shithub: opus

--- a/doc/draft-ietf-codec-oggopus.xml

+++ b/doc/draft-ietf-codec-oggopus.xml

@@ -330,7 +330,7 @@

  gap.

 This also requires four bytes to describe the synthesized packet data (two

  bytes for a CBR code 3 and one byte each for two code 0 packets) but three

- bytes of Ogg lacing overhead are necessary to mark the packet boundaries.

+ bytes of Ogg lacing overhead are needed to mark the packet boundaries.

 At 0.6 kbps, this is still a minimal bitrate impact over a naive, low quality

  solution.

 </t>

@@ -349,7 +349,7 @@

 There is some amount of latency introduced during the decoding process, to

  allow for overlap in the CELT mode, stereo mixing in the SILK mode, and

  resampling.

-The encoder introduces additional latency through its own resampling

+The encoder might have introduced additional latency through its own resampling

  and analysis (though the exact amount is not specified).

 Therefore, the first few samples produced by the decoder do not correspond to

  real input audio, but are instead composed of padding inserted by the encoder

@@ -364,7 +364,7 @@

 A 'pre-skip' field in the ID header (see <xref target="id_header"/>) signals

  the number of samples which SHOULD be skipped (decoded but discarded) at the

  beginning of the stream.

-This amount MAY not be a multiple of 2.5&nbsp;ms, MAY be smaller than a single

+This amount need not be a multiple of 2.5&nbsp;ms, MAY be smaller than a single

  packet, or MAY span the contents of several packets.

 These samples are not valid audio, and SHOULD NOT be played.

 </t>

@@ -804,7 +804,7 @@

 </t>

 <t><spanx style="strong">Channel Mapping</spanx> (8*C bits):

 This contains one octet per output channel, indicating which decoded channel

- to be used for each one.

+ is to be used for each one.

 Let 'index' be the value of this octet for a particular output channel.

 This value MUST either be smaller than (M+N), or be the special value 255.

 If 'index' is less than 2*M, the output MUST be taken from decoding stream