聊天平臺原始碼,聊天平臺如何獲取到音訊流
聊天平臺原始碼,聊天平臺如何獲取到音訊流的相關程式碼
我們重取樣的引數要和SDL的引數是一致的,還要考慮到雖然沒有AVPacket了,但是解碼器中還有未解碼的資料,我們要用avcodec_send_packet(aCodecCtx, NULL)告訴解碼器沒有資料了,重新整理解碼器。
下面這段程式碼的邏輯是如果沒有資料要解碼了,就跳轉到獲取解碼資料的邏輯。
int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size) { static AVPacket pkt; static uint8_t *audio_pkt_data = NULL; static int audio_pkt_size = 0; static AVFrame frame; av_frame_get_buffer(&frame, 0); int len1, data_size = 0; int ret = 0; int is_no_pkt = 0; for (;;) { if (pkt.data) av_packet_unref(&pkt); if (quit) { return -1; } if (packet_queue_get(&audioq, &pkt, 1) < 0) { ret = avcodec_send_packet(aCodecCtx, NULL); if (ret < 0) { return -1; } is_no_pkt = 1; goto __RECEIVE; } ret = avcodec_send_packet(aCodecCtx, &pkt); if (ret < 0) { ret = -1; printf("decode error"); av_packet_unref(&pkt); return -1; } __RECEIVE: if (is_no_pkt) { return -1; } else if (pkt.data) { av_packet_unref(&pkt); } ret = avcodec_receive_frame(aCodecCtx, &frame); if (ret < 0) { continue; } data_size = av_get_bytes_per_sample(out_format) * out_channel * SDL_AUDIO_BUFFER_SIZE; swr_convert(audio_convert_ctx, &audio_buf, SDL_AUDIO_BUFFER_SIZE, (const uint8_t **)frame.data, frame.nb_samples); return data_size; } }
轉碼後的資料大小=取樣大小 * 單通道取樣個數 * 通道數
data_size = av_get_bytes_per_sample(out_format) * out_channel * SDL_AUDIO_BUFFER_SIZE;
下面是完整的程式碼
#include <stdio.h> #include <assert.h> #include <SDL2/SDL.h> #include <libavcodec/avcodec.h> #include <libavformat/avformat.h> #include <libswscale/swscale.h> #include <libswresample/swresample.h> #include <libavutil/samplefmt.h> #define SDL_AUDIO_BUFFER_SIZE 1024 #define MAX_AUDIO_FRAME_SIZE 192000 typedef struct PacketQueue { AVPacketList *first_pkt, *last_pkt; int nb_packets; int size; SDL_mutex *mutex; SDL_cond *cond; } PacketQueue; PacketQueue audioq; int quit = 0; struct SwrContext *audio_convert_ctx = NULL; static Uint8 out_channel = 2; static SDL_AudioFormat out_format = AV_SAMPLE_FMT_S16; static int out_nb_samples = 0; static int sample_rate = 48000; void audio_callback(void *userdata, Uint8 *stream, int len); void packet_queue_init(PacketQueue *q); int packet_queue_put(PacketQueue *q, AVPacket *pkt); int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block); int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size); int main(int argc, char *argv[]) { int ret = 0; AVFormatContext *fmt_ctx = NULL; char *in_file; int video_stream_index = -1; int audio_stream_index = -1; AVCodec *video_codec = NULL, *audio_codec = NULL; AVCodecParameters *audio_codecpar = NULL, *video_codecpar = NULL; AVCodecContext *audio_codec_ctx = NULL, *video_codec_ctx = NULL; SDL_AudioSpec wanted_spec, src_spec; int64_t in_channel_layout; int64_t out_channel_layout; AVPacket packet; SDL_Event event; av_log_set_level(AV_LOG_INFO); if (argc < 2) { av_log(NULL, AV_LOG_ERROR, "need infile\n"); return -1; } in_file = argv[1]; //初始化SDL ret = SDL_Init(SDL_INIT_AUDIO | SDL_INIT_VIDEO | SDL_INIT_TIMER); if (ret != 0) { av_log(NULL, AV_LOG_ERROR, "sdl init error\n"); return ret; } //獲取輸入檔案上下文 ret = avformat_open_input(&fmt_ctx, in_file, NULL, NULL); if (!fmt_ctx) { av_log(NULL, AV_LOG_ERROR, "open input file fail\n"); goto __FAIL; } //獲取視訊流索引和音訊流索引 ret = avformat_find_stream_info(fmt_ctx, NULL); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "find stream fail\n"); goto __FAIL; } av_dump_format(fmt_ctx, 0, in_file, 0); video_stream_index = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &video_codec, -1); audio_stream_index = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &audio_codec, -1); if (video_stream_index < 0 || audio_stream_index < 0) { av_log(NULL, AV_LOG_ERROR, "find audio or video fail\n"); goto __FAIL; } if (!audio_codec || !video_codec) { ret = -1; av_log(NULL, AV_LOG_ERROR, "find audio codec or video codec fail"); goto __FAIL; } //獲取解碼器上下文並開啟解碼器 audio_codecpar = fmt_ctx->streams[audio_stream_index]->codecpar; out_nb_samples = audio_codecpar->frame_size; video_codecpar = fmt_ctx->streams[video_stream_index]->codecpar; audio_codec_ctx = avcodec_alloc_context3(audio_codec); video_codec_ctx = avcodec_alloc_context3(video_codec); avcodec_parameters_to_context(audio_codec_ctx, audio_codecpar); avcodec_parameters_to_context(video_codec_ctx, video_codecpar); ret = avcodec_open2(audio_codec_ctx, audio_codec, NULL); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "open audio codec fail\n"); goto __FAIL; } ret = avcodec_open2(video_codec_ctx, video_codec, NULL); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "open video codec fail\n"); goto __FAIL; } //設定SDL_音訊引數 wanted_spec.freq = sample_rate; //取樣率 wanted_spec.format = AUDIO_S16LSB; //取樣大小 wanted_spec.channels = out_channel; //聲道數 wanted_spec.silence = 0; wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE; //取樣個數 wanted_spec.callback = audio_callback; wanted_spec.userdata = audio_codec_ctx; ret = SDL_OpenAudio(&wanted_spec, &src_spec); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "open audio device fail\n"); goto __FAIL; } //設定重取樣 in_channel_layout = audio_codecpar->channel_layout; out_channel_layout = av_get_default_channel_layout(out_channel); audio_convert_ctx = swr_alloc(); if (!audio_convert_ctx) { ret = -1; av_log(NULL, AV_LOG_ERROR, "alloc swr fail\n"); goto __FAIL; } swr_alloc_set_opts(audio_convert_ctx, out_channel_layout, out_format, sample_rate, in_channel_layout, audio_codecpar->format, audio_codecpar->sample_rate, 0, NULL); swr_init(audio_convert_ctx); SDL_PauseAudio(0); packet_queue_init(&audioq); while (av_read_frame(fmt_ctx, &packet) >= 0) { if (packet.stream_index == audio_stream_index) { packet_queue_put(&audioq, &packet); } else { av_packet_unref(&packet); } SDL_PollEvent(&event); switch (event.type) { case SDL_QUIT: quit = 1; goto __FAIL; break; default: break; } } while (1) { SDL_WaitEvent(&event); switch (event.type) { case SDL_QUIT: quit = 1; goto __FAIL; break; default: break; } } __FAIL: SDL_CloseAudio(); SDL_Quit(); if (audio_codecpar) { avcodec_parameters_free(&audio_codecpar); } if (video_codecpar) { avcodec_parameters_free(&video_codecpar); } if (video_codec_ctx) { avcodec_close(video_codec_ctx); avcodec_free_context(&video_codec_ctx); } if (audio_codec_ctx) { avcodec_close(audio_codec_ctx); avcodec_free_context(&audio_codec_ctx); } if (fmt_ctx) { avformat_close_input(&fmt_ctx); avformat_free_context(fmt_ctx); } return ret; } void audio_callback(void *userdata, Uint8 *stream, int len) { AVCodecContext *aCodecCtx = (AVCodecContext *)userdata; int len1, audio_size; static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2]; static unsigned int audio_buf_size = 0; static unsigned int audio_buf_index = 0; while (len > 0) { if (audio_buf_index >= audio_buf_size) { /* We have already sent all our data; get more */ audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf)); if (audio_size < 0) { /* If error, output silence */ audio_buf_size = 1024; // arbitrary? memset(audio_buf, 0, audio_buf_size); } else { audio_buf_size = audio_size; } audio_buf_index = 0; } len1 = audio_buf_size - audio_buf_index; if (len1 > len) len1 = len; fprintf(stderr, "index=%d, len1=%d, len=%d\n", audio_buf_index, len, len1); memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1); len -= len1; stream += len1; audio_buf_index += len1; } } void packet_queue_init(PacketQueue *q) { memset(q, 0, sizeof(PacketQueue)); q->mutex = SDL_CreateMutex(); q->cond = SDL_CreateCond(); } int packet_queue_put(PacketQueue *q, AVPacket *pkt) { AVPacketList *pkt1; AVPacket dst_pkt; av_init_packet(&dst_pkt); if (av_packet_ref(&dst_pkt, pkt) < 0) { return -1; } pkt1 = av_malloc(sizeof(AVPacketList)); if (!pkt1) return -1; pkt1->pkt = dst_pkt; pkt1->next = NULL; SDL_LockMutex(q->mutex); if (!q->last_pkt) { q->first_pkt = pkt1; } else { q->last_pkt->next = pkt1; } q->last_pkt = pkt1; q->nb_packets++; q->size += pkt1->pkt.size; SDL_CondSignal(q->cond); SDL_UnlockMutex(q->mutex); return 0; } int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block) { AVPacketList *pkt1; int ret; SDL_LockMutex(q->mutex); for (;;) { if (quit) { ret = -1; break; } pkt1 = q->first_pkt; if (pkt1) { q->first_pkt = pkt1->next; if (!q->first_pkt) q->last_pkt = NULL; q->nb_packets--; q->size -= pkt1->pkt.size; *pkt = pkt1->pkt; av_free(pkt1); ret = 1; break; } else if (!block) { ret = 0; break; } else { SDL_CondWait(q->cond, q->mutex); } } SDL_UnlockMutex(q->mutex); return ret; } int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size) { static AVPacket pkt; static uint8_t *audio_pkt_data = NULL; static int audio_pkt_size = 0; static AVFrame frame; av_frame_get_buffer(&frame, 0); int len1, data_size = 0; int ret = 0; int is_no_pkt = 0; for (;;) { if (pkt.data) av_packet_unref(&pkt); if (quit) { return -1; } if (packet_queue_get(&audioq, &pkt, 1) < 0) { ret = avcodec_send_packet(aCodecCtx, NULL); if (ret < 0) { return -1; } is_no_pkt = 1; goto __RECEIVE; } ret = avcodec_send_packet(aCodecCtx, &pkt); if (ret < 0) { ret = -1; printf("decode error"); av_packet_unref(&pkt); return -1; } __RECEIVE: if (is_no_pkt) { return -1; } else if (pkt.data) { av_packet_unref(&pkt); } ret = avcodec_receive_frame(aCodecCtx, &frame); if (ret < 0) { continue; } data_size = av_get_bytes_per_sample(out_format) * out_channel * SDL_AUDIO_BUFFER_SIZE; swr_convert(audio_convert_ctx, &audio_buf, SDL_AUDIO_BUFFER_SIZE, (const uint8_t **)frame.data, frame.nb_samples); return data_size; } }
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