Merge "hal: Fix combo device configuration for 24 bit playback on speaker" (9d1bcfbc) · Commits · e / os / android_hardware_qcom_audio

hal/audio_extn/audio_extn.h

+1 −0

Original line number	Diff line number	Diff line
		@@ -309,6 +309,7 @@ void audio_extn_utils_release_streams_output_cfg_list(
		struct listnode *streams_output_cfg_list);
		void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		struct listnode *streams_output_cfg_list,
		audio_devices_t devices,
		audio_output_flags_t flags,
		audio_format_t format,
		uint32_t sample_rate,

hal/audio_extn/utils.c

+10 −2

Original line number	Diff line number	Diff line
		@@ -422,6 +422,7 @@ static bool set_output_cfg(struct streams_output_cfg *so_info,

		void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		struct listnode *streams_output_cfg_list,
		audio_devices_t devices,
		audio_output_flags_t flags,
		audio_format_t format,
		uint32_t sample_rate,
		@@ -433,7 +434,14 @@ void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		struct stream_format *sf_info;
		struct stream_sample_rate *ss_info;

		ALOGV("%s: flags: %x, format: %x", __func__, flags, format);
		if ((24 == bit_width) &&
		(devices & AUDIO_DEVICE_OUT_SPEAKER)) {
		sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		ALOGI("%s Allowing 24-bit playback on speaker ONLY at default sampling rate", __func__);
		}

		ALOGV("%s: flags: %x, format: %x sample_rate %d",
		__func__, flags, format, sample_rate);
		list_for_each(node_i, streams_output_cfg_list) {
		so_info = node_to_item(node_i, struct streams_output_cfg, list);
		if (so_info->flags == flags) {
		@@ -516,7 +524,7 @@ int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase)
		}

		if ((24 == usecase->stream.out->bit_width) &&
		(AUDIO_DEVICE_OUT_SPEAKER == snd_device)) {
		(usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
		sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		} else {
		sample_rate = out->app_type_cfg.sample_rate;

hal/audio_hw.c

+7 −26

Original line number	Diff line number	Diff line
		@@ -874,26 +874,17 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
		usecase->out_snd_device = out_snd_device;

		if (usecase->type == PCM_PLAYBACK) {
		if ((24 == usecase->stream.out->bit_width) &&
		(AUDIO_DEVICE_OUT_SPEAKER == usecase->stream.out->devices)) {
		audio_extn_utils_update_stream_app_type_cfg(adev->platform,
		&adev->streams_output_cfg_list,
		usecase->stream.out->flags,
		usecase->stream.out->format,
		DEFAULT_OUTPUT_SAMPLING_RATE,
		usecase->stream.out->bit_width,
		&usecase->stream.out->app_type_cfg);
		} else {
		audio_extn_utils_update_stream_app_type_cfg(adev->platform,
		&adev->streams_output_cfg_list,
		usecase->stream.out->devices,
		usecase->stream.out->flags,
		usecase->stream.out->format,
		usecase->stream.out->sample_rate,
		usecase->stream.out->bit_width,
		&usecase->stream.out->app_type_cfg);
		}
		ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
		}

		enable_audio_route(adev, usecase);

		/* Applicable only on the targets that has external modem.
		@@ -2527,7 +2518,6 @@ static int adev_open_output_stream(struct audio_hw_device *dev,
		struct stream_out *out;
		int i, ret = 0;
		audio_format_t format;
		int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;

		*stream_out = NULL;

		@@ -2730,20 +2720,11 @@ static int adev_open_output_stream(struct audio_hw_device *dev,
		out->sample_rate = out->config.rate;
		}

		if ((24 == out->bit_width) &&
		(devices == AUDIO_DEVICE_OUT_SPEAKER)) {
		sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		ALOGI("%s 24-bit playback on Speaker is allowed ONLY at 48khz. Hence changing sample rate to: %d",
		__func__, sample_rate);
		} else {
		sample_rate = out->sample_rate;
		}

		ALOGV("%s flags %x, format %x, sample_rate %d, out->bit_width %d",
		__func__, flags, format, sample_rate, out->bit_width);
		ALOGV("%s devices %d,flags %x, format %x, out->sample_rate %d, out->bit_width %d",
		__func__, devices, flags, format, out->sample_rate, out->bit_width);
		audio_extn_utils_update_stream_app_type_cfg(adev->platform,
		&adev->streams_output_cfg_list,
		flags, format, sample_rate,
		devices, flags, format, out->sample_rate,
		out->bit_width, &out->app_type_cfg);
		if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) \|\|
		(flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {

hal/msm8974/platform.c

+4 −2

Original line number	Diff line number	Diff line
		@@ -2666,9 +2666,11 @@ bool platform_check_codec_backend_cfg(struct audio_device* adev,
		}
		}

		// 16 bit playbacks is allowed through 16 bit/48 khz backend only
		// 24 bit playback on speakers and all 16 bit playbacks is allowed through
		// 16 bit/48 khz backend only
		if ((16 == bit_width) \|\|
		((24 == bit_width) && (SND_DEVICE_OUT_SPEAKER == usecase->devices))) {
		((24 == bit_width) &&
		(usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER))) {
		sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		}
		// Force routing if the expected bitwdith or samplerate