Re: Changing sample rate, Web Audio API, getUserMedia -- forward from [chromium-discuss]

I can confirm that on my MacBook.

But interestingly, 'Audio MIDI Setup' app in Mac OS X has 2 sample rate
settings, one for input and one for output. You are right changing the
output sample rate will have an effect on recorded format (also crashes
Chrome when the microphone is active). But I can't help feel a bit weird
about having to use the 'output format' to record 'input'. It's just not
good practice. In Flash, we can specify the recording format.

On Fri, Jan 11, 2013 at 8:46 AM, Chris Wilson <cwilso@google.com> wrote:

> It's not really that only the OfflineAudioContext does transcoding; the
> issue is that you want to control your output rate/format, which can only
> be done by using an OfflineAudioContext (or setting the hardware settings
> on your audio output device).  If you could set your output format for
> audio output to 16kHz, you would see that all the audio output (from
> recorderJS, for example) would be in that format.
>
> -Chris
>
>
> On Thu, Jan 10, 2013 at 4:16 PM, He Shiming <heshiming@gmail.com> wrote:
>
>> I see. So OfflineAudioContext can do transcoding. Hope it will be exposed
>> soon. Thanks again Chris.
>>
>>
>> On Fri, Jan 11, 2013 at 2:39 AM, Chris Wilson <cwilso@google.com> wrote:
>>
>>> The more problematic portion of this, He, is that I suspect your input
>>> device is giving a higher bitrate than you want anyway, so there's no
>>> getting around that you'll need to transcode your input stream (unless your
>>> input hardware really is giving a 16kHz mono stream) - it's just whether
>>> there would be built-in ways to do that, and right now, there aren't.  In
>>> our current design, the input is promoted to the output's hardware rate -
>>> it's not locked to 44kHz/stereo/16bit, that just happens to be what your
>>> output hardware is.
>>>
>>> Reducing from stereo to mono is easy, of course.  As I mentioned before,
>>> dropping from 44kHz to 22kHz is also easy; but if you really need to encode
>>> at specifically 16kHz, and don't want to write the interpolation code, you
>>> can always record the buffer in your normal AudioContext, then create an
>>> OfflineAudioContext at 16kHz mono 16bit, call offlineContext.createBuffer(
>>> 1, length, sampleRate ); and copy the data from your recorded buffer, then
>>> play in the offline context and encode the resulting buffer you're handed
>>> in oncomplete.
>>>
>>>
>>> On Thu, Jan 10, 2013 at 6:44 AM, He Shiming <heshiming@gmail.com> wrote:
>>>
>>>> Thanks Rob.
>>>>
>>>> But the so called 'StreamRecorder' is a bit confusing. A google search
>>>> came up nothing related. Though the sample code here:
>>>> https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/webrtc-integration.htmldid mention the '.record' method on LocalMediaStream interface, the actual
>>>> draft: http://dev.w3.org/2011/webrtc/editor/getusermedia.htmlmentioned 'Apr 23 2012 - Remove MediaStreamRecorder' with no further
>>>> explanation.
>>>>
>>>>
>>>> On Thu, Jan 10, 2013 at 6:47 PM, Robert O'Callahan <
>>>> robert@ocallahan.org> wrote:
>>>>
>>>>> The StreamRecorder API proposal in public-media-capture is a better
>>>>> fit for this use-case. It will let developers record directly to WAV with
>>>>> minimal JS code, at whatever the native sample rate of the stream is.
>>>>>
>>>>> Rob
>>>>> --
>>>>> Jesus called them together and said, “You know that the rulers of the
>>>>> Gentiles lord it over them, and their high officials exercise authority
>>>>> over them. Not so with you. Instead, whoever wants to become great
>>>>> among you must be your servant, and whoever wants to be first must be
>>>>> your slave — just as the Son of Man did not come to be served, but to
>>>>> serve, and to give his life as a ransom for many.” [Matthew 20:25-28]
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Best regards,
>>>> He Shiming
>>>>
>>>
>>>
>>
>>
>> --
>> Best regards,
>> He Shiming
>>
>
>


-- 
Best regards,
He Shiming