我正在解码一个OGG视频(theora& vorbis作为编解码器),并希望在播放声音时在屏幕上显示它(使用Ogre 3D).我可以很好地解码图像流,并且视频以正确的帧速率等完美播放.
但是,我无法通过OpenAL获得声音.
编辑：我设法使播放声音至少在某种程度上与视频中的实际音频相似.更新了示例代码.
编辑2：我现在能够获得“几乎”正确的声音.我必须将OpenAL设置为使用AL_FORMAT_STEREO_FLOAT32(初始化扩展后)而不是仅使用STEREO16.现在声音“只”极高音调和口吃,但速度正确.
以下是我解码音频数据包的方法(在后台线程中,对于视频文件的图像流,等效工作正常)：
//------------------------------------------------------------------------------
int decodeAudioPacket( AVPacket& p_packet, AVCodecContext* p_audioCodecContext, AVFrame* p_frame,
FFmpegVideoPlayer* p_player, VideoInfo& p_videoInfo)
{
// Decode audio frame
int got_frame = 0;
int decoded = avcodec_decode_audio4(p_audioCodecContext, p_frame, &got_frame, &p_packet);
if (decoded < 0)
{
p_videoInfo.error = "Error decoding audio frame.";
return decoded;
}
// Frame is complete, store it in audio frame queue
if (got_frame)
{
int bufferSize = av_samples_get_buffer_size(NULL, p_audioCodecContext->channels, p_frame->nb_samples,
p_audioCodecContext->sample_fmt, 0);
int64_t duration = p_frame->pkt_duration;
int64_t dts = p_frame->pkt_dts;
if (staticOgreLog)
{
staticOgreLog->logMessage("Audio frame bufferSize / duration / dts: "
+ boost::lexical_cast<std::string>(bufferSize) + " / "
+ boost::lexical_cast<std::string>(duration) + " / "
+ boost::lexical_cast<std::string>(dts), Ogre::LML_NORMAL);
}
// Create the audio frame
AudioFrame* frame = new AudioFrame();
frame->dataSize = bufferSize;
frame->data = new uint8_t[bufferSize];
if (p_frame->channels == 2)
{
memcpy(frame->data, p_frame->data[0], bufferSize >> 1);
memcpy(frame->data + (bufferSize >> 1), p_frame->data[1], bufferSize >> 1);
}
else
{
memcpy(frame->data, p_frame->data, bufferSize);
}
double timeBase = ((double)p_audioCodecContext->time_base.num) / (double)p_audioCodecContext->time_base.den;
frame->lifeTime = duration * timeBase;
p_player->addAudioFrame(frame);
}
return decoded;
}
所以,正如你所看到的,我解码了帧,将它memcpy到我自己的struct,AudioFrame.现在,当播放声音时,我使用这样的音频帧：
int numBuffers = 4;
ALuint buffers[4];
alGenBuffers(numBuffers, buffers);
ALenum success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error on alGenBuffers : " + Ogre::StringConverter::toString(success) + alGetString(success));
return;
}
// Fill a number of data buffers with audio from the stream
std::vector<AudioFrame*> audioBuffers;
std::vector<unsigned int> audioBufferSizes;
unsigned int numReturned = FFMPEG_PLAYER->getDecodedAudioFrames(numBuffers, audioBuffers, audioBufferSizes);
// Assign the data buffers to the OpenAL buffers
for (unsigned int i = 0; i < numReturned; ++i)
{
alBufferData(buffers[i], _streamingFormat, audioBuffers[i]->data, audioBufferSizes[i], _streamingFrequency);
success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error on alBufferData : " + Ogre::StringConverter::toString(success) + alGetString(success)
+ " size: " + Ogre::StringConverter::toString(audioBufferSizes[i]));
return;
}
}
// Queue the buffers into OpenAL
alSourceQueueBuffers(_source, numReturned, buffers);
success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error queuing streaming buffers: " + Ogre::StringConverter::toString(success) + alGetString(success));
return;
}
}
alSourcePlay(_source);
我给OpenAL的格式和频率是AL_FORMAT_STEREO_FLOAT32(它是立体声声音流,我确实初始化了FLOAT32扩展)和48000(这是音频流的AVCodecContext的采样率).
在播放过程中,我会执行以下操作来重新填充OpenAL的缓冲区：
ALint numBuffersProcessed;
// Check if OpenAL is done with any of the queued buffers
alGetSourcei(_source, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
if(numBuffersProcessed <= 0)
return;
// Fill a number of data buffers with audio from the stream
std::vector<AudiFrame*> audioBuffers;
std::vector<unsigned int> audioBufferSizes;
unsigned int numFilled = FFMPEG_PLAYER->getDecodedAudioFrames(numBuffersProcessed, audioBuffers, audioBufferSizes);
// Assign the data buffers to the OpenAL buffers
ALuint buffer;
for (unsigned int i = 0; i < numFilled; ++i)
{
// Pop the oldest queued buffer from the source,
// fill it with the new data, then re-queue it
alSourceUnqueueBuffers(_source, 1, &buffer);
ALenum success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error Unqueuing streaming buffers: " + Ogre::StringConverter::toString(success));
return;
}
alBufferData(buffer, _streamingFormat, audioBuffers[i]->data, audioBufferSizes[i], _streamingFrequency);
success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error on re- alBufferData: " + Ogre::StringConverter::toString(success));
return;
}
alSourceQueueBuffers(_source, 1, &buffer);
success = alGetError();
if(success != AL_NO_ERROR)
{
CONSOLE_LOG("Error re-queuing streaming buffers: " + Ogre::StringConverter::toString(success) + " "
+ alGetString(success));
return;
}
}
// Make sure the source is still playing,
// and restart it if needed.
ALint playStatus;
alGetSourcei(_source, AL_SOURCE_STATE, &playStatus);
if(playStatus != AL_PLAYING)
alSourcePlay(_source);
如你所见,我做了很多错误检查.但我不会从OpenAL或FFmpeg那里得到任何错误.
编辑：我听到的有点类似于视频中的实际音频,但非常高音调和口吃很多.此外,它似乎是在电视噪音的顶部播放.很奇怪.另外,它的播放速度比正确的音频慢得多.
编辑：2使用AL_FORMAT_STEREO_FLOAT32后,声音以正确的速度播放,但仍然是非常高的音调和口吃(虽然比以前少).
视频本身没有被破坏,它可以在任何播放器上播放. OpenAL也可以在同一个应用程序中播放* .way文件,所以它也可以正常工作.
任何想法在这里可能是错误的或如何正确地做到这一点？
我唯一的猜测是,不知何故,FFmpeg的解码函数不会产生OpenGL可以读取的数据.但这就是FFmpeg解码示例,所以我不知道缺少什么.据我了解,decode_audio4函数将帧解码为原始数据. OpenAL应该能够使用RAW数据(或者更确切地说,不能与其他任何数据一起使用).
最佳答案 所以,我终于想出了如何做到这一点.哎呀,真是一团糟.这是来自libav-users邮件列表上的用户的
hint,它让我走上了正确的道路.
这是我的错误：
>在alBufferData函数中使用错误的格式.我使用了AL_FORMAT_STEREO16(因为这是OpenAL使用的每个流媒体示例).我应该使用AL_FORMAT_STEREO_FLOAT32,因为视频I流是Ogg而vorbis存储在浮点中.使用swr_convert从AV_SAMPLE_FMT_FLTP转换为AV_SAMPLE_FMT_S16只会崩溃.不知道为什么.
>不使用swr_convert将解码的音频帧转换为目标格式.在我尝试使用swr_convert从FLTP转换为S16之后,它会在没有给出理由的情况下崩溃,我认为它被破坏了.但在弄清楚我的第一个错误之后,我再次尝试,从FLTP转换为FLT(非平面)然后它工作了！因此OpenAL使用交错格式,而不是平面格式.很高兴知道.
所以这里有一个decodeAudioPacket函数,它适用于Ogg视频,vorbis音频流：
int decodeAudioPacket( AVPacket& p_packet, AVCodecContext* p_audioCodecContext, AVFrame* p_frame,
SwrContext* p_swrContext, uint8_t** p_destBuffer, int p_destLinesize,
FFmpegVideoPlayer* p_player, VideoInfo& p_videoInfo)
{
// Decode audio frame
int got_frame = 0;
int decoded = avcodec_decode_audio4(p_audioCodecContext, p_frame, &got_frame, &p_packet);
if (decoded < 0)
{
p_videoInfo.error = "Error decoding audio frame.";
return decoded;
}
if(decoded <= p_packet.size)
{
/* Move the unread data to the front and clear the end bits */
int remaining = p_packet.size - decoded;
memmove(p_packet.data, &p_packet.data[decoded], remaining);
av_shrink_packet(&p_packet, remaining);
}
// Frame is complete, store it in audio frame queue
if (got_frame)
{
int outputSamples = swr_convert(p_swrContext,
p_destBuffer, p_destLinesize,
(const uint8_t**)p_frame->extended_data, p_frame->nb_samples);
int bufferSize = av_get_bytes_per_sample(AV_SAMPLE_FMT_FLT) * p_videoInfo.audioNumChannels
* outputSamples;
int64_t duration = p_frame->pkt_duration;
int64_t dts = p_frame->pkt_dts;
if (staticOgreLog)
{
staticOgreLog->logMessage("Audio frame bufferSize / duration / dts: "
+ boost::lexical_cast<std::string>(bufferSize) + " / "
+ boost::lexical_cast<std::string>(duration) + " / "
+ boost::lexical_cast<std::string>(dts), Ogre::LML_NORMAL);
}
// Create the audio frame
AudioFrame* frame = new AudioFrame();
frame->dataSize = bufferSize;
frame->data = new uint8_t[bufferSize];
memcpy(frame->data, p_destBuffer[0], bufferSize);
double timeBase = ((double)p_audioCodecContext->time_base.num) / (double)p_audioCodecContext->time_base.den;
frame->lifeTime = duration * timeBase;
p_player->addAudioFrame(frame);
}
return decoded;
}
这是我如何初始化上下文和目标缓冲区：
// Initialize SWR context
SwrContext* swrContext = swr_alloc_set_opts(NULL,
audioCodecContext->channel_layout, AV_SAMPLE_FMT_FLT, audioCodecContext->sample_rate,
audioCodecContext->channel_layout, audioCodecContext->sample_fmt, audioCodecContext->sample_rate,
0, NULL);
int result = swr_init(swrContext);
// Create destination sample buffer
uint8_t** destBuffer = NULL;
int destBufferLinesize;
av_samples_alloc_array_and_samples( &destBuffer,
&destBufferLinesize,
videoInfo.audioNumChannels,
2048,
AV_SAMPLE_FMT_FLT,
0);
