period_size = snd_pcm_lib_period_bytes(substream);
+
+#if !IS_ENABLED(CONFIG_SOC_GS101)
+	/* Set the audio formats and flush the DRAM buffer */
+	err = aoc_hifi_incall_set_params(alsa_stream);
+	if (err < 0)
+		pr_notice("Failed to set %d Hifi/Incall params\n", err);
+#endif
+
 	channels = alsa_stream->channels;
 
 	dev_dbg(component->dev, "channels = %d, rate = %d, bits = %d, float-fmt = %d\n", channels,
@@ -413,8 +426,6 @@ static int snd_aoc_pcm_prepare(struct snd_soc_component *component,
 
 	/* in preparation of the stream */
 	/* aoc_audio_set_ctls(alsa_stream->chip); */
-	alsa_stream->buffer_size = snd_pcm_lib_buffer_bytes(substream);
-	alsa_stream->period_size = snd_pcm_lib_period_bytes(substream);
 	alsa_stream->pos = 0;
 	alsa_stream->prev_pos = 0;
 	alsa_stream->pos_delta = 0;
diff --git a/alsa/aoc_alsa_pcm.c b/alsa/aoc_alsa_pcm.c
index 98d4bae..31ebaa9 100644
--- a/alsa/aoc_alsa_pcm.c
+++ b/alsa/aoc_alsa_pcm.c
@@ -183,7 +183,7 @@ static struct snd_pcm_hardware snd_aoc_playback_hw = {
 	.channels_max = 6,
 	.buffer_bytes_max = 16384 * 6,
 	.period_bytes_min = 16,
-	.period_bytes_max = 11520,
+	.period_bytes_max = 15360,
 	.periods_min = 2,
 	.periods_max = 1024 * 6,
 };
@@ -568,6 +568,9 @@ static int snd_aoc_pcm_prepare(struct snd_soc_component *component,
 		}
 	}
 
+	alsa_stream->buffer_size = snd_pcm_lib_buffer_bytes(substream);
+	alsa_stream->period_size = snd_pcm_lib_period_bytes(substream);
+
 	/* Set the audio formats and flush the DRAM buffer */
 	err = aoc_audio_set_params(alsa_stream, channels, alsa_stream->params_rate,
 				   alsa_stream->pcm_format_width, alsa_stream->pcm_float_fmt,
@@ -585,8 +588,6 @@ static int snd_aoc_pcm_prepare(struct snd_soc_component *component,
 
 	/* in preparation of the stream */
 	/* aoc_audio_set_ctls(alsa_stream->chip); */
-	alsa_stream->buffer_size = snd_pcm_lib_buffer_bytes(substream);
-	alsa_stream->period_size = snd_pcm_lib_period_bytes(substream);
 	alsa_stream->pos = 0;
 	alsa_stream->prev_pos = 0;
 	alsa_stream->pos_delta = 0;
diff --git a/alsa/aoc_alsa_voip.c b/alsa/aoc_alsa_voip.c
index f9f85ba..b186330 100644
--- a/alsa/aoc_alsa_voip.c
+++ b/alsa/aoc_alsa_voip.c
@@ -376,6 +376,16 @@ static int snd_aoc_pcm_prepare(struct snd_soc_component *component,
 	if (mutex_lock_interruptible(&chip->audio_mutex))
 		return -EINTR;
 
+	alsa_stream->buffer_size = snd_pcm_lib_buffer_bytes(substream);
+	alsa_stream->period_size = snd_pcm_lib_period_bytes(substream);
+
+#if !IS_ENABLED(CONFIG_SOC_GS101)
+	/* Set the audio formats and flush the DRAM buffer */
+	err = aoc_voip_set_params(alsa_stream);
+	if (err < 0)
+		pr_notice("Failed to set %d VoIP Rx params\n", err);
+#endif
+
 	channels = alsa_stream->channels;
 
 	pr_debug("channels = %d, rate = %d, bits = %d, float-fmt = %d\n",