为ffmpeg rtspstream播放设置audio队列
我正在使用ffmpeg为iOSdevise一个rtspstream（AAC格式）客户端。 现在我只能说我的应用程序是可行的，但stream媒体的声音非常嘈杂，甚至有点扭曲，远远低于当它由vlc或mplayer播放时。
该stream由av_read_frame（）读取，由avcodec_decode_audio3（）解码。 然后，我只是将解码的原始audio发送到audio队列。
当用我的应用程序解码本地aac文件时，声音看起来并不那么嘈杂。 我知道初始编码会显着影响结果。 但至less我应该尝试让它听起来像其他stream媒体客户端…
我的实现/修改中的许多部分实际上来自尝试和错误。 我相信我在设置audio队列方面做了一些错误，并且在填充audio缓冲区时使用了callback函数。
任何提示，build议或帮助，不胜感激。
// – 由av_dump_format（）转储的testing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 // – audio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; } }
// – 填充audio缓冲区的callback函数 –
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; }
声音失真的最可能的原因是简单的丢包，RTSP可能容易受到影响，尤其是在无线连接上。
我build议你考虑configurationffmpeg，以便在可能的时候使用基于TCP的连接，而不是默认的RTP / UDP。
如果您要在audio队列callback中下载或解码audio，则NUM_BUFFERS和/或bufferSize可能需要大得多才能覆盖较差的networking延迟和抖动。 或者您可以预先解码audiocallback之外的audio，并在callback之前排队足够的数据以处理下载和解码时间和差异。