Merge "hal: Fix backend for 24 bit playback on speaker" (a959cbd0) · Commits · e / devices / Fairphone 4 / android_hardware_qcom_audio

hal/audio_extn/utils.c

+11 −2

Original line number	Diff line number	Diff line
		@@ -472,6 +472,7 @@ int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase)
		struct audio_device *adev;
		struct mixer_ctl *ctl;
		int pcm_device_id, acdb_dev_id, snd_device = usecase->out_snd_device;
		int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;

		ALOGV("%s", __func__);

		@@ -513,13 +514,21 @@ int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase)
		rc = -EINVAL;
		goto exit_send_app_type_cfg;
		}

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

		app_type_cfg[len++] = out->app_type_cfg.app_type;
		app_type_cfg[len++] = acdb_dev_id;
		app_type_cfg[len++] = out->app_type_cfg.sample_rate;
		app_type_cfg[len++] = sample_rate;

		mixer_ctl_set_array(ctl, app_type_cfg, len);
		ALOGI("%s app_type %d, acdb_dev_id %d, sample_rate %d",
		__func__, out->app_type_cfg.app_type, acdb_dev_id, out->app_type_cfg.sample_rate);
		__func__, out->app_type_cfg.app_type, acdb_dev_id, sample_rate);
		rc = 0;
		exit_send_app_type_cfg:
		return rc;

hal/audio_hw.c

+34 −3

Original line number	Diff line number	Diff line
		@@ -797,6 +797,27 @@ int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
		usecase->in_snd_device = in_snd_device;
		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->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.
		@@ -2382,6 +2403,7 @@ 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;

		@@ -2567,11 +2589,20 @@ static int adev_open_output_stream(struct audio_hw_device *dev,
		out->sample_rate = out->config.rate;
		}

		ALOGV("%s flags %x, format %x, out->sample_rate %d, out->bit_width %d",
		__func__, flags, format, out->sample_rate, out->bit_width);
		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);
		audio_extn_utils_update_stream_app_type_cfg(adev->platform,
		&adev->streams_output_cfg_list,
		flags, format, out->sample_rate,
		flags, format, 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

+3 −1

Original line number	Diff line number	Diff line
		@@ -2560,8 +2560,10 @@ bool platform_check_codec_backend_cfg(struct audio_device* adev,
		}

		// 16 bit playbacks is allowed through 16 bit/48 khz backend only
		if (16 == bit_width)
		if ((16 == bit_width) \|\|
		((24 == bit_width) && (SND_DEVICE_OUT_SPEAKER == usecase->devices))) {
		sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		}
		// Force routing if the expected bitwdith or samplerate
		// is not same as current backend comfiguration
		if ((bit_width != adev->cur_codec_backend_bit_width) \|\|