audio: Enable 24 bit packed direct pcm support.
-Add support for 24 bit packed audio in audio hal.
-Disable gapless for AV playback and direct pcm usecase,
this ensures that the buffering in DSP is not more.
-Simulate rendered time stamp for direct pcm usecase
based on the number of frames written to the compress
driver, bufferring in the driver and DSP latency.
-Pass mixer instance to offload effects library to avoid
an unnecessary mixer_open call, this optimizes audio
start delay in compress playback.
Change-Id: I422a53af5632eaf6cc362a6c44f62ff8412965f7
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 857d9e1..46611fc 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -760,6 +760,7 @@
#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
+#define PCM_OFFLOAD_PLATFORM_DELAY (30*1000LL)
#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
static bool is_misc_usecase(audio_usecase_t usecase) {
@@ -3651,7 +3652,7 @@
free(kv_pairs);
}
-/* Delay in Us */
+/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
switch (usecase) {
@@ -3659,6 +3660,9 @@
return DEEP_BUFFER_PLATFORM_DELAY;
case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
return LOW_LATENCY_PLATFORM_DELAY;
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD:
+ case USECASE_AUDIO_PLAYBACK_OFFLOAD2:
+ return PCM_OFFLOAD_PLATFORM_DELAY;
default:
return 0;
}
@@ -3846,19 +3850,17 @@
uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info)
{
uint32_t fragment_size = 0;
- uint32_t bits_per_sample = 16;
+ uint32_t bytes_per_sample;
uint32_t pcm_offload_time = PCM_OFFLOAD_BUFFER_DURATION;
- if (info->format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) {
- bits_per_sample = 32;
- }
+ bytes_per_sample = audio_bytes_per_sample(info->format);
//duration is set to 40 ms worth of stereo data at 48Khz
//with 16 bit per sample, modify this when the channel
//configuration is different
fragment_size = (pcm_offload_time
* info->sample_rate
- * (bits_per_sample >> 3)
+ * bytes_per_sample
* popcount(info->channel_mask))/1000;
if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
@@ -3867,23 +3869,18 @@
// To have same PCM samples for all channels, the buffer size requires to
// be multiple of (number of channels * bytes per sample)
// For writes to succeed, the buffer must be written at address which is multiple of 32
- fragment_size = ALIGN(fragment_size, ((bits_per_sample >> 3)* popcount(info->channel_mask) * 32));
+ fragment_size = ALIGN(fragment_size, (bytes_per_sample * popcount(info->channel_mask) * 32));
ALOGI("PCM offload Fragment size to %d bytes", fragment_size);
return fragment_size;
}
-bool platform_use_small_buffer(audio_offload_info_t* info)
-{
- return OFFLOAD_USE_SMALL_BUFFER;
-}
-
/*
* configures afe with bit width and Sample Rate
*/
static int platform_set_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- unsigned int bit_width, unsigned int sample_rate)
+ snd_device_t snd_device, unsigned int bit_width,
+ unsigned int sample_rate, audio_format_t format)
{
int ret = 0;
int backend_idx = DEFAULT_CODEC_BACKEND;
@@ -3908,13 +3905,17 @@
}
if (bit_width == 24) {
+ if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
+ mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+ else
mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else {
mixer_ctl_set_enum_by_string(ctl, "S16_LE");
}
my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:becf: afe: %s mixer set to %d bit", __func__,
- my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width);
+ ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl,
+ bit_width, format);
}
/*
@@ -4150,11 +4151,13 @@
int new_snd_devices[SND_DEVICE_OUT_END];
int i, num_devices = 1;
bool ret = false;
+ audio_format_t format;
backend_idx = platform_get_backend_index(snd_device);
new_bit_width = usecase->stream.out->bit_width;
new_sample_rate = usecase->stream.out->sample_rate;
+ format = usecase->stream.out->format;
ALOGI("%s:becf: afe: bitwidth %d, samplerate %d"
", backend_idx %d usecase = %d device (%s)", __func__, new_bit_width,
@@ -4171,7 +4174,7 @@
if (platform_check_codec_backend_cfg(adev, usecase, new_snd_devices[i],
&new_bit_width, &new_sample_rate)) {
platform_set_codec_backend_cfg(adev, new_snd_devices[i],
- new_bit_width, new_sample_rate);
+ new_bit_width, new_sample_rate, format);
ret = true;
}
}
@@ -4691,8 +4694,8 @@
format = DTS_HD;
break;
case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD:
- case AUDIO_FORMAT_PCM_24_BIT_OFFLOAD:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_8_24_BIT:
ALOGV("%s:PCM", __func__);
format = LPCM;
break;