msm8909(android O)CTA通话录音失败，生成0KB文件

最新推荐文章于 2023-02-25 19:40:57 发布

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最新推荐文章于 2023-02-25 19:40:57 发布

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分类专栏： android_audio

本文链接：https://blog.csdn.net/qq_28534581/article/details/89330508

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公司odm一款移动设备需要过CTA认证，发现通话录音不会弹出权限提示框并且生成0kb录音文件，故该测试项不通过。

通过log初步分析是获取输入设备属性返回-19导致初始化失败，从而无法录音：
03-29 16:05:32.005 454 6208 E AudioPolicyIntefaceImpl: inputType = 1
03-29 16:05:32.005 469 2397 E AudioRecord: openRecord_l status = -19

inputTyoe = 1表示
typedef enum {
API_INPUT_INVALID = -1,
API_INPUT_LEGACY = 0,// e.g. audio recording from a microphone
API_INPUT_MIX_CAPTURE,// used for “remote submix”, capture of the media to play it remotely
API_INPUT_MIX_EXT_POLICY_REROUTE,// used for platform audio rerouting, where mixes are
// handled by external and dynamically installed
// policies which reroute audio mixes
API_INPUT_TELEPHONY_RX, // used for capture from telephony RX path
} input_type_t;

进一步分析，注意加粗部分：
CTA异常：
03-29 16:05:31.843 418 4715 D audio_hw_primary: enable_audio_route: apply mixer and update path: voicemmode1-call
03-29 16:05:31.843 418 4715 D audio_route: Apply path: voicemmode1-call
03-29 16:05:31.848 418 4715 D audio_hw_primary: select_devices: done

03-29 16:05:31.972 418 4715 D voice : voice_set_sidetone: enable, out_snd_device: 16 (SND_DEVICE_OUT_VOICE_HANDSET， 有输出没输入)
03-29 16:05:31.972 418 4715 D audio_route: Apply path: sidetone-handset (免提)
03-29 16:05:31.977 418 4715 D msm8916_platform: platform_set_voice_volume Setting voice volume: 20
03-29 16:05:31.977 418 4715 D msm8916_platform: usecase_id = 34
03-29 16:05:31.977 418 4715 D msm8916_platform: usecase_id:voicemmode1-call

03-29 16:05:32.005 454 6208 E AudioPolicyIntefaceImpl: AudioPolicyService::getInputForAttr - E, attr->source = 1 (AUDIO_SOURCE_MIC)
03-29 16:05:32.005 454 6208 D AudioPolicyManagerCustom: voice_conc:BLOCK VoIP requst incall mode for inputSource: 1
03-29 16:05:32.005 454 6208 E AudioPolicyIntefaceImpl: inputType = 1 (API_INPUT_MIX_CAPTURE)
03-29 16:05:32.005 469 2397 E AudioRecord: openRecord_l status = -19 (#define ENODEV 19)
03-29 16:05:32.005 469 2397 E AudioRecord: Could not get audio input for session 113, record source 1, sample rate 8000, format 0x1, channel mask 0x10, flags 0

正常通话录音：
04-01 15:03:22.755 438 5896 D audio_hw_primary: enable_audio_route: apply mixer and update path: incall-rec-uplink-and-downlink
04-01 15:03:22.755 438 5896 D audio_route: Apply path: incall-rec-uplink-and-downlink
04-01 15:03:22.761 438 5896 D audio_hw_primary: select_devices: done

04-01 15:03:22.753 438 5896 D voice : voice_get_incall_rec_snd_device: in_snd_device(94: voice-dmic-ef) incall_record_device(67: handset-mic) （有输入有输出）
04-01 15:03:22.755 438 5896 D audio_hw_primary: enable_audio_route: apply mixer and update path: incall-rec-uplink-and-downlink
04-01 15:03:22.755 438 5896 D audio_route: Apply path: incall-rec-uplink-and-downlink

04-01 15:03:21.205 511 5384 E AudioPolicyIntefaceImpl: AudioPolicyService::getInputForAttr - E, attr->source = 4 (AUDIO_SOURCE_VOICE_CALL)
04-01 15:03:21.205 511 5384 D AudioPolicyManagerCustom: voice_conc:Creating input during incall mode for inputSource: 4
04-01 15:03:21.214 511 5384 E AudioPolicyIntefaceImpl: inputType = 3 (API_INPUT_TELEPHONY_RX)
04-01 15:03:21.214 511 5384 E AudioPolicyIntefaceImpl: skip check android.permission.CAPTURE_AUDIO_OUTPUT for CTA request!
04-01 15:03:21.214 523 4058 E AudioRecord: openRecord_l status = 0

我怀疑只要把inputsource从1改为4就好了，不过实验室不肯改。

通过一系列的加log，复测发现一句重要的系统log！！！
skip check android.permission.CAPTURE_AUDIO_OUTPUT for CTA request!

在HAL层发现对应代码：

status_t AudioPolicyManagerCustom::getInputForAttr(const audio_attributes_t *attr,
                                         audio_io_handle_t *input,
                                         audio_session_t session,
                                         uid_t uid,
                                         const audio_config_base_t *config,
                                         audio_input_flags_t flags,
                                         audio_port_handle_t *selectedDeviceId,
                                         input_type_t *inputType,
                                         audio_port_handle_t *portId)
{
    audio_source_t inputSource;
    inputSource = attr->source;
#ifdef VOICE_CONCURRENCY

    char propValue[PROPERTY_VALUE_MAX];
    bool prop_rec_enabled=false, prop_voip_enabled = false;

    if(property_get("vendor.voice.record.conc.disabled", propValue, NULL)) {
        prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
    }

    if(property_get("vendor.voice.voip.conc.disabled", propValue, NULL)) {
        prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
     }

    if (prop_rec_enabled && mvoice_call_state) {
         //check if voice call is active  / running in background
         //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
         //Need to block input request
        if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
           ((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
             (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
        {
            switch(inputSource) {
                case AUDIO_SOURCE_VOICE_UPLINK:
                case AUDIO_SOURCE_VOICE_DOWNLINK:
                case AUDIO_SOURCE_VOICE_CALL:
                    ALOGD("voice_conc:Creating input during incall mode for inputSource: %d",
                        inputSource);
                break;

                case AUDIO_SOURCE_VOICE_COMMUNICATION:
                    if(prop_voip_enabled) {
                       ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
                        inputSource);
                       return NO_INIT;
                    }
                break;
                default:
                    ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
                        inputSource);
                return NO_INIT;
            }
        }
    }//check for VoIP flag
    else if(prop_voip_enabled && mvoice_call_state) {
         //check if voice call is active  / running in background
         //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
         //Need to block input request
        if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
           ((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
             (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
        {
            if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {
                ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
                return NO_INIT;
            }
        }
    }

#endif

    return AudioPolicyManager::getInputForAttr(attr,
                                               input,
                                               session,
                                               uid,
                                               config,
                                               flags,
                                               selectedDeviceId,
                                               inputType,
                                               portId);
}

原来是属性vendor.voice.record.conc.disabled搞的鬼，可见高通代码很严谨，不允许MIC通话录音，应该是安全层面考虑。不过我grep其他平台，该属性竟然都是false，只有8909为true！
/hardware/qcom/audio/configs/msm8909/msm8909.mk

还有通话录音这个功能原生代码是没有的，需要各odm自己添加该功能。
CTA没有用正常的通话录音导致测试失败。
HAL层代码多看看，混口饭吃还能多吃一口，不过真的很费时。