wav 的 AudioStreamBasicDescription 设置值
AudioStreamBasicDescription setting values for wav
我正在尝试在 iOS 上播放一个简单的 PCM 文件,但无法绕过 AudioStreamBasicDescription
,而且 link 没有提供足够的信息。
我从终端获得这些值
afinfo BlameItOnTheNight.wav
File: BlameItOnTheNight.wav
File type ID: WAVE
Num Tracks: 1
----
Data format: 2 ch, 44100 Hz, 'lpcm' (0x0000000C) 16-bit little-endian signed integer
no channel layout.
estimated duration: 9.938141 sec
audio bytes: 1753088
audio packets: 438272
bit rate: 1411200 bits per second
packet size upper bound: 4
maximum packet size: 4
audio data file offset: 44
optimized
source bit depth: I16
----
然后我在代码中选择值
- (void)setupAudioFormat:(AudioStreamBasicDescription*)format
{
format->mSampleRate = 44100.0;
format->mFormatID = kAudioFormatLinearPCM;
format->mFramesPerPacket = 1;
format->mChannelsPerFrame = 2;
format->mBytesPerFrame = format->mChannelsPerFrame * sizeof(Float32);
format->mBytesPerPacket = format->mFramesPerPacket * format->mBytesPerFrame;
format->mBitsPerChannel = sizeof(Float32) * 8;
format->mReserved = 0;
format->mFormatFlags = kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagsNativeEndian |
kAudioFormatFlagIsPacked;
}
音频播放速度非常快。
根据实际音频文件计算此值的正确方法是什么?
当我更改值时出现以下错误。
error for object 0x7fba72c50db8: incorrect checksum for freed object - object was probably modified after being freed.
*** set a breakpoint in malloc_error_break to debug
然后我终于发现我的 AudioStreamBasicDescription
bitsperchannel
值不正确,缓冲区大小也不够。
所以首先我将值更改为
- (void)setupAudioFormat:(AudioStreamBasicDescription*)format
{
format->mSampleRate = 44100.0;
format->mFormatID = kAudioFormatLinearPCM;
format->mFramesPerPacket = 1; //For uncompressed audio, the value is 1. For variable bit-rate formats, the value is a larger fixed number, such as 1024 for AAC
format->mChannelsPerFrame = 2;
format->mBytesPerFrame = format->mChannelsPerFrame * 2;
format->mBytesPerPacket = format->mFramesPerPacket * format->mBytesPerFrame;
format->mBitsPerChannel = 16;
format->mReserved = 0;
format->mFormatFlags = kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagsNativeEndian |
kLinearPCMFormatFlagIsPacked;
}
然后当我分配缓冲区时,我增加了大小
// Allocate and prime playback buffers
playState.playing = true;
for (int i = 0; i < NUM_BUFFERS && playState.playing; i++)
{
AudioQueueAllocateBuffer(playState.queue, 32000, &playState.buffers[i]);
AudioOutputCallback(&playState, playState.queue, playState.buffers[i]);
}
在我的原始代码中它被设置为8000
,现在将其更改为32000 解决了问题。
我正在尝试在 iOS 上播放一个简单的 PCM 文件,但无法绕过 AudioStreamBasicDescription
,而且 link 没有提供足够的信息。
我从终端获得这些值
afinfo BlameItOnTheNight.wav
File: BlameItOnTheNight.wav
File type ID: WAVE
Num Tracks: 1
----
Data format: 2 ch, 44100 Hz, 'lpcm' (0x0000000C) 16-bit little-endian signed integer
no channel layout.
estimated duration: 9.938141 sec
audio bytes: 1753088
audio packets: 438272
bit rate: 1411200 bits per second
packet size upper bound: 4
maximum packet size: 4
audio data file offset: 44
optimized
source bit depth: I16
----
然后我在代码中选择值
- (void)setupAudioFormat:(AudioStreamBasicDescription*)format
{
format->mSampleRate = 44100.0;
format->mFormatID = kAudioFormatLinearPCM;
format->mFramesPerPacket = 1;
format->mChannelsPerFrame = 2;
format->mBytesPerFrame = format->mChannelsPerFrame * sizeof(Float32);
format->mBytesPerPacket = format->mFramesPerPacket * format->mBytesPerFrame;
format->mBitsPerChannel = sizeof(Float32) * 8;
format->mReserved = 0;
format->mFormatFlags = kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagsNativeEndian |
kAudioFormatFlagIsPacked;
}
音频播放速度非常快。
根据实际音频文件计算此值的正确方法是什么?
当我更改值时出现以下错误。
error for object 0x7fba72c50db8: incorrect checksum for freed object - object was probably modified after being freed.
*** set a breakpoint in malloc_error_break to debug
然后我终于发现我的 AudioStreamBasicDescription
bitsperchannel
值不正确,缓冲区大小也不够。
所以首先我将值更改为
- (void)setupAudioFormat:(AudioStreamBasicDescription*)format
{
format->mSampleRate = 44100.0;
format->mFormatID = kAudioFormatLinearPCM;
format->mFramesPerPacket = 1; //For uncompressed audio, the value is 1. For variable bit-rate formats, the value is a larger fixed number, such as 1024 for AAC
format->mChannelsPerFrame = 2;
format->mBytesPerFrame = format->mChannelsPerFrame * 2;
format->mBytesPerPacket = format->mFramesPerPacket * format->mBytesPerFrame;
format->mBitsPerChannel = 16;
format->mReserved = 0;
format->mFormatFlags = kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagsNativeEndian |
kLinearPCMFormatFlagIsPacked;
}
然后当我分配缓冲区时,我增加了大小
// Allocate and prime playback buffers
playState.playing = true;
for (int i = 0; i < NUM_BUFFERS && playState.playing; i++)
{
AudioQueueAllocateBuffer(playState.queue, 32000, &playState.buffers[i]);
AudioOutputCallback(&playState, playState.queue, playState.buffers[i]);
}
在我的原始代码中它被设置为8000
,现在将其更改为32000 解决了问题。