=====================================================
=====================================================
简介：
ffplay的音频输出通过SDL实现，主要流程分为如下几步：
- 打开音频设备，设置参数
- 启动SDL音频设备播放
- SDL音频回调函数读取数据
代码如下：
case AVMEDIA_TYPE_AUDIO: //从avctx(即AVCodecContext)中获取音频格式参数 sample_rate = avctx->sample_rate; nb_channels = avctx->channels; channel_layout = avctx->channel_layout; /* prepare audio output 准备音频输出*/ //调用audio_open打开sdl音频输出，实际打开的设备参数保存在audio_tgt，返回值表示输出设备的缓冲区大小 if ((ret = audio_open(is, channel_layout, nb_channels, sample_rate, &is->audio_tgt)) < 0) goto fail; is->audio_hw_buf_size = ret; is->audio_src = is->audio_tgt; //暂且将数据源参数等同于目标输出参数 //初始化audio_buf相关参数 is->audio_buf_size = 0; is->audio_buf_index = 0; /* init averaging filter 初始化averaging滤镜, 非audio master时使用 */ is->audio_diff_avg_coef = exp(log(0.01) / AUDIO_DIFF_AVG_NB); is->audio_diff_avg_count = 0; /* 由于我们没有精确的音频数据填充FIFO,故只有在大于该阈值时才进行校正音频同步*/ is->audio_diff_threshold = (double)(is->audio_hw_buf_size) / is->audio_tgt.bytes_per_sec; is->audio_stream = stream_index; // 获取audio的stream索引 is->audio_st = ic->streams[stream_index]; // 获取audio的stream指针 // 初始化ffplay封装的音频解码器 decoder_init(&is->auddec, avctx, &is->audioq, is->continue_read_thread); if ((is->ic->iformat->flags & (AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH | AVFMT_NO_BYTE_SEEK)) && !is->ic->iformat->read_seek) { is->auddec.start_pts = is->audio_st->start_time; is->auddec.start_pts_tb = is->audio_st->time_base; } // 启动音频解码线程 if ((ret = decoder_start(&is->auddec, audio_thread, "audio_decoder", is)) < 0) goto out; // 0:播放 非0：暂停 SDL_PauseAudioDevice(audio_dev, 0); break;
打开音频设备
static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb_channels, int wanted_sample_rate, struct AudioParams *audio_hw_params) { SDL_AudioSpec wanted_spec, spec; const char *env; static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6}; static const int next_sample_rates[] = {0, 44100, 48000, 96000, 192000}; int next_sample_rate_idx = FF_ARRAY_ELEMS(next_sample_rates) - 1; env = SDL_getenv("SDL_AUDIO_CHANNELS"); if (env) { // 若环境变量有设置，优先从环境变量取得声道数和声道布局 wanted_nb_channels = atoi(env); wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels); } if (!wanted_channel_layout || wanted_nb_channels != av_get_channel_layout_nb_channels(wanted_channel_layout)) { wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels); wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX; } // 根据channel_layout获取nb_channels，当传入参数wanted_nb_channels不匹配时，此处会作修正 wanted_nb_channels = av_get_channel_layout_nb_channels(wanted_channel_layout); wanted_spec.channels = wanted_nb_channels; wanted_spec.freq = wanted_sample_rate; if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) { av_log(NULL, AV_LOG_ERROR, "Invalid sample rate or channel count! "); return -1; } while (next_sample_rate_idx && next_sample_rates[next_sample_rate_idx] >= wanted_spec.freq) next_sample_rate_idx--; // 从采样率数组中找到第一个不大于传入参数wanted_sample_rate的值 // 音频采样格式有两大类型：planar和packed，假设一个双声道音频文件，一个左声道采样点记作L，一个右声道采样点记作R，则： // planar存储格式：(plane1)LLLLLLLL...LLLL (plane2)RRRRRRRR...RRRR // packed存储格式：(plane1)LRLRLRLR...........................LRLR // 在这两种采样类型下，又细分多种采样格式，如AV_SAMPLE_FMT_S16、AV_SAMPLE_FMT_S16P等， // 注意SDL2.0目前不支持planar格式 // channel_layout是int64_t类型，表示音频声道布局，每bit代表一个特定的声道，参考channel_layout.h中的定义，一目了然 // 数据量(bits/秒) = 采样率(Hz) * 采样深度(bit) * 声道数 wanted_spec.format = AUDIO_S16SYS; wanted_spec.silence = 0; /* * 一次读取多长的数据 * SDL_AUDIO_MAX_CALLBACKS_PER_SEC一秒最多回调次数，避免频繁的回调 * Audio buffer size in samples (power of 2) */ wanted_spec.samples = FFMAX(SDL_AUDIO_MIN_BUFFER_SIZE, 2 << av_log2(wanted_spec.freq / SDL_AUDIO_MAX_CALLBACKS_PER_SEC)); wanted_spec.callback = sdl_audio_callback; wanted_spec.userdata = opaque; // 打开音频设备并创建音频处理线程。期望的参数是wanted_spec，实际得到的硬件参数是spec // 1) SDL提供两种使音频设备取得音频数据方法： // a. push，SDL以特定的频率调用回调函数，在回调函数中取得音频数据 // b. pull，用户程序以特定的频率调用SDL_QueueAudio()，向音频设备提供数据。此种情况wanted_spec.callback=NULL // 2) 音频设备打开后播放静音，不启动回调，调用SDL_PauseAudio(0)后启动回调，开始正常播放音频 // SDL_OpenAudioDevice()第一个参数为NULL时，等价于SDL_OpenAudio() while (!(audio_dev = SDL_OpenAudioDevice(NULL, 0, &wanted_spec, &spec, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE | SDL_AUDIO_ALLOW_CHANNELS_CHANGE))) { av_log(NULL, AV_LOG_WARNING, "SDL_OpenAudio (%d channels, %d Hz): %s ", wanted_spec.channels, wanted_spec.freq, SDL_GetError()); wanted_spec.channels = next_nb_channels[FFMIN(7, wanted_spec.channels)]; if (!wanted_spec.channels) { wanted_spec.freq = next_sample_rates[next_sample_rate_idx--]; wanted_spec.channels = wanted_nb_channels; if (!wanted_spec.freq) { av_log(NULL, AV_LOG_ERROR, "No more combinations to try, audio open failed "); return -1; } } wanted_channel_layout = av_get_default_channel_layout(wanted_spec.channels); } // 检查打开音频设备的实际参数：采样格式 if (spec.format != AUDIO_S16SYS) { av_log(NULL, AV_LOG_ERROR, "SDL advised audio format %d is not supported! ", spec.format); return -1; } // 检查打开音频设备的实际参数：声道数 if (spec.channels != wanted_spec.channels) { wanted_channel_layout = av_get_default_channel_layout(spec.channels); if (!wanted_channel_layout) { av_log(NULL, AV_LOG_ERROR, "SDL advised channel count %d is not supported! ", spec.channels); return -1; } } // wanted_spec是期望的参数，spec是实际的参数，wanted_spec和spec都是SDL中的结构。 // 此处audio_hw_params是FFmpeg中的参数，输出参数供上级函数使用 // audio_hw_params保存的参数，就是在做重采样的时候要转成的格式。 audio_hw_params->fmt = AV_SAMPLE_FMT_S16; audio_hw_params->freq = spec.freq; audio_hw_params->channel_layout = wanted_channel_layout; audio_hw_params->channels = spec.channels; /* audio_hw_params->frame_size这里只是计算一个采样点占用的字节数 */ audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels, 1, audio_hw_params->fmt, 1); audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->channels, audio_hw_params->freq, audio_hw_params->fmt, 1); if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) { av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed "); return -1; } // 比如2帧数据，一帧就是1024个采样点， 1024*2*2 * 2 = 8192字节 return spec.size; /* SDL内部缓存的数据字节, samples * channels *byte_per_sample */ }
读取数据
static void sdl_audio_callback(void *opaque, Uint8 *stream, int len) { VideoState *is = opaque; int audio_size, len1; audio_callback_time = av_gettime_relative(); while (len > 0) { // 循环读取，直到读取到足够的数据 /* (1)如果is->audio_buf_index < is->audio_buf_size则说明上次拷贝还剩余一些数据， * 先拷贝到stream再调用audio_decode_frame * (2)如果audio_buf消耗完了，则调用audio_decode_frame重新填充audio_buf */ if (is->audio_buf_index >= is->audio_buf_size) { audio_size = audio_decode_frame(is); if (audio_size < 0) { /* if error, just output silence */ is->audio_buf = NULL; is->audio_buf_size = SDL_AUDIO_MIN_BUFFER_SIZE / is->audio_tgt.frame_size * is->audio_tgt.frame_size; } else { if (is->show_mode != SHOW_MODE_VIDEO) update_sample_display(is, (int16_t *)is->audio_buf, audio_size); is->audio_buf_size = audio_size; // 讲字节 多少字节 } is->audio_buf_index = 0; } //根据缓冲区剩余大小量力而行 len1 = is->audio_buf_size - is->audio_buf_index; if (len1 > len) // len = 3000 < len1 4096 len1 = len; //根据audio_volume决定如何输出audio_buf /* 判断是否为静音，以及当前音量的大小，如果音量为最大则直接拷贝数据 */ if (!is->muted && is->audio_buf && is->audio_volume == SDL_MIX_MAXVOLUME) memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1); else { memset(stream, 0, len1); // 3.调整音量 /* 如果处于mute状态则直接使用stream填0数据, 暂停时is->audio_buf = NULL */ if (!is->muted && is->audio_buf) SDL_MixAudioFormat(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, AUDIO_S16SYS, len1, is->audio_volume); } len -= len1; stream += len1; /* 更新is->audio_buf_index，指向audio_buf中未被拷贝到stream的数据（剩余数据）的起始位置 */ is->audio_buf_index += len1; } is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index; /* Let's assume the audio driver that is used by SDL has two periods. */ if (!isnan(is->audio_clock)) { set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / is->audio_tgt.bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0); sync_clock_to_slave(&is->extclk, &is->audclk); } }