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Opus wrapper: Use const for inputs and uint8[] for byte streams

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About half of the functions already followed the desired pattern; this
patch fixes the other half.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7409 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2014-10-09 11:21:10 +00:00
parent 1bada48401
commit 4bd2db9a55
3 changed files with 41 additions and 44 deletions
Download Patch File Download Diff File Expand all files Collapse all files

49
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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@ -63,17 +63,21 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
}
}
int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
int16_t length_encoded_buffer, uint8_t* encoded) {
opus_int16* audio = (opus_int16*) audio_in;
unsigned char* coded = encoded;
int16_t WebRtcOpus_Encode(OpusEncInst* inst,
const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded) {
int res;
if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
return -1;
}
res = opus_encode(inst->encoder, audio, samples, coded,
res = opus_encode(inst->encoder,
(const opus_int16*)audio_in,
samples,
encoded,
length_encoded_buffer);
if (res > 0) {
@ -222,13 +226,11 @@ int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst) {
/* |frame_size| is set to maximum Opus frame size in the normal case, and
* is set to the number of samples needed for PLC in case of losses.
* It is up to the caller to make sure the value is correct. */
static int DecodeNative(OpusDecoder* inst, const int16_t* encoded,
static int DecodeNative(OpusDecoder* inst, const uint8_t* encoded,
int16_t encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) {
unsigned char* coded = (unsigned char*) encoded;
opus_int16* audio = (opus_int16*) decoded;
int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 0);
int res = opus_decode(
inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 0);
/* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0;
@ -239,13 +241,11 @@ static int DecodeNative(OpusDecoder* inst, const int16_t* encoded,
return -1;
}
static int DecodeFec(OpusDecoder* inst, const int16_t* encoded,
static int DecodeFec(OpusDecoder* inst, const uint8_t* encoded,
int16_t encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) {
unsigned char* coded = (unsigned char*) encoded;
opus_int16* audio = (opus_int16*) decoded;
int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 1);
int res = opus_decode(
inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 1);
/* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0;
@ -259,12 +259,12 @@ static int DecodeFec(OpusDecoder* inst, const int16_t* encoded,
int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int16_t* coded = (int16_t*)encoded;
int decoded_samples;
decoded_samples = DecodeNative(inst->decoder_left, coded, encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel,
decoded, audio_type);
int decoded_samples = DecodeNative(inst->decoder_left,
encoded,
encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel,
decoded,
audio_type);
if (decoded_samples < 0) {
return -1;
}
@ -275,7 +275,7 @@ int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
return decoded_samples;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int decoded_samples;
@ -310,7 +310,7 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
return decoded_samples;
}
int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int decoded_samples;
@ -439,7 +439,6 @@ int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded,
int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int16_t* coded = (int16_t*)encoded;
int decoded_samples;
int fec_samples;
@ -449,7 +448,7 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
fec_samples = opus_packet_get_samples_per_frame(encoded, 48000);
decoded_samples = DecodeFec(inst->decoder_left, coded, encoded_bytes,
decoded_samples = DecodeFec(inst->decoder_left, encoded, encoded_bytes,
fec_samples, decoded, audio_type);
if (decoded_samples < 0) {
return -1;