From abc1cf31108cbe1a9fab78777c6fcce6bfd8c6f8 Mon Sep 17 00:00:00 2001 From: "DESKTOP-4RNDQIC\\29019" <290198252@qq.com> Date: Mon, 22 Jun 2020 01:13:44 +0800 Subject: [PATCH] no message --- client/qt_gl_/yuvgl/mainwindow.cpp | 2 +- client/qt_gl_/yuvgl/mainwindow.h | 2 +- client/qt_gl_/yuvgl/media/AudioCapture.cpp | 91 +---- client/qt_gl_/yuvgl/media/AudioCapture.h | 41 +- client/qt_gl_/yuvgl/media/audiocaptureff.cpp | 406 +++++++++++++++++++ client/qt_gl_/yuvgl/media/audiocaptureff.h | 85 ++++ client/qt_gl_/yuvgl/yuvgl.pro | 4 +- 7 files changed, 507 insertions(+), 124 deletions(-) create mode 100644 client/qt_gl_/yuvgl/media/audiocaptureff.cpp create mode 100644 client/qt_gl_/yuvgl/media/audiocaptureff.h diff --git a/client/qt_gl_/yuvgl/mainwindow.cpp b/client/qt_gl_/yuvgl/mainwindow.cpp index 87210dd..2556c30 100644 --- a/client/qt_gl_/yuvgl/mainwindow.cpp +++ b/client/qt_gl_/yuvgl/mainwindow.cpp @@ -110,5 +110,5 @@ void MainWindow::on_pushButton_2_clicked() void MainWindow::on_pushButton_3_clicked() { - + qDebug()<comboBox_2->currentText(); } diff --git a/client/qt_gl_/yuvgl/mainwindow.h b/client/qt_gl_/yuvgl/mainwindow.h index 170db42..33e074f 100644 --- a/client/qt_gl_/yuvgl/mainwindow.h +++ b/client/qt_gl_/yuvgl/mainwindow.h @@ -10,7 +10,7 @@ #include "utils.h" #include "media/RtmpPusher.h" #include "Qss/Qss.h" -#include "media/AudioCapture.h" +#include "media/audiocaptureff.h" #include using namespace std; diff --git a/client/qt_gl_/yuvgl/media/AudioCapture.cpp b/client/qt_gl_/yuvgl/media/AudioCapture.cpp index 86f7189..9abf18c 100644 --- a/client/qt_gl_/yuvgl/media/AudioCapture.cpp +++ b/client/qt_gl_/yuvgl/media/AudioCapture.cpp @@ -10,7 +10,14 @@ CaptureAudio::CaptureAudio(uint16_t rate, uint8_t channel) { this->observer = nullptr; } - +/** + * @brief 设置捕获到的回调通知类 + * + * @param CaptureAudioObserver 参数说明 + * @return 返回说明 + * @retval -1 输入为空指针 + * @retval 0成功 + */ int CaptureAudio::SetObserver(CaptureAudioObserver* ob) { if (nullptr == ob) return -1; this->observer = ob; @@ -124,85 +131,3 @@ void CaptureAudio::StopCapture() this->mStatus = STOP; } } -std::string WString2String(const std::wstring& ws) -{ - std::string strLocale = setlocale(LC_ALL, ""); - const wchar_t* wchSrc = ws.c_str(); - size_t nDestSize = wcstombs(NULL, wchSrc, 0) + 1; - char *chDest = new char[nDestSize]; - memset(chDest, 0, nDestSize); - wcstombs(chDest, wchSrc, nDestSize); - std::string strResult = chDest; - delete[]chDest; - setlocale(LC_ALL, strLocale.c_str()); - return strResult; -} -vector CaptureAudioFfmpeg::EnumSpeakers() -{ - vector ret; - std::vector names; - IEnumMoniker *pEnum = nullptr; - // Create the System Device Enumerator. - ICreateDevEnum *pDevEnum; - HRESULT hr = CoCreateInstance(CLSID_SystemDeviceEnum, nullptr, - CLSCTX_INPROC_SERVER, IID_PPV_ARGS(&pDevEnum)); - - if (SUCCEEDED(hr)) - { - // Create an enumerator for the category. - hr = pDevEnum->CreateClassEnumerator(CLSID_AudioInputDeviceCategory, &pEnum, 0); - if (hr == S_FALSE) - { - hr = VFW_E_NOT_FOUND; // The category is empty. Treat as an error. - } - pDevEnum->Release(); - } - - if (!SUCCEEDED(hr)) - return ret; - - IMoniker *pMoniker = nullptr; - while (pEnum->Next(1, &pMoniker, nullptr) == S_OK) - { - IPropertyBag *pPropBag; - IBindCtx* bindCtx = nullptr; - LPOLESTR str = nullptr; - VARIANT var; - VariantInit(&var); - - HRESULT hr = pMoniker->BindToStorage(0, 0, IID_PPV_ARGS(&pPropBag)); - if (FAILED(hr)) - { - pMoniker->Release(); - continue; - } - - // Get description or friendly name. - hr = pPropBag->Read(L"Description", &var, 0); - if (FAILED(hr)) - { - hr = pPropBag->Read(L"FriendlyName", &var, 0); - } - if (SUCCEEDED(hr)) - { - names.push_back(var.bstrVal); - std::string x = WString2String(var.bstrVal); - CaptureAudioFfmpeg::MICInfo ele; - ele.name = x; - ret.push_back(ele); - VariantClear(&var); - } - - pPropBag->Release(); - pMoniker->Release(); - } - - pEnum->Release(); - - return ret; -} - -CaptureAudioFfmpeg::CaptureAudioFfmpeg(uint16_t rate, uint8_t channel) -{ - -} diff --git a/client/qt_gl_/yuvgl/media/AudioCapture.h b/client/qt_gl_/yuvgl/media/AudioCapture.h index c025208..61433de 100644 --- a/client/qt_gl_/yuvgl/media/AudioCapture.h +++ b/client/qt_gl_/yuvgl/media/AudioCapture.h @@ -14,6 +14,8 @@ extern "C" #include "libavutil/opt.h" #include "libavutil/imgutils.h" #include "libavdevice/avdevice.h" +#include "libavfilter/avfilter.h" + }; #include @@ -22,6 +24,7 @@ extern "C" #include "qedit.h" #include #include +#include #include "guiddef.h" using namespace std; @@ -66,43 +69,5 @@ private: }; -class CaptureAudioFfmpeg { -public: - class CaptureAudioObserver { - public: - virtual void OnAudioData(const void *frameaddress, uint32_t framelen) {}; - }; - typedef struct _T_MicInfo - { - string name; - int index; - }MICInfo; - enum CAP_STATUS { - RUNNING = 1, - STOP = 2, - PAUSE = 3, - FAIL = 4, - }; - - vector EnumSpeakers(); - CaptureAudioFfmpeg(uint16_t rate, uint8_t channel); - - /* - ~CaptureAudio(); - int StartCapture(); - int InitCapture(int index,uint16_t rate,uint8_t channel); - void StopCapture(); - int SetObserver(CaptureAudioObserver*); - int OnCallBack(const void* input, void* output, unsigned long frameCount); - void AddCnt(unsigned int x) {this->mSize += x;}; - */ -private: - uint16_t mSampleRate; //采样率 - uint16_t mChanel; //通道号 - - unsigned long mSize; - CAP_STATUS mStatus; - CaptureAudioObserver *observer; -}; #endif //__CAPTUREAUDIO_H__ diff --git a/client/qt_gl_/yuvgl/media/audiocaptureff.cpp b/client/qt_gl_/yuvgl/media/audiocaptureff.cpp new file mode 100644 index 0000000..f760762 --- /dev/null +++ b/client/qt_gl_/yuvgl/media/audiocaptureff.cpp @@ -0,0 +1,406 @@ +#include "audiocaptureff.h" + + +std::string WString2String(const std::wstring& ws) +{ + std::string strLocale = setlocale(LC_ALL, ""); + const wchar_t* wchSrc = ws.c_str(); + size_t nDestSize = wcstombs(NULL, wchSrc, 0) + 1; + char *chDest = new char[nDestSize]; + memset(chDest, 0, nDestSize); + wcstombs(chDest, wchSrc, nDestSize); + std::string strResult = chDest; + delete[]chDest; + setlocale(LC_ALL, strLocale.c_str()); + return strResult; +} +vector CaptureAudioFfmpeg::EnumSpeakers() +{ + vector ret; + std::vector names; + IEnumMoniker *pEnum = nullptr; + // Create the System Device Enumerator. + ICreateDevEnum *pDevEnum; + HRESULT hr = CoCreateInstance(CLSID_SystemDeviceEnum, nullptr, + CLSCTX_INPROC_SERVER, IID_PPV_ARGS(&pDevEnum)); + + if (SUCCEEDED(hr)) + { + // Create an enumerator for the category. + hr = pDevEnum->CreateClassEnumerator(CLSID_AudioInputDeviceCategory, &pEnum, 0); + if (hr == S_FALSE) + { + hr = VFW_E_NOT_FOUND; // The category is empty. Treat as an error. + } + pDevEnum->Release(); + } + + if (!SUCCEEDED(hr)) + return ret; + + IMoniker *pMoniker = nullptr; + while (pEnum->Next(1, &pMoniker, nullptr) == S_OK) + { + IPropertyBag *pPropBag; + IBindCtx* bindCtx = nullptr; + LPOLESTR str = nullptr; + VARIANT var; + VariantInit(&var); + + HRESULT hr = pMoniker->BindToStorage(0, 0, IID_PPV_ARGS(&pPropBag)); + if (FAILED(hr)) + { + pMoniker->Release(); + continue; + } + + // Get description or friendly name. + hr = pPropBag->Read(L"Description", &var, 0); + if (FAILED(hr)) + { + hr = pPropBag->Read(L"FriendlyName", &var, 0); + } + if (SUCCEEDED(hr)) + { + names.push_back(var.bstrVal); + CaptureAudioFfmpeg::MICInfo ele; + ele.name = var.bstrVal; + ret.push_back(ele); + VariantClear(&var); + } + + pPropBag->Release(); + pMoniker->Release(); + } + + pEnum->Release(); + + return ret; +} + + +CaptureAudioFfmpeg::CaptureAudioFfmpeg(uint16_t rate, uint8_t channel) +{ + mSampleRate = rate; + mChanel = channel; + +} + + +static char *dup_wchar_to_utf8(wchar_t *w) +{ + char *s = NULL; + int l = WideCharToMultiByte(CP_UTF8, 0, w, -1, 0, 0, 0, 0); + s = (char *)av_malloc(l); + if (s) + WideCharToMultiByte(CP_UTF8, 0, w, -1, s, l, 0, 0); + return s; +} + +int CaptureAudioFfmpeg::InitCapture(wstring url, uint16_t rate, uint8_t channel) +{ + string fileAudioInput = dup_wchar_to_utf8((wchar_t *)url.c_str()); + AVInputFormat* imft = av_find_input_format("dshow"); + AVDictionary *format_opts = nullptr; + av_dict_set_int(&format_opts, "audio_buffer_size", 20, 0); + if (0 > avformat_open_input(&mInfmt_ctx, fileAudioInput.c_str(), imft, &format_opts)) { + printf("failed input file\n"); + return -1; + } + if (0 > avformat_find_stream_info(mInfmt_ctx, NULL)) { + printf("failed find stream info\n"); + avformat_close_input(&mInfmt_ctx); + return -1; + } + int audio_index = -1; + audio_index = av_find_best_stream(mInfmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0); + if (-1 == audio_index) { + printf("failed find best stream\n"); + avformat_close_input(&mInfmt_ctx); + return -1; + } + //av_dump_format(infmt_ctx, 0, fileAudioInput.c_str(), 1); + //END杈撳叆鏂囦欢 + + //鎵撳紑瑙ｇ爜鍣 + static AVCodec* decodec = avcodec_find_decoder(mInfmt_ctx->streams[0]->codec->codec_id); + if (!decodec) { + printf("failed find decoder\n"); + return -1; + } + if (0 > avcodec_open2(mInfmt_ctx->streams[0]->codec, decodec, NULL)) { + printf("failed open decoder\n"); + return -1; + } + //END瑙ｇ爜鍣 + //閲嶉噰鏍峰垵濮嬪寲 + initAudioFilters(); + //END閲嶉噰鏍峰垵濮嬪寲 + //缂栫爜鍣 + static AVCodec* codec = NULL; + //codec = avcodec_find_encoder_by_name("libmp3lame"); + codec = avcodec_find_encoder(AV_CODEC_ID_AAC); + static AVCodecContext* codec_ctx = NULL; + codec_ctx = avcodec_alloc_context3(codec); + // codec_ctx->bit_rate = 64000; + // inputContext->streams[0]->codec + codec_ctx->codec = codec; + codec_ctx->sample_rate = 48000; + codec_ctx->channel_layout = 3; + codec_ctx->channels = 2; + //codec_ctx->frame_size = 1024; + codec_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP; + codec_ctx->codec_tag = 0; + codec_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; + + if (0 > avcodec_open2(codec_ctx, codec, NULL)) { + printf("failed open coder\n"); + avformat_close_input(&mInfmt_ctx); + avcodec_free_context(&codec_ctx); + return -1; + } + //END缂栫爜鍣 + //杈撳嚭鏂囦欢 + AVFormatContext* outfmt_ctx = NULL; + if (0 > avformat_alloc_output_context2(&outfmt_ctx, NULL, NULL, "aac.aac")) { + printf("failed alloc outputcontext\n"); + avformat_close_input(&mInfmt_ctx); + avcodec_free_context(&codec_ctx); + return -1; + } + AVStream* out_stream = avformat_new_stream(outfmt_ctx, codec_ctx->codec); + if (!out_stream) { + printf("failed new stream\n"); + avformat_close_input(&mInfmt_ctx); + avcodec_free_context(&codec_ctx); + avformat_close_input(&outfmt_ctx); + return -1; + } + avcodec_copy_context(out_stream->codec, codec_ctx); + // if (0 > avio_open(&outfmt_ctx->pb, "rtmp://localhost/testlive", AVIO_FLAG_WRITE)) { + if (0 > avio_open(&outfmt_ctx->pb, "aac.aac", AVIO_FLAG_WRITE)) { + printf("failed to open outfile\n"); + avformat_close_input(&mInfmt_ctx); + avcodec_free_context(&codec_ctx); + avformat_close_input(&outfmt_ctx); + return -1; + } + avformat_write_header(outfmt_ctx, NULL); + //END杈撳嚭鏂囦欢 + #if 0 + AVFrame* Frame = av_frame_alloc(); + Frame->nb_samples = codec_ctx->frame_size; + Frame->format = codec_ctx->sample_fmt; + Frame->channel_layout = codec_ctx->channel_layout; + int size = av_samples_get_buffer_size(NULL, codec_ctx->channels, codec_ctx->frame_size, + codec_ctx->sample_fmt, 1); + uint8_t* frame_buf = (uint8_t *)av_malloc(size); + avcodec_fill_audio_frame(Frame, codec_ctx->channels, codec_ctx->sample_fmt, (const uint8_t*)frame_buf, size, 1); + int64_t in_channel_layout = av_get_default_channel_layout(codec_ctx->channels); + AVPacket pkt; + av_new_packet(&pkt, size); + pkt.data = NULL; + int got_frame = -1; + int delayedFrame = 0; + static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2]; + int audioCount = 0; + const uint8_t *indata[AV_NUM_DATA_POINTERS] = { 0 }; + AVFrame* Frame1 = av_frame_alloc(); + #endif + int loop = 1; + int delayedFrame = 0; + AVPacket packet; + av_init_packet(&packet); + packet.data = NULL; + packet.size = 0; + AVPacket pkt; + av_init_packet(&pkt); + pkt.data = NULL; + pkt.size = 0; + + AVFrame* pSrcAudioFrame = av_frame_alloc(); + int got_frame = 0; + + while (1) { + av_read_frame(mInfmt_ctx, &packet); + loop++; + if (packet.stream_index == audio_index) { + auto filterFrame = DecodeAudio(&packet, pSrcAudioFrame); + if (filterFrame) { + avcodec_encode_audio2(codec_ctx, &pkt, filterFrame, &got_frame); + if (got_frame) { + #if 1 + auto streamTimeBase = outfmt_ctx->streams[pkt.stream_index]->time_base.den; + auto codecTimeBase = outfmt_ctx->streams[pkt.stream_index]->codec->time_base.den; + pkt.pts = pkt.dts = (1024 * streamTimeBase * mAudioCount) / codecTimeBase; + mAudioCount++; + auto inputStream = mInfmt_ctx->streams[pkt.stream_index]; + auto outputStream = outfmt_ctx->streams[pkt.stream_index]; + av_packet_rescale_ts(&pkt, inputStream->time_base, outputStream->time_base); + #endif + // pkt.stream_index = out_stream->index; + av_interleaved_write_frame(outfmt_ctx, &pkt); + av_packet_unref(&pkt); + printf("output frame %3d\n", loop - delayedFrame); + } + else { + delayedFrame++; + av_packet_unref(&pkt); + printf("no output frame\n"); + } + } + } + av_packet_unref(&packet); + } + flush_encoder(outfmt_ctx, 0); + av_write_trailer(outfmt_ctx); + //av_free(Frame); + av_free(pSrcAudioFrame); + avio_close(outfmt_ctx->pb); + avformat_close_input(&mInfmt_ctx); + //avformat_close_input(&outfmt_ctx); + return 0; +} + +int CaptureAudioFfmpeg::initAudioFilters() +{ + char args[512]; + int ret; + AVFilter *abuffersrc = (AVFilter *)avfilter_get_by_name("abuffer"); + AVFilter *abuffersink = (AVFilter *)avfilter_get_by_name("abuffersink"); + AVFilterInOut *outputs = avfilter_inout_alloc(); + AVFilterInOut *inputs = avfilter_inout_alloc(); + + auto audioDecoderContext = mInfmt_ctx->streams[0]->codec; + if (!audioDecoderContext->channel_layout) + audioDecoderContext->channel_layout = av_get_default_channel_layout(audioDecoderContext->channels); + + static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }; + static const uint64_t out_channel_layouts[] = { audioDecoderContext->channel_layout}; + static const int out_sample_rates[] = { audioDecoderContext->sample_rate , -1 }; + + AVRational time_base = mInfmt_ctx->streams[0]->time_base; + mFilterGraph = avfilter_graph_alloc(); + mFilterGraph->nb_threads = 1; + + sprintf_s(args, sizeof(args), + "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%I64x", + time_base.num, time_base.den, audioDecoderContext->sample_rate, + av_get_sample_fmt_name(audioDecoderContext->sample_fmt), audioDecoderContext->channel_layout); + + ret = avfilter_graph_create_filter(&mBuffersrcCtx, abuffersrc, "in", + args, NULL, mFilterGraph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n"); + return ret; + } + + /* buffer audio sink: to terminate the filter chain. */ + ret = avfilter_graph_create_filter(&mBuffersinkCtx, abuffersink, "out", + NULL, NULL, mFilterGraph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n"); + return ret; + } + + ret = av_opt_set_int_list(mBuffersinkCtx, "sample_fmts", out_sample_fmts, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n"); + return ret; + } + + ret = av_opt_set_int_list(mBuffersinkCtx, "channel_layouts", out_channel_layouts, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n"); + return ret; + } + + ret = av_opt_set_int_list(mBuffersinkCtx, "sample_rates", out_sample_rates, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n"); + return ret; + } + + /* Endpoints for the filter graph. */ + outputs->name = av_strdup("in"); + outputs->filter_ctx = mBuffersrcCtx;; + outputs->pad_idx = 0; + outputs->next = NULL; + + inputs->name = av_strdup("out"); + inputs->filter_ctx = mBuffersinkCtx; + inputs->pad_idx = 0; + inputs->next = NULL; + + if ((ret = avfilter_graph_parse_ptr(mFilterGraph, "anull", + &inputs, &outputs, nullptr)) < 0) + return ret; + + if ((ret = avfilter_graph_config(mFilterGraph, NULL)) < 0) + return ret; + + av_buffersink_set_frame_size(mBuffersinkCtx, 1024); + return 0; +} + +int CaptureAudioFfmpeg::flush_encoder(AVFormatContext *fmt_ctx, unsigned int stream_index) +{ + int ret; + int got_frame; + AVPacket enc_pkt; + if (!(fmt_ctx->streams[stream_index]->codec->codec->capabilities & + 0x0020)) + return 0; + while (1) { + enc_pkt.data = NULL; + enc_pkt.size = 0; + av_init_packet(&enc_pkt); + ret = avcodec_encode_audio2(fmt_ctx->streams[stream_index]->codec, &enc_pkt, + NULL, &got_frame); + av_frame_free(NULL); + if (ret < 0) + break; + if (!got_frame) { + ret = 0; + break; + } + printf("Flush Encoder: Succeed to encode 1 frame!\tsize:%5d\n", enc_pkt.size); + /* mux encoded frame */ + ret = av_write_frame(fmt_ctx, &enc_pkt); + if (ret < 0) + break; + } + return ret; + + +} + +AVFrame *CaptureAudioFfmpeg::DecodeAudio(AVPacket *packet, AVFrame *pSrcAudioFrame) +{ + AVStream * stream = mInfmt_ctx->streams[0]; + AVCodecContext* codecContext = stream->codec; + int gotFrame; + AVFrame *filtFrame = nullptr; + auto length = avcodec_decode_audio4(codecContext, pSrcAudioFrame, &gotFrame, packet); + if (length >= 0 && gotFrame != 0) + { + if (av_buffersrc_add_frame_flags(mBuffersrcCtx, pSrcAudioFrame, AV_BUFFERSRC_FLAG_PUSH) < 0) { + av_log(NULL, AV_LOG_ERROR, "buffe src add frame error!\n"); + return nullptr; + } + + filtFrame = av_frame_alloc(); + int ret = av_buffersink_get_frame_flags(mBuffersinkCtx, filtFrame, AV_BUFFERSINK_FLAG_NO_REQUEST); + if (ret < 0) + { + av_frame_free(&filtFrame); + goto error; + } + return filtFrame; + } + error: + return nullptr; +} diff --git a/client/qt_gl_/yuvgl/media/audiocaptureff.h b/client/qt_gl_/yuvgl/media/audiocaptureff.h new file mode 100644 index 0000000..7dae1c8 --- /dev/null +++ b/client/qt_gl_/yuvgl/media/audiocaptureff.h @@ -0,0 +1,85 @@ +#ifndef AUDIOCAPTUREFF_H +#define AUDIOCAPTUREFF_H + +#include "stdint.h" +#include "../third/portaudio/portaudio.h" +#include +#include +//Windows +extern "C" +{ +#include "libavcodec/avcodec.h" +#include "libavformat/avformat.h" +#include "libavutil/avutil.h" +#include "libswscale/swscale.h" +#include "libavutil/opt.h" +#include "libavutil/imgutils.h" +#include "libavdevice/avdevice.h" +#include "libavfilter/avfilter.h" +#include "libavfilter/buffersrc.h" +#include "libavfilter/buffersink.h" +}; +#include +#include +#include +#include "qedit.h" +#include +#include +#include +#include "guiddef.h" +using namespace std; + + +class CaptureAudioFfmpeg { +public: + class CaptureAudioObserver { + public: + virtual void OnAudioData(const void *frameaddress, uint32_t framelen) {}; + }; + typedef struct _T_MicInfo + { + wstring name; + int index; + }MICInfo; + enum CAP_STATUS { + RUNNING = 1, + STOP = 2, + PAUSE = 3, + FAIL = 4, + }; + + vector EnumSpeakers(); + CaptureAudioFfmpeg(uint16_t rate, uint8_t channel); + int InitCapture(wstring url,uint16_t rate,uint8_t channel); + + /* + ~CaptureAudio(); + int StartCapture(); + void StopCapture(); + int SetObserver(CaptureAudioObserver*); + int OnCallBack(const void* input, void* output, unsigned long frameCount); + void AddCnt(unsigned int x) {this->mSize += x;}; + */ +private: + std::thread mThread; + uint16_t mSampleRate; //閲囨牱鐜 + uint16_t mChanel; //閫氶亾鍙 + uint16_t mSamplefmt; + unsigned long mSize; + CAP_STATUS mStatus; + CaptureAudioObserver *observer; + int initAudioFilters(); + AVFormatContext *mInfmt_ctx = nullptr; + AVFormatContext * mOutfmt_ctx = NULL; + int64_t mLastReadPacktTime; + AVFilterContext *mBuffersinkCtx = NULL; + AVFilterContext *mBuffersrcCtx = NULL; + AVFilterGraph *mFilterGraph = NULL; + AVCodecContext* mOutPutAudioEncContext = NULL; + int64_t mAudioCount = 0; + + int flush_encoder(AVFormatContext *fmt_ctx, unsigned int stream_index); + AVFrame *DecodeAudio(AVPacket* packet, AVFrame*pSrcAudioFrame); +}; + +#endif // AUDIOCAPTUREFF_H diff --git a/client/qt_gl_/yuvgl/yuvgl.pro b/client/qt_gl_/yuvgl/yuvgl.pro index 8604f34..e919a68 100644 --- a/client/qt_gl_/yuvgl/yuvgl.pro +++ b/client/qt_gl_/yuvgl/yuvgl.pro @@ -38,6 +38,7 @@ SOURCES += \ media/CameraCapture.cpp \ media/RtmpPusher.cpp \ media/VideoCoder.cpp \ + media/audiocaptureff.cpp \ media/sps_decode.cpp \ utils/Base64.cpp \ utils/Debuger.cpp \ @@ -47,7 +48,8 @@ HEADERS += \ mainwindow.h \ cplaywidget.h \ cplaywidget.h \ - Qss/Qss.h + Qss/Qss.h \ + media/audiocaptureff.h FORMS += \ components/toast.ui \ mainwindow.ui