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Commit 1d89604d authored by Amit Shekhar's avatar Amit Shekhar
Browse files

hal: Fix combo device configuration for 24 bit playback on speaker 

-Fix combo device configuration for 24 bit playback on speaker
-Move exception hadndling code to audio_extn

Change-Id: Ia0985a284042a5ac5e3de64aaf5e4d57462ceb85
parent 987af8e8

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
@@ -822,26 +822,17 @@ 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.
@@ -2433,7 +2424,6 @@ 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;
@@ -2619,20 +2609,11 @@ 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
@@ -2559,9 +2559,11 @@ 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