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Commit 4674047f authored by Haynes Mathew George's avatar Haynes Mathew George
Browse files

audio: Cleanup adev_open_input_stream 

Simplify logic a bit and also fix bug
where config for 24 bit recording was not
being set correctly

Change-Id: I48cbcb3a3775aa222bc8387e02795d4e4d234057
CRs-Fixed: 2133170
parent adbbe4f9

hal/audio_hw.c

+57 −52
Original line number Diff line number Diff line
@@ -933,7 +933,6 @@ static void check_and_enable_effect(struct audio_device *adev) 
#define check_and_enable_effect(x) ENOSYS

#endif





int pcm_ioctl(struct pcm *pcm, int request, ...)

{

    va_list ap;
@@ -6595,6 +6594,15 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 
    in->standby = 1;

    in->capture_handle = handle;

    in->flags = flags;

    in->bit_width = 16;

    in->af_period_multiplier = 1;



    /* Update config params with the requested sample rate and channels */

    if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) &&

          (adev->mode != AUDIO_MODE_IN_CALL)) {

        ret = -EINVAL;

        goto err_open;

    }



    /* restrict 24 bit capture for unprocessed source only

     * for other sources if 24 bit requested reject 24 and set 16 bit capture only
@@ -6617,7 +6625,9 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 
            goto err_open;

        }

        channel_count = audio_channel_count_from_in_mask(config->channel_mask);

    } else if (config->format == AUDIO_FORMAT_DEFAULT) {

    }



    if (config->format == AUDIO_FORMAT_DEFAULT) {

        config->format = AUDIO_FORMAT_PCM_16_BIT;

    } else if ((config->format == AUDIO_FORMAT_PCM_FLOAT) ||

               (config->format == AUDIO_FORMAT_PCM_32_BIT) ||
@@ -6640,11 +6650,8 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 
            if (config->sample_rate > 48000)

                config->sample_rate = 48000;

            ret_error = true;

        } else if (config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED) {

            in->config.format = PCM_FORMAT_S24_3LE;

        } else if (config->format == AUDIO_FORMAT_PCM_8_24_BIT) {

            in->config.format = PCM_FORMAT_S24_LE;

        } else {

        } else if (!(config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||

                     config->format == AUDIO_FORMAT_PCM_8_24_BIT)) {

            config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;

            ret_error = true;

        }
@@ -6673,32 +6680,18 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 
        in->realtime = 0;

        in->usecase = USECASE_AUDIO_RECORD_MMAP;

        in->config = pcm_config_mmap_capture;

        in->config.format = pcm_format_from_audio_format(config->format);

        in->stream.start = in_start;

        in->stream.stop = in_stop;

        in->stream.create_mmap_buffer = in_create_mmap_buffer;

        in->stream.get_mmap_position = in_get_mmap_position;

        in->af_period_multiplier = 1;

        ALOGV("%s: USECASE_AUDIO_RECORD_MMAP", __func__);

    } else if (in->realtime) {

        in->config = pcm_config_audio_capture_rt;

        in->config.format = pcm_format_from_audio_format(config->format);

        in->sample_rate = in->config.rate;

        in->af_period_multiplier = af_period_multiplier;

    } else {

        in->config = pcm_config_audio_capture;

        in->config.rate = config->sample_rate;

        in->sample_rate = config->sample_rate;

        in->af_period_multiplier = 1;

    }

    in->bit_width = 16;



    /* Update config params with the requested sample rate and channels */

    if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) &&

          (adev->mode != AUDIO_MODE_IN_CALL)) {

        ret = -EINVAL;

        goto err_open;

    }



    if (is_usb_dev && may_use_hifi_record) {

    } else if (is_usb_dev && may_use_hifi_record) {

        in->usecase = USECASE_AUDIO_RECORD_HIFI;

        in->config = pcm_config_audio_capture;

        frame_size = audio_stream_in_frame_size(&in->stream);
@@ -6708,7 +6701,6 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 
                                            false /*is_low_latency*/);

        in->config.period_size = buffer_size / frame_size;

        in->config.rate = config->sample_rate;

        in->af_period_multiplier = 1;

        in->config.format = pcm_format_from_audio_format(config->format);

        in->config.channels = channel_count;

    } else if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) ||
@@ -6751,41 +6743,54 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 
        ret = audio_extn_cin_configure_input_stream(in);

        if (ret)

            goto err_open;

    } else {

        in->config.channels = channel_count;

        if (!in->realtime) {

            in->format = config->format;

            frame_size = audio_stream_in_frame_size(&in->stream);

            buffer_size = get_input_buffer_size(config->sample_rate,

                                                config->format,

                                                channel_count,

                                                is_low_latency);

            in->config.period_size = buffer_size / frame_size;

        }



    } else if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {

        bool valid_rate = (config->sample_rate == 8000 ||

                           config->sample_rate == 16000 ||

                           config->sample_rate == 32000 ||

                           config->sample_rate == 48000);

        bool valid_ch = audio_channel_count_from_in_mask(in->channel_mask) == 1;

        //XXX needed for voice_extn_compress_voip_open_input_stream

        in->config.rate = config->sample_rate;

#ifndef COMPRESS_VOIP_ENABLED

        if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&

            (in->config.rate == 8000 || in->config.rate == 16000 ||

             in->config.rate == 32000 || in->config.rate == 48000) &&

            (audio_channel_count_from_in_mask(in->channel_mask) == 1)) {



        if (valid_rate && valid_ch) {

            in->usecase = USECASE_AUDIO_RECORD_VOIP;

            in->config = default_pcm_config_voip_copp;

            in->config.period_size = VOIP_IO_BUF_SIZE(in->sample_rate, DEFAULT_VOIP_BUF_DURATION_MS, DEFAULT_VOIP_BIT_DEPTH_BYTE)/2;

            in->config.rate = in->sample_rate;

            in->config.period_size = VOIP_IO_BUF_SIZE(in->sample_rate,

                                                      DEFAULT_VOIP_BUF_DURATION_MS,

                                                      DEFAULT_VOIP_BIT_DEPTH_BYTE)/2;

        }

#else

        if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&

               (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION ||

        if ((in->dev->mode == AUDIO_MODE_IN_COMMUNICATION ||

             voice_extn_compress_voip_is_active(in->dev)) &&

            (voice_extn_compress_voip_is_format_supported(in->format)) &&

               (in->config.rate == 8000 || in->config.rate == 16000 ||

                in->config.rate == 32000 || in->config.rate == 48000) &&

               (audio_channel_count_from_in_mask(in->channel_mask) == 1)) {

            valid_rate && valid_ch) {

            voice_extn_compress_voip_open_input_stream(in);

        }

#endif

        else {

            ALOGE("%s AUDIO_SOURCE_VOICE_COMMUNICATION invalid args", __func__);

            ret = -EINVAL;

            if (!valid_ch) config->channel_mask = 1;

            if (!valid_rate) config->sample_rate = 48000;

            goto err_open;

        }

        // update back to whatever was overwritten

        in->config.rate = config->sample_rate;

        in->sample_rate = config->sample_rate;

    } else {

        in->config = pcm_config_audio_capture;

        in->config.rate = config->sample_rate;

        in->config.format = pcm_format_from_audio_format(config->format);

        in->config.channels = channel_count;

        in->sample_rate = config->sample_rate;

        in->format = config->format;

        frame_size = audio_stream_in_frame_size(&in->stream);

        buffer_size = get_input_buffer_size(config->sample_rate,

                                            config->format,

                                            channel_count,

                                            is_low_latency);

        in->config.period_size = buffer_size / frame_size;

    }



    audio_extn_utils_update_stream_input_app_type_cfg(adev->platform,

                                                &adev->streams_input_cfg_list,

                                                devices, flags, in->format, in->sample_rate,