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Commit 36f3d378 authored by Preetam Singh Ranawat's avatar Preetam Singh Ranawat
Browse files

hal: native audio backward compatibility 

- all changes for native audio backward
  compatibility and related.
- add set parameters from UI
- enable dynamic device switching
- featurize the code
- various backend concurrency changes.

Change-Id: Id0f824c4b4c033f42008a4e8868652c8f6fe5c42
parent 1871ddbe

hal/audio_extn/utils.c

+5 −0
Original line number Diff line number Diff line
@@ -574,6 +574,10 @@ int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase) 
    if ((24 == usecase->stream.out->bit_width) &&

        (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {

        sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;

    } else if ((snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&

        usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) ||

        (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {

        sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;

    } else {

        sample_rate = out->app_type_cfg.sample_rate;

    }
@@ -594,6 +598,7 @@ int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase) 
        app_type_cfg[len++] = sample_rate * 4;

    else

        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, sample_rate);

hal/audio_hw.c

+19 −18
Original line number Diff line number Diff line
@@ -582,6 +582,8 @@ static void check_usecases_codec_backend(struct audio_device *adev, 
    struct audio_usecase *usecase;

    bool switch_device[AUDIO_USECASE_MAX];

    int i, num_uc_to_switch = 0;

    int backend_idx = DEFAULT_CODEC_BACKEND;

    int usecase_backend_idx = DEFAULT_CODEC_BACKEND;



    /*

     * This function is to make sure that all the usecases that are active on
@@ -598,39 +600,38 @@ static void check_usecases_codec_backend(struct audio_device *adev, 
     * If there is a backend configuration change for the device when a

     * new stream starts, then ADM needs to be closed and re-opened with the new

     * configuraion. This call check if we need to re-route all the streams

     * associated with the backend. Touch tone + 24 bit playback.

     * associated with the backend. Touch tone + 24 bit + native playback.

     */

    bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info);



    bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info,

                         snd_device);

    backend_idx = platform_get_backend_index(snd_device);

    /* Disable all the usecases on the shared backend other than the

     * specified usecase.

     * For native(44.1k) usecases, we don't need this as it uses a different

     * backend, but we need to make sure that we reconfigure the backend

     * if there is bit_width change, this should not affect shared backend

     * usecases.

     */

    for (i = 0; i < AUDIO_USECASE_MAX; i++)

        switch_device[i] = false;



    list_for_each(node, &adev->usecase_list) {

        usecase = node_to_item(node, struct audio_usecase, list);



        if (usecase == uc_info)

            continue;

        usecase_backend_idx = platform_get_backend_index(usecase->out_snd_device);

        ALOGV("%s: backend_idx: %d,"

              "usecase_backend_idx: %d, curr device: %s, usecase device:"

              "%s", __func__, backend_idx, usecase_backend_idx, platform_get_snd_device_name(snd_device),

            platform_get_snd_device_name(usecase->out_snd_device));



        if (usecase->type != PCM_CAPTURE &&

                usecase != uc_info &&

                (usecase->out_snd_device != snd_device || force_routing)  &&

                usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND

                && usecase->stream.out->sample_rate != OUTPUT_SAMPLING_RATE_44100) {

            ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",

                  __func__, use_case_table[usecase->id],

                usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND &&

                usecase_backend_idx == backend_idx) {

            ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", __func__,

                  use_case_table[usecase->id],

                  platform_get_snd_device_name(usecase->out_snd_device));

            disable_audio_route(adev, usecase);

            switch_device[usecase->id] = true;

            num_uc_to_switch++;

        } else if (usecase->type == PCM_PLAYBACK &&

                  usecase->stream.out->sample_rate ==

                  OUTPUT_SAMPLING_RATE_44100 && force_routing){

           disable_audio_route(adev, usecase);

           switch_device[usecase->id] = true;

           num_uc_to_switch++;

        }

    }

hal/audio_hw.h

+2 −0
Original line number Diff line number Diff line
@@ -309,6 +309,8 @@ struct audio_device { 
    bool bt_wb_speech_enabled;



    int snd_card;

    unsigned int cur_codec_backend_samplerate;

    unsigned int cur_codec_backend_bit_width;

    void *platform;

    unsigned int offload_usecases_state;

    void *visualizer_lib;

hal/msm8916/platform.c

+15 −2
Original line number Diff line number Diff line
@@ -1730,6 +1730,16 @@ int platform_get_snd_device_bit_width(snd_device_t snd_device) 
    return backend_bit_width_table[snd_device];

}



int platform_set_native_support(bool codec_support __unused)

{

    return 0;

}



int platform_get_backend_index(snd_device_t snd_device __unused)

{

    return 0;

}



int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,

                                    int app_type, int sample_rate)

{
@@ -2016,12 +2026,13 @@ int platform_set_device_mute(void *platform, bool state, char *dir) 
    return ret;

}



snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices)

snd_device_t platform_get_output_snd_device(void *platform, struct stream_out *out)

{

    struct platform_data *my_data = (struct platform_data *)platform;

    struct audio_device *adev = my_data->adev;

    audio_mode_t mode = adev->mode;

    snd_device_t snd_device = SND_DEVICE_NONE;

    audio_devices_t devices = out->devices;

#ifdef RECORD_PLAY_CONCURRENCY

    bool use_voip_out_devices = false;

    bool prop_rec_play_enabled = false;
@@ -3553,7 +3564,9 @@ bool platform_check_codec_backend_cfg(struct audio_device* adev, 
    return backend_change;

}



bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev, struct audio_usecase *usecase)

bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev,

                                              struct audio_usecase *usecase,

                                              snd_device_t snd_device __unused)

{

    ALOGV("platform_check_and_set_codec_backend_cfg usecase = %d",usecase->id );

hal/msm8916/platform.h

+8 −0
Original line number Diff line number Diff line
@@ -60,6 +60,7 @@ enum { 
    SND_DEVICE_OUT_SPEAKER_REVERSE,

    SND_DEVICE_OUT_SPEAKER_WSA,

    SND_DEVICE_OUT_HEADPHONES,

    SND_DEVICE_OUT_HEADPHONES_44_1,

    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,

    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1,

    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2,
@@ -162,6 +163,13 @@ enum { 
};



#define DEFAULT_OUTPUT_SAMPLING_RATE 48000

#define OUTPUT_SAMPLING_RATE_44100   44100



enum {

    DEFAULT_CODEC_BACKEND,

    HEADPHONE_44_1_BACKEND,

    MAX_CODEC_BACKENDS

};



#define ALL_SESSION_VSID                0xFFFFFFFF

#define DEFAULT_MUTE_RAMP_DURATION_MS   20