Merge "hal: add support for low-latency capture" (78227db6) · Commits · e / devices / Fairphone 4 / android_hardware_qcom_audio

hal/audio_hw.c

+75 −20

Original line number	Diff line number	Diff line
		@@ -63,6 +63,9 @@

		#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER

		static unsigned int configured_low_latency_capture_period_size =
		LOW_LATENCY_CAPTURE_PERIOD_SIZE;

		struct pcm_config pcm_config_deep_buffer = {
		.channels = 2,
		.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
		@@ -288,6 +291,19 @@ static int enable_audio_route_for_voice_usecases(struct audio_device *adev,
		return 0;
		}

		int pcm_ioctl(struct pcm *pcm, int request, ...)
		{
		va_list ap;
		void * arg;
		int pcm_fd = (int)pcm;

		va_start(ap, request);
		arg = va_arg(ap, void *);
		va_end(ap);

		return ioctl(pcm_fd, request, arg);
		}

		int enable_audio_route(struct audio_device *adev,
		struct audio_usecase *usecase)
		{
		@@ -1439,7 +1455,8 @@ static int check_input_parameters(uint32_t sample_rate,

		static size_t get_input_buffer_size(uint32_t sample_rate,
		audio_format_t format,
		int channel_count)
		int channel_count,
		bool is_low_latency)
		{
		size_t size = 0;

		@@ -1447,13 +1464,19 @@ static size_t get_input_buffer_size(uint32_t sample_rate,
		return 0;

		size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
		if (is_low_latency)
		size = configured_low_latency_capture_period_size;
		/* ToDo: should use frame_size computed based on the format and
		channel_count here. */
		size = sizeof(short) channel_count;

		/* make sure the size is multiple of 64 */
		size += 0x3f;
		size &= ~0x3f;
		/* make sure the size is multiple of 32 bytes
		* At 48 kHz mono 16-bit PCM:
		* 5.000 ms = 240 frames = 15161*2 = 480, a whole multiple of 32 (15)
		* 3.333 ms = 160 frames = 10161*2 = 320, a whole multiple of 32 (10)
		*/
		size += 0x1f;
		size &= ~0x1f;

		return size;
		}
		@@ -2970,7 +2993,8 @@ static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev __unu
		{
		int channel_count = audio_channel_count_from_in_mask(config->channel_mask);

		return get_input_buffer_size(config->sample_rate, config->format, channel_count);
		return get_input_buffer_size(config->sample_rate, config->format, channel_count,
		false /* is_low_latency: since we don't know, be conservative */);
		}

		static int adev_open_input_stream(struct audio_hw_device *dev,
		@@ -2986,7 +3010,7 @@ static int adev_open_input_stream(struct audio_hw_device *dev,
		struct stream_in *in;
		int ret = 0, buffer_size, frame_size;
		int channel_count = audio_channel_count_from_in_mask(config->channel_mask);

		bool is_low_latency = false;

		*stream_in = NULL;
		if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
		@@ -3030,6 +3054,13 @@ static int adev_open_input_stream(struct audio_hw_device *dev,

		/* Update config params with the requested sample rate and channels */
		in->usecase = USECASE_AUDIO_RECORD;
		if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
		(flags & AUDIO_INPUT_FLAG_FAST) != 0) {
		is_low_latency = true;
		#if LOW_LATENCY_CAPTURE_USE_CASE
		in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
		#endif
		}
		in->config = pcm_config_audio_capture;
		in->config.rate = config->sample_rate;
		in->format = config->format;
		@@ -3053,7 +3084,8 @@ static int adev_open_input_stream(struct audio_hw_device *dev,
		frame_size = audio_stream_in_frame_size(&in->stream);
		buffer_size = get_input_buffer_size(config->sample_rate,
		config->format,
		channel_count);
		channel_count,
		is_low_latency);
		in->config.period_size = buffer_size / frame_size;
		}

		@@ -3135,6 +3167,23 @@ static int adev_close(hw_device_t *device)
		return 0;
		}

		/* This returns 1 if the input parameter looks at all plausible as a low latency period size,
		* or 0 otherwise. A return value of 1 doesn't mean the value is guaranteed to work,
		* just that it _might_ work.
		*/
		static int period_size_is_plausible_for_low_latency(int period_size)
		{
		switch (period_size) {
		case 160:
		case 240:
		case 320:
		case 480:
		return 1;
		default:
		return 0;
		}
		}

		static int adev_open(const hw_module_t module, const char name,
		hw_device_t **device)
		{
		@@ -3258,23 +3307,29 @@ static int adev_open(const hw_module_t module, const char name,
		&adev->streams_output_cfg_list);

		audio_device_ref_count++;
		pthread_mutex_unlock(&adev_init_lock);

		ALOGV("%s: exit", __func__);
		return 0;
		char value[PROPERTY_VALUE_MAX];
		int trial;
		if (property_get("audio_hal.period_size", value, NULL) > 0) {
		trial = atoi(value);
		if (period_size_is_plausible_for_low_latency(trial)) {
		pcm_config_low_latency.period_size = trial;
		pcm_config_low_latency.start_threshold = trial / 4;
		pcm_config_low_latency.avail_min = trial / 4;
		configured_low_latency_capture_period_size = trial;
		}
		}
		if (property_get("audio_hal.in_period_size", value, NULL) > 0) {
		trial = atoi(value);
		if (period_size_is_plausible_for_low_latency(trial)) {
		configured_low_latency_capture_period_size = trial;
		}
		}

		int pcm_ioctl(struct pcm *pcm, int request, ...)
		{
		va_list ap;
		void * arg;
		int pcm_fd = (int)pcm;

		va_start(ap, request);
		arg = va_arg(ap, void *);
		va_end(ap);
		pthread_mutex_unlock(&adev_init_lock);

		return ioctl(pcm_fd, request, arg);
		ALOGV("%s: exit", __func__);
		return 0;
		}

		static struct hw_module_methods_t hal_module_methods = {

hal/audio_hw.h

+0 −3

Original line number	Diff line number	Diff line
		@@ -32,9 +32,6 @@
		#define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
		#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"

		#define BT_SCO_SAMPLE_RATE "bt-sco-samplerate"
		#define BT_SCO_WB_SAMPLE_RATE "bt-sco-wb-samplerate"

		/* Flags used to initialize acdb_settings variable that goes to ACDB library */
		#define NONE_FLAG 0x00000000
		#define ANC_FLAG 0x00000001

hal/msm8916/platform.h

+4 −0

Original line number	Diff line number	Diff line
		@@ -164,6 +164,10 @@ enum {
		#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
		#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2

		#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
		#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
		#define LOW_LATENCY_CAPTURE_USE_CASE 0

		#define HDMI_MULTI_PERIOD_SIZE 336
		#define HDMI_MULTI_PERIOD_COUNT 8
		#define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6

hal/msm8960/platform.h

+4 −0

Original line number	Diff line number	Diff line
		@@ -138,4 +138,8 @@ enum {
		/* Define macro for Internal FM volume mixer */
		#define FM_RX_VOLUME "Internal FM RX Volume"

		#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
		#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
		#define LOW_LATENCY_CAPTURE_USE_CASE 0

		#endif // QCOM_AUDIO_PLATFORM_H

hal/msm8974/platform.h

+4 −0

Original line number	Diff line number	Diff line
		@@ -178,6 +178,10 @@ enum {
		#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
		#define AUDIO_CAPTURE_PERIOD_COUNT 2

		#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
		#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
		#define LOW_LATENCY_CAPTURE_USE_CASE 1

		#define DEVICE_NAME_MAX_SIZE 128
		#define HW_INFO_ARRAY_MAX_SIZE 32