Dave Rice dericed.com> writes:
> Not sure if ffmpeg supports any lossy codec in
> WAV but the specification does allow it.
Among other lossy codecs, FFmpeg supports muxing
adpcm_ima_wav and at least a handful other adpcm
codecs including G726 and G723, it supports muxing
mp2, mp3, aac, wma1,
On Fri, Jan 23, 2015 at 09:54:04 -0500, Dave Rice wrote:
> Just to be picky, the WAV container can contain compressed audio data
> so if pcm is not needed, one could control a lossy bitrate if using a
> lossy codec that WAV supports. Not sure if ffmpeg supports any lossy
> codec in WAV but the spe
> On Jan 22, 2015, at 12:52 PM, Moritz Barsnick wrote:
>
> On Thu, Jan 22, 2015 at 10:44:01 -0700, jd1008 wrote:
>> I tried with the params: -ac 2 -ar 44.1k -ab 1600k but to no avail.
> [...]
>> Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo,
>> s16, 1411 kb/s
>
On 01/22/2015 10:52 AM, Moritz Barsnick wrote:
On Thu, Jan 22, 2015 at 10:44:01 -0700, jd1008 wrote:
I tried with the params: -ac 2 -ar 44.1k -ab 1600k but to no avail.
[...]
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo,
s16, 1411 kb/s
I think you don't
On Thu, Jan 22, 2015 at 10:44:01 -0700, jd1008 wrote:
> I tried with the params: -ac 2 -ar 44.1k -ab 1600k but to no avail.
[...]
> Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo,
> s16, 1411 kb/s
I think you don't understand: A 16bit stereo WAV file with 44.1 kH
On 01/22/2015 10:23 AM, Moritz Barsnick wrote:
On Thu, Jan 22, 2015 at 10:08:35 -0700, jd1008 wrote:
it does the conversion, but does not honour the bitrate param.
How do I force the bitrate param for the ourput?
You didn't show us the output of your ffmpeg command.
Anyway, your approach is w
On Thu, Jan 22, 2015 at 10:08:35 -0700, jd1008 wrote:
> it does the conversion, but does not honour the bitrate param.
> How do I force the bitrate param for the ourput?
You didn't show us the output of your ffmpeg command.
Anyway, your approach is wrong. The bitrate of a WAV file is defined by
i
Hi, I tried to transcode a few mp3 files to wav using the command
for f in *.mp3; do
ffmpeg -i "$f" -ab 1600k -y ${f%mp3}wav
done
it does the conversion, but does not honour the bitrate param.
How do I force the bitrate param for the ourput?
___
ffmpeg
