因此,爲了說明我的問題,我將給出一些上下文。如何降低linux上的wav文件的質量和規格
在SDL2可以加載WAV文件,例如從wiki:
SDL_AudioSpec wav_spec;
Uint32 wav_length;
Uint8 *wav_buffer;
/* Load the WAV */
if (SDL_LoadWAV("test.wav", &wav_spec, &wav_buffer, &wav_length) == NULL) {
fprintf(stderr, "Could not open test.wav: %s\n", SDL_GetError());
} else {
/* Do stuff with the WAV data, and then... */
SDL_FreeWAV(wav_buffer);
}
我從SDL_GetError得到的問題是Complex WAVE files not supported
現在我打算WAV文件打開具有以下屬性:
Playing test.wav.
Detected file format: WAV/WAVE (Waveform Audio) (libavformat)
ID_AUDIO_ID=0
[lavf] stream 0: audio (pcm_s24le), -aid 0
Clip info:
encoded_by: Pro Tools
ID_CLIP_INFO_NAME0=encoded_by
ID_CLIP_INFO_VALUE0=Pro Tools
originator_reference:
ID_CLIP_INFO_NAME1=originator_reference
ID_CLIP_INFO_VALUE1=
date: 2016-05-1
ID_CLIP_INFO_NAME2=date
ID_CLIP_INFO_VALUE2=2016-05-1
creation_time: 20:13:34
ID_CLIP_INFO_NAME3=creation_time
ID_CLIP_INFO_VALUE3=20:13:34
time_reference:
ID_CLIP_INFO_NAME4=time_reference
ID_CLIP_INFO_VALUE4=
ID_CLIP_INFO_N=5
Load subtitles in dir/
ID_FILENAME=dir/test.wav
ID_DEMUXER=lavfpref
ID_AUDIO_FORMAT=1
ID_AUDIO_BITRATE=2304000
ID_AUDIO_RATE=48000
ID_AUDIO_NCH=2
ID_START_TIME=0.00
ID_LENGTH=135.53
ID_SEEKABLE=1
ID_CHAPTERS=0
Selected audio codec: Uncompressed PCM [pcm]
AUDIO: 48000 Hz, 2 ch, s24le, 2304.0 kbit/100.00% (ratio: 288000->288000)
ID_AUDIO_BITRATE=2304000
ID_AUDIO_RATE=48000
ID_AUDIO_NCH=2
AO: [pulse] 48000Hz 2ch s16le (2 bytes per sample)
ID_AUDIO_CODEC=pcm
從wiki.libsdl.org/SDL_OpenAudioDevice頁面和隨後的頁面wiki.libsdl.org/SDL_AudioSpec#Remarks我至少可以推測的wav文件:
freq = 48000;
format = AUDIO_F32;
channels = 2;
samples = 4096;
質量應該工作。
我能看到的主要問題是我的wav文件有s16le
格式,而它沒有在SDL_AudioSpec頁面上列出。
這使我相信我需要降低test.wav的質量,因此它在SDL中顯示爲「複雜」。
當我搜索引擎Complex WAVE files not supported
什麼有用的出現,除了它出現在SDL_Mixer庫中,據我所知我不使用。
可以通過ffmepg更改格式以便在SDL2中工作?
編輯:這似乎是SDL2中它投訴的實際代碼。我真的不知道有足夠的瞭解C到挖通了廣大SDL2庫中的所有方法,但我想如果有人注意到的東西剛提示的變量名和等可能會有所幫助:
/* Read the audio data format chunk */
chunk.data = NULL;
do {
if (chunk.data != NULL) {
SDL_free(chunk.data);
chunk.data = NULL;
}
lenread = ReadChunk(src, &chunk);
if (lenread < 0) {
was_error = 1;
goto done;
}
/* 2 Uint32's for chunk header+len, plus the lenread */
headerDiff += lenread + 2 * sizeof(Uint32);
} while ((chunk.magic == FACT) || (chunk.magic == LIST));
/* Decode the audio data format */
format = (WaveFMT *)chunk.data;
if (chunk.magic != FMT) {
SDL_SetError("Complex WAVE files not supported");
was_error = 1;
goto done;
}