我正在使用ffmpeg为iOS创建一个rtsp流媒体（AAC格式）客户端。现在我只能说我的应用程序是可行的，但是流声音非常嘈杂，甚至有点失真，比vlc或mplayer播放时更糟糕。
流由av_read_frame（）读取，由avcodec_decode_audio3（）解码。然后我将解码后的原始音频发送到音频队列。
使用我的应用程序解码本地aac文件时，声音似乎不那么嘈杂。我知道初始编码会显着影响结果。但至少我应该尝试让它听起来像其他流媒体客户端......
我的实现/修改中的许多部分实际上来自try和error。我相信我在设置音频队列方面做错了，还有用于填充音频缓冲区的回调函数。
非常感谢任何提示，建议或帮助。
// - 由av_dump_format（） -
转储的测试材料的信息Metadata:
title : /demo/test.3gp
Duration: 00:00:30.11, start: 0.000000, bitrate: N/A
Stream #0:0: Audio: aac, 32000 Hz, stereo, s16
aac Advanced Audio Coding
// - 音频队列设置程序 -
- (void) startPlayback
{
OSStatus err = 0;
if(playState.playing) return;
playState.started = false;
if(!playState.queue)
{
UInt32 bufferSize;
playState.format.mSampleRate = _av->audio.sample_rate;
playState.format.mFormatID = kAudioFormatLinearPCM;
playState.format.mFormatFlags = kAudioFormatFlagsCanonical;
playState.format.mChannelsPerFrame = _av->audio.channels_per_frame;
playState.format.mBytesPerPacket = sizeof(AudioSampleType) *_av->audio.channels_per_frame;
playState.format.mBytesPerFrame = sizeof(AudioSampleType) *_av->audio.channels_per_frame;
playState.format.mBitsPerChannel = 8 * sizeof(AudioSampleType);
playState.format.mFramesPerPacket = 1;
playState.format.mReserved = 0;
pauseStart = 0;
DeriveBufferSize(playState.format,playState.format.mBytesPerPacket,BUFFER_DURATION,&bufferSize,&numPacketsToRead);
err= AudioQueueNewOutput(&playState.format, aqCallback, &playState, NULL, kCFRunLoopCommonModes, 0, &playState.queue);
if(err != 0)
{
printf("AQHandler.m startPlayback: Error creating new AudioQueue: %d \n", (int)err);
}
for(int i = 0 ; i < NUM_BUFFERS ; i ++)
{
err = AudioQueueAllocateBufferWithPacketDescriptions(playState.queue, bufferSize, numPacketsToRead , &playState.buffers[i]);
if(err != 0)
printf("AQHandler.m startPlayback: Error allocating buffer %d", i);
fillAudioBuffer(&playState,playState.queue, playState.buffers[i]);
}
}
startTime = mu_currentTimeInMicros();
err=AudioQueueStart(playState.queue, NULL);
if(err)
{
char sErr[4];
printf("AQHandler.m startPlayback: Could not start queue %ld %s.", err, FormatError(sErr,err));
playState.playing = NO;
}
else
{
AudioSessionSetActive(true);
playState.playing = YES;
}
}
// - 用于填充音频缓冲区的回调 -
static int ct = 0;
static void fillAudioBuffer(void *info,AudioQueueRef queue, AudioQueueBufferRef buffer)
{
int lengthCopied = INT32_MAX;
int dts= 0;
int isDone = 0;
buffer->mAudioDataByteSize = 0;
buffer->mPacketDescriptionCount = 0;
OSStatus err = 0;
AudioTimeStamp bufferStartTime;
AudioQueueGetCurrentTime(queue, NULL, &bufferStartTime, NULL);
PlayState *ps = (PlayState *)info;
if (!ps->started)
ps->started = true;
while(buffer->mPacketDescriptionCount < numPacketsToRead && lengthCopied > 0)
{
lengthCopied = getNextAudio(_av,
buffer->mAudioDataBytesCapacity-buffer->mAudioDataByteSize,
(uint8_t*)buffer->mAudioData+buffer->mAudioDataByteSize,
&dts,&isDone);
ct+= lengthCopied;
if(lengthCopied < 0 || isDone)
{
printf("nothing to read....\n\n");
PlayState *ps = (PlayState *)info;
ps->finished = true;
ps->started = false;
break;
}
if(aqStartDts < 0) aqStartDts = dts;
if(buffer->mPacketDescriptionCount ==0)
{
bufferStartTime.mFlags = kAudioTimeStampSampleTimeValid;
bufferStartTime.mSampleTime = (Float64)(dts-aqStartDts);//* _av->audio.frame_size;
if (bufferStartTime.mSampleTime <0 )
bufferStartTime.mSampleTime = 0;
printf("AQHandler.m fillAudioBuffer: DTS for %x: %lf time base: %lf StartDTS: %d\n",
(unsigned int)buffer,
bufferStartTime.mSampleTime,
_av->audio.time_base,
aqStartDts);
}
buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mStartOffset = buffer->mAudioDataByteSize;
buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mDataByteSize = lengthCopied;
buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mVariableFramesInPacket = 0;
buffer->mPacketDescriptionCount++;
buffer->mAudioDataByteSize += lengthCopied;
}
int audioBufferCount, audioBufferTotal, videoBufferCount, videoBufferTotal;
bufferCheck(_av,&videoBufferCount, &videoBufferTotal, &audioBufferCount, &audioBufferTotal);
if(buffer->mAudioDataByteSize)
{
err = AudioQueueEnqueueBufferWithParameters(queue, buffer, 0, NULL, 0, 0, 0, NULL, &bufferStartTime, NULL);
if(err)
{
char sErr[10];
printf("AQHandler.m fillAudioBuffer: Could not enqueue buffer 0x%x: %d %s.", buffer, err, FormatError(sErr, err));
}
}
}
int getNextAudio(video_data_t* vInst, int maxlength, uint8_t* buf, int* pts, int* isDone)
{
struct video_context_t *ctx = vInst->context;
int datalength = 0;
while(ctx->audio_ring.lock || (ctx->audio_ring.count <= 0 && ((ctx->play_state & STATE_DIE) != STATE_DIE)))
{
if (ctx->play_state & STATE_EOF) return -1;
usleep(100);
}
*pts = 0;
ctx->audio_ring.lock = kLocked;
if(ctx->audio_ring.count>0 && maxlength > ctx->audio_buffer[ctx->audio_ring.read].size)
{
memcpy(buf, ctx->audio_buffer[ctx->audio_ring.read].data,ctx->audio_buffer[ctx->audio_ring.read].size);
*pts = ctx->audio_buffer[ctx->audio_ring.read].pts;
datalength = ctx->audio_buffer[ctx->audio_ring.read].size;
ctx->audio_ring.read++;
ctx->audio_ring.read %= ABUF_SIZE;
ctx->audio_ring.count--;
}
ctx->audio_ring.lock = kUnlocked;
if((ctx->play_state & STATE_EOF) == STATE_EOF && ctx->audio_ring.count == 0) *isDone = 1;
return datalength;
}
答案 0 :(得分：1)
如果要在音频队列回调中下载或解码音频，NUM_BUFFERS和/或bufferSize可能需要更大才能覆盖更糟糕的网络延迟和抖动。或者您可以在音频回调之外预解码音频，并在回调之前排队足够的数据以处理下载和解码时间和方差。
答案 1 :(得分：1)
声音失真的最可能原因是简单的数据包丢失，RTSP很容易受到影响，特别是在无线连接上。
我建议你考虑配置ffmpeg以尽可能使用基于TCP的连接而不是默认的RTP / UDP。