static FILE*
OpenDumpFile(AudioStream* aStream)
{
if (!getenv("MOZ_DUMP_AUDIO"))
return nullptr;
char buf[100];
snprintf_literal(buf, "dumped-audio-%d.wav", gDumpedAudioCount);
FILE* f = fopen(buf, "wb");
if (!f)
return nullptr;
++gDumpedAudioCount;
uint8_t header[] = {
// RIFF header
0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, 0x45,
// fmt chunk. We always write 16-bit samples.
0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0xFF, 0xFF,
0xFF, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x10, 0x00,
// data chunk
0x64, 0x61, 0x74, 0x61, 0xFE, 0xFF, 0xFF, 0x7F
};
static const int CHANNEL_OFFSET = 22;
static const int SAMPLE_RATE_OFFSET = 24;
static const int BLOCK_ALIGN_OFFSET = 32;
SetUint16LE(header + CHANNEL_OFFSET, aStream->GetChannels());
SetUint32LE(header + SAMPLE_RATE_OFFSET, aStream->GetRate());
SetUint16LE(header + BLOCK_ALIGN_OFFSET, aStream->GetChannels()*2);
fwrite(header, sizeof(header), 1, f);
return f;
}
typename EnableIf<IsSame<T, float>::value, void>::Type
WriteDumpFileHelper(T* aInput, size_t aSamples, FILE* aFile) {
AutoTArray<uint8_t, 1024*2> buf;
buf.SetLength(aSamples*2);
uint8_t* output = buf.Elements();
for (uint32_t i = 0; i < aSamples; ++i) {
SetUint16LE(output + i*2, int16_t(aInput[i]*32767.0f));
}
fwrite(output, 2, aSamples, aFile);
fflush(aFile);
}
static FILE*
OpenDumpFile(uint32_t aChannels, uint32_t aRate)
{
/**
* When MOZ_DUMP_AUDIO is set in the environment (to anything),
* we'll drop a series of files in the current working directory named
* dumped-audio-<nnn>.wav, one per AudioStream created, containing
* the audio for the stream including any skips due to underruns.
*/
static Atomic<int> gDumpedAudioCount(0);
if (!getenv("MOZ_DUMP_AUDIO"))
return nullptr;
char buf[100];
snprintf_literal(buf, "dumped-audio-%d.wav", ++gDumpedAudioCount);
FILE* f = fopen(buf, "wb");
if (!f)
return nullptr;
uint8_t header[] = {
// RIFF header
0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, 0x45,
// fmt chunk. We always write 16-bit samples.
0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0xFF, 0xFF,
0xFF, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x10, 0x00,
// data chunk
0x64, 0x61, 0x74, 0x61, 0xFE, 0xFF, 0xFF, 0x7F
};
static const int CHANNEL_OFFSET = 22;
static const int SAMPLE_RATE_OFFSET = 24;
static const int BLOCK_ALIGN_OFFSET = 32;
SetUint16LE(header + CHANNEL_OFFSET, aChannels);
SetUint32LE(header + SAMPLE_RATE_OFFSET, aRate);
SetUint16LE(header + BLOCK_ALIGN_OFFSET, aChannels * 2);
fwrite(header, sizeof(header), 1, f);
return f;
}
static void
WriteDumpFile(FILE* aDumpFile, AudioStream* aStream, uint32_t aFrames,
void* aBuffer)
{
if (!aDumpFile)
return;
uint32_t samples = aStream->GetOutChannels()*aFrames;
if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
fwrite(aBuffer, 2, samples, aDumpFile);
return;
}
NS_ASSERTION(AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_FLOAT32, "bad format");
nsAutoTArray<uint8_t, 1024*2> buf;
buf.SetLength(samples*2);
float* input = static_cast<float*>(aBuffer);
uint8_t* output = buf.Elements();
for (uint32_t i = 0; i < samples; ++i) {
SetUint16LE(output + i*2, int16_t(input[i]*32767.0f));
}
fwrite(output, 2, samples, aDumpFile);
fflush(aDumpFile);
}
static void SetUint32LE(uint8_t* aDest, uint32_t aValue)
{
SetUint16LE(aDest, aValue & 0xFFFF);
SetUint16LE(aDest + 2, aValue >> 16);
}
static void SetUint32LE(PRUint8* aDest, PRUint32 aValue)
{
SetUint16LE(aDest, aValue & 0xFFFF);
SetUint16LE(aDest + 2, aValue >> 16);
}
