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在上一篇文章中，我们叙述了直播技术的环境配置（包括服务端nginx，nginx-rtmp-module， ffmpeg, android编译，ios编译）。从本文开始，我们将叙述播放相关的东西，播放是直播技术中关键的一步，它包括很多技术如：解码，缩放，时间基线选择，缓存队列，画面渲染，声音播放等等。我将分为三个部分为大家讲述整个播放流程；
Android
第一部分是基于NativeWindow的视频渲染，主要使用的OpenGL ES2通过传入surface来将视频数据渲染到surface上显示出来。第二部分是基于OpenSL ES来音频播放。第三部分，音视频同步。我们使用的都是android原生自带的一些库来做音视频渲染处理。
IOS
同样IOS也分成三个部分，第一部分视频渲染：使用OpenGLES.framework，通过OpenGL来渲染视频画面，第二部分是音频播放，基于AudioToolbox.framework做音频播放；第三部分，视音频同步。
利用原生库可以减少资源的利用，降低内存，提高性能；一般而言，如果不是通晓android、ios的程序员会选择一个统一的视频显示和音频播放库（SDL），这个库可以实现视频显示和音频播。但是增加额外的库意味着资源的浪费和性能的降低。
我们首先带来android端的视频播放功能，我们分成三个部分，1、视频渲染；2、音频播放；3、时间基线（音视频同步）来阐述。
ffmpeg为我们提供浏览丰富的编解码类型（ffmpeg所具备编解码能力都是软件编解码，不是指硬件编解码。具体之后文章会详细介绍ffmpeg），视频解码包括flv, mpeg, mov 等；音频包括aac, mp3等。对于整个播放，FFmpeg主要处理流程如下：
av_register_all(); // 注册所有的文件格式和编解码器的库,打开的合适格式的文件上会自动选择相应的编解码库
avformat_network_init(); // 注册网络服务
avformat_alloc_context(); // 分配FormatContext内存，
avformat_open_input(); // 打开输入流，获取头部信息，配合av_close_input_file（）关闭流
avformat_find_stream_info(); // 读取packets，来获取流信息，并在pFormatCtx->streams 填充上正确的信息
avcodec_find_decoder(); // 获取解码器，
avcodec_open2(); // 通过AVCodec来初始化AVCodecContext
av_read_frame(); // 读取每一帧
avcodec_decode_video2(); // 解码帧数据
avcodec_close(); // 关闭编辑器上下文
avformat_close_input(); // 关闭文件流我们先来看一段代码：
av_register_all();
avformat_network_init();
pFormatCtx = avformat_alloc_context();
if (avformat_open_input(&pFormatCtx, pathStr, NULL, NULL) != 0) {
LOGE("Couldn‘t open file: %s\n", pathStr);
return;
}
if (avformat_find_stream_info(pFormatCtx, &dictionary) < 0) {
LOGE("Couldn‘t find stream information.");
return;
}
av_dump_format(pFormatCtx, 0, pathStr, 0);
这段代码可以算是初始化FFmpeg，首先注册编解码库，为FormatContext分配内存，调用avformat_open_input打开输入流，获取头部信息，配合avformat_find_stream_info来填充FormatContext中相关内容，av_dump_format这个是dump出流信息。这个信息是这个样子的：
video infomation：
Input #0, flv, from ‘rtmp:127.0.0.1:1935/live/steam‘:
Metadata:
Server : NGINX RTMP (github.com/sergey-dryabzhinsky/nginx-rtmp-module)
displayWidth : 320
displayHeight : 240
fps : 15
profile :
level :
Duration: 00:00:00.00, start: 15.400000, bitrate: N/A
Stream #0:0: Video: flv1 (flv), yuv420p, 320x240, 15 tbr, 1k tbn, 1k tbc
Stream #0:1: Audio: mp3, 11025 Hz, stereo, s16p, 32 kb/s接下来就是找到视频的解码器上下文AVCodecContext和AVCodec解码器。
videoStream = -1;
for (int i = 0; i < pFormatCtx->nb_streams; i++) {
if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
videoStream = i;
break;
}
}
if (videoStream == -1) {
LOGE("Didn‘t find a video stream.");
return;
}
/**
获取视频解码器上下文和解码器
*/
pVideoCodecCtx = pFormatCtx->streams[videoStream]->codec;
pVideoCodec = avcodec_find_decoder(pVideoCodecCtx->codec_id);
if (pVideoCodec == NULL) {
LOGE("pVideoCodec not found.");
return;
}
/**
获取视频宽度和高度
*/
width = pVideoCodecCtx->width;
height = pVideoCodecCtx->height;
/**
获取设置NativeWindow buffer属性，
*/
ANativeWindow_setBuffersGeometry(nativeWindow, width, height, WINDOW_FORMAT_RGBA_8888);
LOGD("the width is %d, the height is %d", width, height);
// 分配每一帧的内存，pFrame原始帧，pFrameRGB为转换帧
pFrame = avcodec_alloc_frame();
pFrameRGB = avcodec_alloc_frame();
if (pFrameRGB == NULL || pFrameRGB == NULL) {
LOGE("Could not allocate video frame.");
return;
}
int numBytes = avpicture_get_size(PIX_FMT_RGBA, pVideoCodecCtx->width,
pVideoCodecCtx->height);
buffer = (uint8_t *) av_malloc(numBytes * sizeof(uint8_t));
if (buffer == NULL) {
LOGE("buffer is null");
return;
}
// 填充AVPicture信息
avpicture_fill((AVPicture *) pFrameRGB, buffer, PIX_FMT_RGBA,
pVideoCodecCtx->width, pVideoCodecCtx->height);
// 获取视频缩放上下文
pSwsCtx = sws_getContext(pVideoCodecCtx->width,
pVideoCodecCtx->height,
pVideoCodecCtx->pix_fmt,
width,
height,
PIX_FMT_RGBA,
SWS_BILINEAR,
NULL,
NULL,
NULL);
int frameFinished = 0;
// 循环读取每一帧
while (av_read_frame(pFormatCtx, &packet) >= 0) {
if (packet.stream_index == videoStream) {
// 解码每一帧数据
avcodec_decode_video2(pVideoCodecCtx, pFrame, &frameFinished, &packet);
if (frameFinished) {
LOGD("av_read_frame");
// 调用NativeWindows展示画面
ANativeWindow_lock(nativeWindow, &windowBuffer, 0);
// 缩放视频
sws_scale(pSwsCtx, (uint8_t const * const *)pFrame->data,
pFrame->linesize, 0, pVideoCodecCtx->height,
pFrameRGB->data, pFrameRGB->linesize);
uint8_t * dst = (uint8_t *)windowBuffer.bits;
int dstStride = windowBuffer.stride * 4;
uint8_t * src = (pFrameRGB->data[0]);
int srcStride = pFrameRGB->linesize[0];
for (int h = 0; h < height; h++) {
memcpy(dst + h * dstStride, src + h * srcStride, srcStride);
}
ANativeWindow_unlockAndPost(nativeWindow);
}
}
av_free_packet(&packet); // 释放packet
}从整个解码到缩放，再到渲染到nativeWindow思路非常清晰，当然我们这里没有考虑时间基线，这就意味着播放的时候会很快或者很慢（这个取决于视频的帧率）。我们在第三部分详细讨论时间基线（音视频同步问题）。
音频渲染我们同样适用的是android原生的库，OpenSL ES。我们会使用一些主要的接口参数如下：
// engine interfaces
static SLObjectItf engineObject;
static SLEngineItf engineEngine;
// output mix interfaces
static SLObjectItf outputMixObject;
static SLEnvironmentalReverbItf outputMixEnvironmentalReverb;
static SLObjectItf bqPlayerObject;
static SLEffectSendItf bqPlayerEffectSend;
static SLMuteSoloItf bqPlayerMuteSolo;
static SLVolumeItf bqPlayerVolume;
static SLPlayItf bqPlayerPlay;
static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue;
// aux effect on the output mix, used by the buffer queue player
const static SLEnvironmentalReverbSettings reverbSettings = SL_I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR;这些参数都是音频播放过程使用到的，而整个音频播放的流程主要包括以下几个方面：
//创建播放引擎，初始化接口参数。
bool createEngine() {
SLresult result;
// 创建引擎engineObject
result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
if (SL_RESULT_SUCCESS != result) {
return false;
}
// 实现引擎engineObject
result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
if (SL_RESULT_SUCCESS != result) {
return false;
}
// 获取引擎接口engineEngine
result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE,
&engineEngine);
if (SL_RESULT_SUCCESS != result) {
return false;
}
//
const SLInterfaceID ids[1] = {SL_IID_ENVIRONMENTALREVERB};
const SLboolean req[1] = {SL_BOOLEAN_FALSE};
// 创建混音器outputMixObject
result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 1,
ids, req);
if (SL_RESULT_SUCCESS != result) {
return false;
}
// 实现混音器outputMixObject
result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
if (SL_RESULT_SUCCESS != result) {
return false;
}
// 获取混音器接口outputMixEnvironmentalReverb
result = (*outputMixObject)->GetInterface(outputMixObject,
SL_IID_ENVIRONMENTALREVERB,
&outputMixEnvironmentalReverb);
if (SL_RESULT_SUCCESS == result) {
result = (*outputMixEnvironmentalReverb)->SetEnvironmentalReverbProperties(
outputMixEnvironmentalReverb, &reverbSettings);
}
return true;
}在创建OpenSL ES音频播放引擎的时候，我们主要针对引擎和混音器进行初始化。之后我们会创建音频播放缓冲。
bool createBufferQueueAudioPlayer(PlayCallBack callback) {
SLresult result;
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
// pcm数据格式
SLDataFormat_PCM format_pcm = {SL_DATAFORMAT_PCM, 1, SL_SAMPLINGRATE_44_1,
SL_PCMSAMPLEFORMAT_FIXED_16, SL_PCMSAMPLEFORMAT_FIXED_16,
SL_SPEAKER_FRONT_CENTER, SL_BYTEORDER_LITTLEENDIAN};
SLDataSource audioSrc = {&loc_bufq, &format_pcm};
SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject};
SLDataSink audioSnk = {&loc_outmix, NULL};
const SLInterfaceID ids[3] = {SL_IID_BUFFERQUEUE, SL_IID_EFFECTSEND, SL_IID_VOLUME};
const SLboolean req[3] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
// 创建音频播放器
result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject,
&audioSrc, &audioSnk, 3, ids, req);
if (SL_RESULT_SUCCESS != result)
return false;
// 实现音频播放器（bqPlayerObject）
result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
if (SL_RESULT_SUCCESS != result)
return false;
// 获取音频播放器接口bqPlayerPlay（获取播放器）
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY,
&bqPlayerPlay);
if (SL_RESULT_SUCCESS != result)
return false;
// 获取音频播放器接口bqPlayerBufferQueue（缓冲buffer）
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE,
&bqPlayerBufferQueue);
if (SL_RESULT_SUCCESS != result)
return false;
// 注册播放回调 callback
result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue,
callback, NULL);
if (SL_RESULT_SUCCESS != result) {
return false;
}
// 获取音频播放器接口bqPlayerEffectSend（音效）
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_EFFECTSEND,
&bqPlayerEffectSend);
if (SL_RESULT_SUCCESS != result)
return false;
// 获取音频播放器接口bqPlayerVolume（音量）
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME,
&bqPlayerVolume);
if (SL_RESULT_SUCCESS != result)
return false;
// 设置播放bqPlayerPlay状态（暂停）
result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_STOPPED);
return result == SL_RESULT_SUCCESS;
}整个音频播放流畅其实看起来也是很简单的，主要分：1、创建实现播放引擎；2、创建实现混音器；3、设置缓冲和pcm格式；4、创建实现播放器；5、获取音频播放器接口；6、获取缓冲buffer；7、注册播放回调；8、获取音效接口；9、获取音量接口；10、获取播放状态接口；
做完这10步，整个音频播放器引擎就创建完毕，接下来就是引擎读取数据播放。
void playBuffer(void *pBuffer, int size) {
// 判断数据可用性
if (pBuffer == NULL || size == -1) {
return;
}
LOGV("PlayBuff!");
// 数据存放进bqPlayerBufferQueue中
SLresult result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue,
pBuffer, size);
if (result != SL_RESULT_SUCCESS)
LOGE("Play buffer error!");
}这段代码主要阐述的播放的过程，通过将数据放进bqPlayerBufferQueue，供播放引擎读取播放。记得我们在创建缓冲buffer的时候，注册了一个callback，这个callBack的作用就是通知可以向缓冲队列中添加数据，这个callBack的原型如下：
void videoPlayCallBack(SLAndroidSimpleBufferQueueItf bq, void *context) {
// 添加数据到bqPlayerBufferQueue中，通过调用playBuffer方法。
void* data = getData();
int size = getDataSize();
playBuffer(data, size);
}这样就循环向缓冲队列中添加数据，可以一直播放。有很多简单的音乐播放器就是基于这种模式设计的。这种方式也很可靠，能够非常清楚的展现整个播放流程。
为了能够完整的播放视频和音频，我们需要对一些数据进行整合，包括音视频数据，时间参数等。在ffmpeg源码的ffplayer.c使用就是这种方式，对齐时间基线有三种方式：1、对齐音频时间基线；2、对齐视频时间基线；3、对齐第三方时间基线。为了能够能够清楚展现播放速度控制和音视频同步，特意给出了一个流程图。音视频队列从ffmpeg的av_read_frame中读取每一帧数据到音视频队列中（保存的是AVPacket数据），然后音频队列和视频队列从音视频队列中不停的拿数据，期间两者做一个同步，最后展示到画面和播放音频。
void getPacket() {
struct timespec time;
time.tv_sec = 10;//网络不好最多等10秒
time.tv_nsec = 0;
struct ThreadMsg msg;
while (true) {
memset(&msg, 0, sizeof(struct ThreadMsg));
msg.data = NULL;
// 等待网络缓冲
ThreadList::queueGet(playInstance->queue, &time, &msg);
if (msg.msgtype == -1) {//正常退出
ThreadList::queueAdd(playInstance->video_queue, NULL, -1);
ThreadList::queueAdd(playInstance->audio_queue, NULL, -1);
break;
}
if (msg.data == NULL) {
ThreadList::queueAdd(playInstance->video_queue, NULL, -1);
ThreadList::queueAdd(playInstance->audio_queue, NULL, -1);
playInstance->timeout_flag = 1;
break;
}
AVPacket *packet_p = (AVPacket *) msg.data;
if (packet_p->stream_index == playInstance->videoState->videoStream) {
ThreadList::queueAdd(playInstance->video_queue, packet_p, 1);
} else if (packet_p->stream_index == playInstance->videoState->audioStream) {
ThreadList::queueAdd(playInstance->audio_queue, packet_p, 1);
}
}
}int queueGet(struct ThreadQueue *queue, const struct timespec *timeout,
struct ThreadMsg *msg) {
struct MsgList *firstrec;
int ret = 0;
struct timespec abstimeout;
if (queue == NULL || msg == NULL) {
return EINVAL;
}
if (timeout) {
struct timeval now;
gettimeofday(&now, NULL);
abstimeout.tv_sec = now.tv_sec + timeout->tv_sec;
abstimeout.tv_nsec = (now.tv_usec * 1000) + timeout->tv_nsec;
if (abstimeout.tv_nsec >= 1000000000) {
abstimeout.tv_sec++;
abstimeout.tv_nsec -= 1000000000;
}
}
pthread_mutex_lock(&queue->mutex);
/* Will wait until awakened by a signal or broadcast */
while (queue->first == NULL && ret != ETIMEDOUT) { //Need to loop to handle spurious wakeups
if (timeout) {
ret = pthread_cond_timedwait(&queue->cond, &queue->mutex, &abstimeout);
} else {
pthread_cond_wait(&queue->cond, &queue->mutex);
}
}
if (ret == ETIMEDOUT) {
pthread_mutex_unlock(&queue->mutex);
return ret;
}
firstrec = queue->first;
queue->first = queue->first->next;
queue->length--;
if (queue->first == NULL) {
queue->last = NULL; // we know this since we hold the lock
queue->length = 0;
}
msg->data = firstrec->msg.data;
msg->msgtype = firstrec->msg.msgtype;
msg->qlength = queue->length;
release_msglist(queue, firstrec);
pthread_mutex_unlock(&queue->mutex);
return 0;
}我们分三个步骤进行操作。
typedef struct VideoState { //解码过程中的数据结构
AVFormatContext *pFormatCtx;
AVCodecContext *pVideoCodecCtx; // 视频解码器上下文
AVCodecContext *pAudioCodecCtx; // 音频解码器上下文
AVCodec *pAudioCodec; // 音频解码器
AVCodec *pVideoCodec; // 视频解码器
AVFrame *pVideoFrame; // 视频为转码帧
AVFrame *pVideoFrameRgba; // 视频转码为rgba数据的帧
struct SwsContext *sws_ctx; // 视频转码工具
void *pVideobuffer; // 视频缓存
AVFrame *pAudioFrame;// 音频帧
struct SwrContext *swr_ctx; // 音频转码工具
int sample_rate_src; //音频采样率
int sample_fmt; // 音频采用格式
int sample_layout; // 音频通道数
int64_t video_start_time; // 视频开始时间
int64_t audio_start_time; // 音频时间
double video_time_base; // 视频基准时间
double audio_time_base; // 音频基准时间
int videoStream; // FormatContext.streams中视频帧位置
int audioStream; // FormatContext.streams中音频帧位置
} VideoState;
typedef struct PlayInstance {
ANativeWindow *window; // nativeWindow // 通过传入surface构建
int display_width; // 显示宽度
int display_height; // 显示高度
int stop; // 停止
int timeout_flag; // 超时标记
int disable_video;
VideoState *videoState;
//队列
struct ThreadQueue *queue; // 音视频帧队列
struct ThreadQueue *video_queue; // 视频帧队列
struct ThreadQueue *audio_queue; // 音频帧队列
} PlayInstance;播放速度控制
播放速度控制可以通过获取音、视频队列中的数据来控制时间，我们是在单独的线程中从音、视频队列中获取相应的数据。然后根据数据时间基准线进行播放速度控制。这一块结合音视频同步一起实现。
音视频同步
下面代码是视频展示过程（在单独线程中做的），我们主要考虑的是延时同步那一块。将音频队列中的数据分类存放的音频队列和视频队列。
void video_thread() {
struct timespec time;
time.tv_sec = 10;//网络不好最多等10秒
time.tv_nsec = 0;
struct ThreadMsg msg;
int packet_count = 0;
while (true) {
if (playInstance->stop) {
break;
}
msg.data = NULL;
// 从视频队列中获取数据，并等待
ThreadList::queueGet(playInstance->video_queue, &time, &msg);
if (msg.msgtype == -1) {
break;
}
if (msg.data == NULL) {
LOGE("视频线程空循环\n");
break;
}
AVPacket *packet_p = (AVPacket *) msg.data;
AVPacket pavpacket = *packet_p;
packet_count++;
if (packet_count == 1) {//拿到第一个视频包
playInstance->videoState->video_start_time = av_gettime();
LOGE("视频开始时间 %lld\n", playInstance->videoState->video_start_time);
}
if (playInstance->disable_video) {
av_free_packet(packet_p);
av_free(msg.data);
continue;
}
ANativeWindow_Buffer windowBuffer;
// 延时同步，控制播放速度。
int64_t pkt_pts = pavpacket.pts;
double show_time = pkt_pts * (playInstance->videoState->video_time_base);
int64_t show_time_micro = show_time * 1000000;
int64_t played_time = av_gettime() - playInstance->videoState->video_start_time;
int64_t delta_time = show_time_micro - played_time;
if (delta_time < -(0.2 * 1000000)) {
LOGE("视频跳帧\n");
continue；
} else if (delta_time > 0) {
av_usleep(delta_time);
}
int frame_finished = 0;
avcodec_decode_video2(playInstance->videoState->pVideoCodecCtx,
playInstance->videoState->pVideoFrame,
&frame_finished, &pavpacket);
if (frame_finished) {
sws_scale(//对解码后的数据进行色彩空间转换，yuv420p 转为rgba8888
playInstance->videoState->sws_ctx,
(uint8_t const *const *) (playInstance->videoState->pVideoFrame)->data,
(playInstance->videoState->pVideoFrame)->linesize,
0,
playInstance->videoState->pVideoCodecCtx->height,
playInstance->videoState->pVideoFrameRgba->data,
playInstance->videoState->pVideoFrameRgba->linesize
);
if (!(playInstance->disable_video) &&
ANativeWindow_lock(playInstance->window, &windowBuffer, NULL) < 0) {
LOGE("cannot lock window");
continue;
} else if (!playInstance->disable_video) {
uint8_t *dst = (uint8_t *) windowBuffer.bits;
int dstStride = windowBuffer.stride * 4;
uint8_t *src = (playInstance->videoState->pVideoFrameRgba->data[0]);
int srcStride = playInstance->videoState->pVideoFrameRgba->linesize[0];
for (int h = 0; h < playInstance->display_height; h++) {
memcpy(dst + h * dstStride, src + h * srcStride, srcStride);
}
ANativeWindow_unlockAndPost(playInstance->window);//释放对surface的锁，并且更新对应surface数据进行显示
}
av_free_packet(packet_p);
av_free(msg.data);
}
}
}我们主要分析延时同步的那一段代码：
// 延时同步
int64_t pkt_pts = pavpacket.pts;
double show_time = pkt_pts * (playInstance->videoState->video_time_base);
int64_t show_time_micro = show_time * 1000000;
int64_t played_time = av_gettime() - playInstance->videoState->video_start_time;
int64_t delta_time = show_time_micro - played_time;
if (delta_time < -(0.2 * 1000000)) {
LOGE("视频跳帧\n");
continue；
} else if (delta_time > 0.2 * 1000000) {
av_usleep(delta_time);
}这段代码主要是音视频同步的，这块采用的是基于第三方时间基准，同步调整音频和视频。在音频处理中也有类似的代码。针对延时做同步。这个基准线是标准时间，通过修改过时间差值来设置跳帧还是等待。当然这里音视频同步实现比较简单，按照正常的使用应该是在一定范围内可以认为是同步的。
由于声音对于人来说比较敏感，一点杂音都能分辨出来，而视频的跳帧相对来说更容易让人接受，这是因为视觉停留的原因。因此，建议采用基于音频时间基准来进行音视频同步。
IOS相对于android而言在视频渲染上来说可能比android端稍微复杂，因为ios没有像android的surfaceView可以直接进行操作，都是通过OpenGL来绘制画面。因此可能会比较难于理解。而对于音频，IOS可以采用AudioToolbox进行处理。对于ffmpeg相关的东西android和ios是一样的，也主要是一个流程。音视频同步采用的是同一种方案。在此就不在介绍ios的音视频同步问题。
前面在android部分已经阐述到使用av_read_frame方法来读取每一帧的信息，这部分ios和android一样，android由于原生支持surfaceView进行绘制渲染，而ios不支持，需要借助opengl来绘制，因此在av_read_frame之后就和android存在区别。主要区别如下代码：
- (NSArray *) decodeFrames: (CGFloat) minDuration
{
if (_videoStream == -1 &&
_audioStream == -1)
return nil;
if (_formatCtx == nil) {
printf("AvFormatContext is nil");
return nil;
}
NSMutableArray *result = [NSMutableArray array];
AVPacket packet;
CGFloat decodedDuration = 0;
BOOL finished = NO;
while (!finished) {
if (av_read_frame(_formatCtx, &packet) < 0) {
_isEOF = YES;
break;
}
if (packet.stream_index ==_videoStream) {
int pktSize = packet.size;
while (pktSize > 0) {
int gotframe = 0;
int len = avcodec_decode_video2(_videoCodecCtx,
_videoFrame,
&gotframe,
&packet);
if (len < 0) {
LoggerVideo(0, @"decode video error, skip packet");
break;
}
if (gotframe) {
if (!_disableDeinterlacing &&
_videoFrame->interlaced_frame) {
avpicture_deinterlace((AVPicture*)_videoFrame,
(AVPicture*)_videoFrame,
_videoCodecCtx->pix_fmt,
_videoCodecCtx->width,
_videoCodecCtx->height);
}
KxVideoFrame *frame = [self handleVideoFrame];
if (frame) {
[result addObject:frame];
_position = frame.position;
decodedDuration += frame.duration;
if (decodedDuration > minDuration)
finished = YES;
}
char *buf = (char *)malloc(_videoFrame->width * _videoFrame->height * 3 / 2);
AVPicture *pict;
int w, h;
char *y, *u, *v;
pict = (AVPicture *)_videoFrame;//这里的frame就是解码出来的AVFrame
w = _videoFrame->width;
h = _videoFrame->height;
y = buf;
u = y + w * h;
v = u + w * h / 4;
for (int i=0; i<h; i++)
memcpy(y + w * i, pict->data[0] + pict->linesize[0] * i, w);
for (int i=0; i<h/2; i++)
memcpy(u + w / 2 * i, pict->data[1] + pict->linesize[1] * i, w / 2);
for (int i=0; i<h/2; i++)
memcpy(v + w / 2 * i, pict->data[2] + pict->linesize[2] * i, w / 2);
[myview setVideoSize:_videoFrame->height height:_videoFrame->width];
[myview displayYUV420pData:buf width:_videoFrame->width height:_videoFrame->height];
free(buf);
}
if (0 == len)
break;
pktSize -= len;
}
} else if (packet.stream_index == _audioStream) {
int pktSize = packet.size;
while (pktSize > 0) {
int gotframe = 0;
int len = avcodec_decode_audio4(_audioCodecCtx,
_audioFrame,
&gotframe,
&packet);
if (len < 0) {
LoggerAudio(0, @"decode audio error, skip packet");
break;
}
if (gotframe) {
KxAudioFrame * frame = [self handleAudioFrame];
if (frame) {
[result addObject:frame];
if (_videoStream == -1) {
_position = frame.position;
decodedDuration += frame.duration;
if (decodedDuration > minDuration)
finished = YES;
}
}
}
if (0 == len)
break;
pktSize -= len;
}
} else if (packet.stream_index == _artworkStream) {
if (packet.size) {
KxArtworkFrame *frame = [[KxArtworkFrame alloc] init];
frame.picture = [NSData dataWithBytes:packet.data length:packet.size];
[result addObject:frame];
}
} else if (packet.stream_index == _subtitleStream) {
int pktSize = packet.size;
while (pktSize > 0) {
AVSubtitle subtitle;
int gotsubtitle = 0;
int len = avcodec_decode_subtitle2(_subtitleCodecCtx,
&subtitle,
&gotsubtitle,
&packet);
if (len < 0) {
LoggerStream(0, @"decode subtitle error, skip packet");
break;
}
if (gotsubtitle) {
KxSubtitleFrame *frame = [self handleSubtitle: &subtitle];
if (frame) {
[result addObject:frame];
}
avsubtitle_free(&subtitle);
}
if (0 == len)
break;
pktSize -= len;
}
}
av_free_packet(&packet);
}
return result;
}
这段代码主要描述的是decodeFrames，将视频帧和音频帧单独处理。视频主要代码如下：
if (packet.stream_index ==_videoStream) {
int pktSize = packet.size;
while (pktSize > 0) {
int gotframe = 0;
int len = avcodec_decode_video2(_videoCodecCtx,
_videoFrame,
&gotframe,
&packet);
if (len < 0) {
LoggerVideo(0, @"decode video error, skip packet");
break;
}
if (gotframe) {
if (!_disableDeinterlacing &&
_videoFrame->interlaced_frame) {
avpicture_deinterlace((AVPicture*)_videoFrame,
(AVPicture*)_videoFrame,
_videoCodecCtx->pix_fmt,
_videoCodecCtx->width,
_videoCodecCtx->height);
}
VideoFrame *frame = [self handleVideoFrame];
if (frame) {
[result addObject:frame];
_position = frame.position;
decodedDuration += frame.duration;
if (decodedDuration > minDuration)
finished = YES;
}
char *buf = (char *)malloc(_videoFrame->width * _videoFrame->height * 3 / 2);
AVPicture *pict;
int w, h;
char *y, *u, *v;
pict = (AVPicture *)_videoFrame;//这里的frame就是解码出来的AVFrame
w = _videoFrame->width;
h = _videoFrame->height;
y = buf;
u = y + w * h;
v = u + w * h / 4;
for (int i=0; i<h; i++)
memcpy(y + w * i, pict->data[0] + pict->linesize[0] * i, w);
for (int i=0; i<h/2; i++)
memcpy(u + w / 2 * i, pict->data[1] + pict->linesize[1] * i, w / 2);
for (int i=0; i<h/2; i++)
memcpy(v + w / 2 * i, pict->data[2] + pict->linesize[2] * i, w / 2);
[myview setVideoSize:_videoFrame->height height:_videoFrame->width];
[myview displayYUV420pData:buf width:_videoFrame->width height:_videoFrame->height];
free(buf);
}
if (0 == len)
break;
pktSize -= len;
}
} else if (packet.stream_index == _audioStream) {
int pktSize = packet.size;
while (pktSize > 0) {
int gotframe = 0;
int len = avcodec_decode_audio4(_audioCodecCtx,
_audioFrame,
&gotframe,
&packet);
if (len < 0) {
LoggerAudio(0, @"decode audio error, skip packet");
break;
}
if (gotframe) {
AudioFrame * frame = [self handleAudioFrame];
if (frame) {
[result addObject:frame];
if (_videoStream == -1) {
_position = frame.position;
decodedDuration += frame.duration;
if (decodedDuration > minDuration)
finished = YES;
}
}
}
if (0 == len)
break;
pktSize -= len;
}
}这块代码和android的类似，将数据转化为yuv格式，然后拿去渲染，myview是一个View，它主要就是通过OpenGL 将数据渲染成画面。主要代码：
- (void)displayYUV420pData:(void *)data width:(NSInteger)w height:(NSInteger)h
{
@synchronized(self)
{
if (w != _videoW || h != _videoH)
{
[self setVideoSize:w height:h];
}
[EAGLContext setCurrentContext:_glContext];
glBindTexture(GL_TEXTURE_2D, _textureYUV[TEXY]);
glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, w, h, GL_RED_EXT, GL_UNSIGNED_BYTE, data);
//[self debugGlError];
glBindTexture(GL_TEXTURE_2D, _textureYUV[TEXU]);
glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, w/2, h/2, GL_RED_EXT, GL_UNSIGNED_BYTE, data + w * h);
// [self debugGlError];
glBindTexture(GL_TEXTURE_2D, _textureYUV[TEXV]);
glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, w/2, h/2, GL_RED_EXT, GL_UNSIGNED_BYTE, data + w * h * 5 / 4);
//[self debugGlError];
[self render];
}
#ifdef DEBUG
GLenum err = glGetError();
if (err != GL_NO_ERROR)
{
printf("GL_ERROR=======>%d\n", err);
}
struct timeval nowtime;
gettimeofday(&nowtime, NULL);
if (nowtime.tv_sec != _time.tv_sec)
{
printf("视频 %d 帧率: %d\n", self.tag, _frameRate);
memcpy(&_time, &nowtime, sizeof(struct timeval));
_frameRate = 1;
}
else
{
_frameRate++;
}
#endif
}
通过OpenGL将数据绘制出来。这就是整个IOS展示画面主要代码。其他的一些东西类似于android。
在上面视频渲染的过程中，列出了部分对音频的处理过程。其主要方法为handleAudioFrame，下面代码书handleAudioFrame方法：
- (AudioFrame *) handleAudioFrame
{
if (!_audioFrame->data[0])
return nil;
id<AudioManager> audioManager = [AudioManager audioManager];
const NSUInteger numChannels = audioManager.numOutputChannels;
NSInteger numFrames;
void * audioData;
if (_swrContext) {
const NSUInteger ratio = MAX(1, audioManager.samplingRate / _audioCodecCtx->sample_rate) *
MAX(1, audioManager.numOutputChannels / _audioCodecCtx->channels) * 2;
const int bufSize = av_samples_get_buffer_size(NULL,
audioManager.numOutputChannels,
_audioFrame->nb_samples * ratio,
AV_SAMPLE_FMT_S16,
1);
if (!_swrBuffer || _swrBufferSize < bufSize) {
_swrBufferSize = bufSize;
_swrBuffer = realloc(_swrBuffer, _swrBufferSize);
}
Byte *outbuf[2] = { _swrBuffer, 0 };
numFrames = swr_convert(_swrContext,
outbuf,
_audioFrame->nb_samples * ratio,
(const uint8_t **)_audioFrame->data,
_audioFrame->nb_samples);
if (numFrames < 0) {
LoggerAudio(0, @"fail resample audio");
return nil;
}
//int64_t delay = swr_get_delay(_swrContext, audioManager.samplingRate);
//if (delay > 0)
// LoggerAudio(0, @"resample delay %lld", delay);
audioData = _swrBuffer;
} else {
if (_audioCodecCtx->sample_fmt != AV_SAMPLE_FMT_S16) {
NSAssert(false, @"bucheck, audio format is invalid");
return nil;
}
audioData = _audioFrame->data[0];
numFrames = _audioFrame->nb_samples;
}
const NSUInteger numElements = numFrames * numChannels;
NSMutableData *data = [NSMutableData dataWithLength:numElements * sizeof(float)];
float scale = 1.0 / (float)INT16_MAX ;
vDSP_vflt16((SInt16 *)audioData, 1, data.mutableBytes, 1, numElements);
vDSP_vsmul(data.mutableBytes, 1, &scale, data.mutableBytes, 1, numElements);
AudioFrame *frame = [[AudioFrame alloc] init];
frame.position = av_frame_get_best_effort_timestamp(_audioFrame) * _audioTimeBase;
frame.duration = av_frame_get_pkt_duration(_audioFrame) * _audioTimeBase;
frame.samples = data;
if (frame.duration == 0) {
// sometimes ffmpeg can‘t determine the duration of audio frame
// especially of wma/wmv format
// so in this case must compute duration
frame.duration = frame.samples.length / (sizeof(float) * numChannels * audioManager.samplingRate);
}
#if DEBUG
LoggerAudio(2, @"AFD: %.4f %.4f | %.4f ",
frame.position,
frame.duration,
frame.samples.length / (8.0 * 44100.0));
#endif
return frame;
}handleAudioFrame主要方法是将数据组合转换，比如采样率重设，转码等等功能，最后放进一个数组中，最后交给AudioToolbox进行处理。我们接着放下看：
func enableAudio(on: Bool) {
let audioManager = AudioManager.audioManager()
if on && decoder.validAudio {
// 闭包
audioManager.outputBlock = {(outData: UnsafeMutablePointer<Float>,numFrames: UInt32, numChannels: UInt32) -> Void in
self.audioCallbackFillData(outData, numFrames: Int(numFrames), numChannels: Int(numChannels))
}
audioManager.play()
}else{
audioManager.pause()
audioManager.outputBlock = nil
}
}
func audioCallbackFillData(outData: UnsafeMutablePointer<Float>, numFrames: Int, numChannels: Int) {
var numFrames = numFrames
var weakOutData = outData
if buffered {
memset(weakOutData, 0, numFrames * numChannels * sizeof(Float))
return
}
autoreleasepool {
while numFrames > 0 {
if currentAudioFrame == nil {
dispatch_sync(lockQueue) {
if let count = self.audioFrames?.count {
if count > 0 {
let frame: AudioFrame = self.audioFrames![0] as! AudioFrame
self.audioFrames!.removeObjectAtIndex(0)
self.moviePosition = frame.position
self.bufferedDuration -= Float(frame.duration)
self.currentAudioFramePos = 0
self.currentAudioFrame = frame.samples
}
}
}
}
if (currentAudioFrame != nil) {
let bytes = currentAudioFrame.bytes + currentAudioFramePos
let bytesLeft: Int = (currentAudioFrame.length - currentAudioFramePos)
let frameSizeOf: Int = numChannels * sizeof(Float)
let bytesToCopy: Int = min(numFrames * frameSizeOf, bytesLeft)
let framesToCopy: Int = bytesToCopy / frameSizeOf
memcpy(weakOutData, bytes, bytesToCopy)
numFrames -= framesToCopy
weakOutData = weakOutData.advancedBy(framesToCopy * numChannels)
if bytesToCopy < bytesLeft {
self.currentAudioFramePos += bytesToCopy
}
else {
self.currentAudioFrame = nil
}
}else{
memset(weakOutData, 0, numFrames * numChannels * sizeof(Float))
//LoggerStream(1, @"silence audio");
self.debugAudioStatus = 3
self.debugAudioStatusTS = NSDate()
break
}
}
}
}
这是一段swift代码。在ios采用的是swift+oc+c++混合编译，正好借此熟悉swift于oc和c++的交互。enableAudio主要是创建一个audioManager实例，进行注册回调，和开始播放和暂停服务。audioManager是一个单例。是一个封装AudioToolbox类。下面的代码是激活AudioSession（初始化Audio）和失效AudioSession代码。
- (BOOL) activateAudioSession
{
if (!_activated) {
if (!_initialized) {
if (checkError(AudioSessionInitialize(NULL,
kCFRunLoopDefaultMode,
sessionInterruptionListener,
(__bridge void *)(self)),
"Couldn‘t initialize audio session"))
return NO;
_initialized = YES;
}
if ([self checkAudioRoute] &&
[self setupAudio]) {
_activated = YES;
}
}
return _activated;
}
- (void) deactivateAudioSession
{
if (_activated) {
[self pause];
checkError(AudioUnitUninitialize(_audioUnit),
"Couldn‘t uninitialize the audio unit");
/*
fails with error (-10851) ?
checkError(AudioUnitSetProperty(_audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input,
0,
NULL,
0),
"Couldn‘t clear the render callback on the audio unit");
*/
checkError(AudioComponentInstanceDispose(_audioUnit),
"Couldn‘t dispose the output audio unit");
checkError(AudioSessionSetActive(NO),
"Couldn‘t deactivate the audio session");
checkError(AudioSessionRemovePropertyListenerWithUserData(kAudioSessionProperty_AudioRouteChange,
sessionPropertyListener,
(__bridge void *)(self)),
"Couldn‘t remove audio session property listener");
checkError(AudioSessionRemovePropertyListenerWithUserData(kAudioSessionProperty_CurrentHardwareOutputVolume,
sessionPropertyListener,
(__bridge void *)(self)),
"Couldn‘t remove audio session property listener");
_activated = NO;
}
}
- (BOOL) setupAudio
{
// --- Audio Session Setup ---
UInt32 sessionCategory = kAudioSessionCategory_MediaPlayback;
//UInt32 sessionCategory = kAudioSessionCategory_PlayAndRecord;
if (checkError(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,
sizeof(sessionCategory),
&sessionCategory),
"Couldn‘t set audio category"))
return NO;
if (checkError(AudioSessionAddPropertyListener(kAudioSessionProperty_AudioRouteChange,
sessionPropertyListener,
(__bridge void *)(self)),
"Couldn‘t add audio session property listener"))
{
// just warning
}
if (checkError(AudioSessionAddPropertyListener(kAudioSessionProperty_CurrentHardwareOutputVolume,
sessionPropertyListener,
(__bridge void *)(self)),
"Couldn‘t add audio session property listener"))
{
// just warning
}
// Set the buffer size, this will affect the number of samples that get rendered every time the audio callback is fired
// A small number will get you lower latency audio, but will make your processor work harder
#if !TARGET_IPHONE_SIMULATOR
Float32 preferredBufferSize = 0.0232;
if (checkError(AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration,
sizeof(preferredBufferSize),
&preferredBufferSize),
"Couldn‘t set the preferred buffer duration")) {
// just warning
}
#endif
if (checkError(AudioSessionSetActive(YES),
"Couldn‘t activate the audio session"))
return NO;
[self checkSessionProperties];
// ----- Audio Unit Setup -----
// Describe the output unit.
AudioComponentDescription description = {0};
description.componentType = kAudioUnitType_Output;
description.componentSubType = kAudioUnitSubType_RemoteIO;
description.componentManufacturer = kAudioUnitManufacturer_Apple;
// Get component
AudioComponent component = AudioComponentFindNext(NULL, &description);
if (checkError(AudioComponentInstanceNew(component, &_audioUnit),
"Couldn‘t create the output audio unit"))
return NO;
UInt32 size;
// Check the output stream format
size = sizeof(AudioStreamBasicDescription);
if (checkError(AudioUnitGetProperty(_audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&_outputFormat,
&size),
"Couldn‘t get the hardware output stream format"))
return NO;
_outputFormat.mSampleRate = _samplingRate;
if (checkError(AudioUnitSetProperty(_audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&_outputFormat,
size),
"Couldn‘t set the hardware output stream format")) {
// just warning
}
_numBytesPerSample = _outputFormat.mBitsPerChannel / 8;
_numOutputChannels = _outputFormat.mChannelsPerFrame;
LoggerAudio(2, @"Current output bytes per sample: %ld", _numBytesPerSample);
LoggerAudio(2, @"Current output num channels: %ld", _numOutputChannels);
// Slap a render callback on the unit
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = renderCallback; // 注册回调，这个回调是用来取数据的，也就是
callbackStruct.inputProcRefCon = (__bridge void *)(self);
if (checkError(AudioUnitSetProperty(_audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input,
0,
&callbackStruct,
sizeof(callbackStruct)),
"Couldn‘t set the render callback on the audio unit"))
return NO;
if (checkError(AudioUnitInitialize(_audioUnit),
"Couldn‘t initialize the audio unit"))
return NO;
return YES;
}正真将数据渲染代码
- (BOOL) renderFrames: (UInt32) numFrames
ioData: (AudioBufferList *) ioData
{
for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
memset(ioData->mBuffers[iBuffer].mData, 0, ioData->mBuffers[iBuffer].mDataByteSize);
}
if (_playing && _outputBlock ) {
// Collect data to render from the callbacks
_outputBlock(_outData, numFrames, _numOutputChannels);
// Put the rendered data into the output buffer
if (_numBytesPerSample == 4) // then we‘ve already got floats
{
float zero = 0.0;
for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
int thisNumChannels = ioData->mBuffers[iBuffer].mNumberChannels;
for (int iChannel = 0; iChannel < thisNumChannels; ++iChannel) {
vDSP_vsadd(_outData+iChannel, _numOutputChannels, &zero, (float *)ioData->mBuffers[iBuffer].mData, thisNumChannels, numFrames);
}
}
}
else if (_numBytesPerSample == 2) // then we need to convert SInt16 -> Float (and also scale)
{
float scale = (float)INT16_MAX;
vDSP_vsmul(_outData, 1, &scale, _outData, 1, numFrames*_numOutputChannels);
for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
int thisNumChannels = ioData->mBuffers[iBuffer].mNumberChannels;
for (int iChannel = 0; iChannel < thisNumChannels; ++iChannel) {
vDSP_vfix16(_outData+iChannel, _numOutputChannels, (SInt16 *)ioData->mBuffers[iBuffer].mData+iChannel, thisNumChannels, numFrames);
}
}
}
}
return noErr;
}
本文主要是讲述了ffmpeg实现播放的逻辑，分为android和ios两端，根据两端平台的特性做了相应的处理。在android端采用的是NativeWindow（surface）实现视频播放，OpenSL ES实现音频播放。实现音视频同步的逻辑是基于第三方时间基准线，音频和视频同时调整的方案。在ios端采用的是OpenGL实现视频渲染，AudioToolbox实现音频播放。音视频同步和android采用的是一样。其中两端的ffmpeg逻辑是一致的。在ios端OpenGL实现视频渲染没有重点阐述如何使用OpenGL。这个有兴趣的同学可以自行研究。
备注：整个代码工程等整理之后会发布出来。
最后添加两张播放效果图
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原文地址：http://blog.csdn.net/xwl198937/article/details/52386331
