Android N Audio: Audio Track play

对于AT在应用层开始play，到AF打开输出通道的调用流程如下图示，代码分析就不赘述了，

AT:AudioTrack  AF:AudioFlinger AS:AudioSystemAPII:AudioPolicyInterfaceImpl APS:AudioPolicyService APM:AudioPolicyManager

 

在openOutput之后，

 

经过艰难的跋涉，终于又回到了AF，在这里，port的类型有所不同，如下将端口类型分为三种，session、混音和物理设备。

/* Audio port type indicates if it is a session (e.g AudioTrack),  * a mix (e.g PlaybackThread output) or a physical device  * (e.g AUDIO_DEVICE_OUT_SPEAKER) */ typedef enum {     AUDIO_PORT_TYPE_NONE,     AUDIO_PORT_TYPE_DEVICE,     AUDIO_PORT_TYPE_MIX,     AUDIO_PORT_TYPE_SESSION, } audio_port_type_t;

 

 

根据source和port的类型不同，有不同的策略，来打开不同的输出通道，对于source为AUDIO_PORT_TYPE_DEVICE的通道，如果source数目为2，则如下，为输出的mPlaybackThread线程和MIC输入的mRecordThread线程各打开一个通道。

status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *patch,                                    audio_patch_handle_t *handle) { …     switch (patch->sources[0].type) {         case AUDIO_PORT_TYPE_DEVICE: { …             if ((patch->num_sources == 2) ||                 ((patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) &&                  ((patch->sinks[0].ext.device.hw_module != srcModule) ||                   (audioHwDevice->version() < AUDIO_DEVICE_API_VERSION_3_0)))) { …                     newPatch->mPlaybackThread = audioflinger->openOutput_l(                                                              patch->sinks[0].ext.device.hw_module,                                                              &output,                                                              &config,                                                              device,                                                              address,                                                              AUDIO_OUTPUT_FLAG_NONE); …                 audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;                 newPatch->mRecordThread = audioflinger->openInput_l(srcModule,                                                                     &input,                                                                     &config,                                                                     device,                                                                     address,                                                                     AUDIO_SOURCE_MIC,                                                                     AUDIO_INPUT_FLAG_NONE);               } else {                 if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {                     sp<ThreadBase> thread = audioflinger->checkRecordThread_l(                                                               patch->sinks[0].ext.mix.handle);                     if (thread == 0) {                         ALOGW("createAudioPatch() bad capture I/O handle %d",                                                               patch->sinks[0].ext.mix.handle);                         status = BAD_VALUE;                         goto exit;                     }                     status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);                 } else {                     if (audioHwDevice->version() < AUDIO_DEVICE_API_VERSION_3_0) {                         status = INVALID_OPERATION;                         goto exit;                     }                       audio_hw_device_t *hwDevice = audioHwDevice->hwDevice();                     status = hwDevice->create_audio_patch(hwDevice,                                                            patch->num_sources,                                                            patch->sources,                                                            patch->num_sinks,                                                            patch->sinks,                                                            &halHandle);                 }             }

 

create_audio_patch在HAL没有被定义，这个分支应该是走不到的。

 

我们以audioflinger->openOutput_l的分析为主吧。

sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module,                                                             audio_io_handle_t *output,                                                             audio_config_t *config,                                                             audio_devices_t devices,                                                             const String8& address,                                                             audio_output_flags_t flags) { AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); …     AudioStreamOut *outputStream = NULL;     status_t status = outHwDev->openOutputStream(             &outputStream,             *output,             devices,             flags,             config,             address.string());       mHardwareStatus = AUDIO_HW_IDLE;       if (status == NO_ERROR) {           PlaybackThread *thread;         if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {             thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady);             ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread);         } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)                 || !isValidPcmSinkFormat(config->format)                 || !isValidPcmSinkChannelMask(config->channel_mask)) {             thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);             ALOGV("openOutput_l() created direct output: ID %d thread %p ", *output, thread);             //Check if this is DirectPCM, if so             if (flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {                 thread->mIsDirectPcm = true;             }         } else {             thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);             ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);         }         mPlaybackThreads.add(*output, thread);         return thread;     }       return 0; }

 

主要就是找到或加载之前提到的3类输入输出设备，说直白一点，就是加载对应的so库文件，便于后面调用其中的api接口；再接着就是调用设备的openOutputStream，字面意思是打开输出流，这里再说直白一点，就是将so库的api和现在framework的中间层管理起来，形成从上向下的一个调用流程；最后，根据flag类型，来决定使用哪一种线程处理音频数据，这一点在AF的线程分析里面另行分析。

 

 

对于AudioHwDevice的openOutputStream，逐层向下调用，最终会到平台厂商的hal层audio_hw.c，

status_t AudioHwDevice::openOutputStream(         AudioStreamOut **ppStreamOut,         audio_io_handle_t handle,         audio_devices_t devices,         audio_output_flags_t flags,         struct audio_config *config,         const char *address) {       struct audio_config originalConfig = *config; AudioStreamOut *outputStream = new AudioStreamOut(this, flags);       status_t status = outputStream->open(handle, devices, config, address); …     *ppStreamOut = outputStream;     return status; }     status_t AudioStreamOut::open(         audio_io_handle_t handle,         audio_devices_t devices,         struct audio_config *config,         const char *address) {     audio_stream_out_t *outStream;     int status = hwDev()->open_output_stream(             hwDev(),             handle,             devices,             customFlags,             config,             &outStream,             address); …     return status; }

 

 

 

在audio_hw.c的adev_open_output_stream看似很复杂，实际也就是根据flag和device的类型，配置stream_out（定义在audio_hw.h文件里面）的成员变量，并将audio_stream_out类型指针赋值，作为后面操作HAL层的接口。

static int adev_open_output_stream(struct audio_hw_device *dev,                                    audio_io_handle_t handle,                                    audio_devices_t devices,                                    audio_output_flags_t flags,                                    struct audio_config *config,                                    struct audio_stream_out **stream_out,                                    const char *address __unused) {     struct audio_device *adev = (struct audio_device *)dev;     struct stream_out *out;     int ret = 0;     audio_format_t format; *stream_out = NULL; …     out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;     out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;     out->handle = handle;     out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;     out->non_blocking = 0; …     /* Init use case and pcm_config */     if ((out->flags & AUDIO_OUTPUT_FLAG_DIRECT) &&         !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ||         (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) &&         (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ||         out->devices & AUDIO_DEVICE_OUT_PROXY)) { …     out->stream.common.get_sample_rate = out_get_sample_rate;     out->stream.common.set_sample_rate = out_set_sample_rate;     out->stream.common.get_buffer_size = out_get_buffer_size;     out->stream.common.get_channels = out_get_channels;     out->stream.common.get_format = out_get_format;     out->stream.common.set_format = out_set_format; …     *stream_out = &out->stream;    }