generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

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

Browse Source 
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

11
webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
View File

@@ -42,8 +42,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
* Return value : >0 - Length (in bytes) of coded data * Return value : >0 - Length (in bytes) of coded data
* -1 - Error * -1 - Error
*/ */
int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples, int16_t WebRtcOpus_Encode(OpusEncInst* inst,
int16_t length_encoded_buffer, uint8_t* encoded); const int16_t* audio_in,
int16_t samples,
int16_t length_encoded_buffer,
uint8_t* encoded);
/**************************************************************************** /****************************************************************************
* WebRtcOpus_SetBitRate(...) * WebRtcOpus_SetBitRate(...)
@@ -190,10 +193,10 @@ int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst);
int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded, int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded, int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type); int16_t* audio_type);
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 encoded_bytes, int16_t* decoded,
int16_t* audio_type); int16_t* audio_type);
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 encoded_bytes, int16_t* decoded,
int16_t* audio_type); int16_t* audio_type);

47
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@@ -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 WebRtcOpus_Encode(OpusEncInst* inst,
int16_t length_encoded_buffer, uint8_t* encoded) { const int16_t* audio_in,
opus_int16* audio = (opus_int16*) audio_in; int16_t samples,
unsigned char* coded = encoded; int16_t length_encoded_buffer,
uint8_t* encoded) {
int res; int res;
if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) { if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
return -1; 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); length_encoded_buffer);
if (res > 0) { 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 /* |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. * 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. */ * 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 encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) { int16_t* decoded, int16_t* audio_type) {
unsigned char* coded = (unsigned char*) encoded; int res = opus_decode(
opus_int16* audio = (opus_int16*) decoded; inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 0);
int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 0);
/* TODO(tlegrand): set to DTX for zero-length packets? */ /* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0; *audio_type = 0;
@@ -239,13 +241,11 @@ static int DecodeNative(OpusDecoder* inst, const int16_t* encoded,
return -1; 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 encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) { int16_t* decoded, int16_t* audio_type) {
unsigned char* coded = (unsigned char*) encoded; int res = opus_decode(
opus_int16* audio = (opus_int16*) decoded; inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 1);
int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 1);
/* TODO(tlegrand): set to DTX for zero-length packets? */ /* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0; *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 WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded, int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) { int16_t* audio_type) {
int16_t* coded = (int16_t*)encoded; int decoded_samples = DecodeNative(inst->decoder_left,
int decoded_samples; encoded,
encoded_bytes,
decoded_samples = DecodeNative(inst->decoder_left, coded, encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel, kWebRtcOpusMaxFrameSizePerChannel,
decoded, audio_type); decoded,
audio_type);
if (decoded_samples < 0) { if (decoded_samples < 0) {
return -1; return -1;
} }
@@ -275,7 +275,7 @@ int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
return decoded_samples; 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 encoded_bytes, int16_t* decoded,
int16_t* audio_type) { int16_t* audio_type) {
int decoded_samples; int decoded_samples;
@@ -310,7 +310,7 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
return decoded_samples; 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 encoded_bytes, int16_t* decoded,
int16_t* audio_type) { int16_t* audio_type) {
int decoded_samples; 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 WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded, int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) { int16_t* audio_type) {
int16_t* coded = (int16_t*)encoded;
int decoded_samples; int decoded_samples;
int fec_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); 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); fec_samples, decoded, audio_type);
if (decoded_samples < 0) { if (decoded_samples < 0) {
return -1; return -1;

25
webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@@ -131,7 +131,6 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) {
int16_t audio_type; int16_t audio_type;
int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes, kOpus20msFrameSamples, kMaxBytes,
bitstream_); bitstream_);
@@ -140,7 +139,7 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) {
encoded_bytes, output_data_decode_new, encoded_bytes, output_data_decode_new,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_mono_decoder_, coded, WebRtcOpus_Decode(opus_mono_decoder_, bitstream_,
encoded_bytes, output_data_decode, encoded_bytes, output_data_decode,
&audio_type)); &audio_type));
@@ -175,7 +174,6 @@ TEST_F(OpusTest, OpusEncodeDecodeStereo) {
int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples];
int16_t output_data_decode_slave[kOpusMaxFrameSamples]; int16_t output_data_decode_slave[kOpusMaxFrameSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes, kOpus20msFrameSamples, kMaxBytes,
bitstream_); bitstream_);
@@ -184,11 +182,11 @@ TEST_F(OpusTest, OpusEncodeDecodeStereo) {
encoded_bytes, output_data_decode_new, encoded_bytes, output_data_decode_new,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, coded, WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode, encoded_bytes, output_data_decode,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode_slave, encoded_bytes, output_data_decode_slave,
&audio_type)); &audio_type));
@@ -259,7 +257,6 @@ TEST_F(OpusTest, OpusDecodeInit) {
int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples];
int16_t output_data_decode_slave[kOpusMaxFrameSamples]; int16_t output_data_decode_slave[kOpusMaxFrameSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes, kOpus20msFrameSamples, kMaxBytes,
bitstream_); bitstream_);
@@ -268,11 +265,11 @@ TEST_F(OpusTest, OpusDecodeInit) {
encoded_bytes, output_data_decode_new, encoded_bytes, output_data_decode_new,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, coded, WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode, encoded_bytes, output_data_decode,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode_slave, encoded_bytes, output_data_decode_slave,
&audio_type)); &audio_type));
@@ -293,11 +290,11 @@ TEST_F(OpusTest, OpusDecodeInit) {
encoded_bytes, output_data_decode_new, encoded_bytes, output_data_decode_new,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, coded, WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode, encoded_bytes, output_data_decode,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode_slave, encoded_bytes, output_data_decode_slave,
&audio_type)); &audio_type));
@@ -399,7 +396,6 @@ TEST_F(OpusTest, OpusDecodePlcMono) {
int16_t audio_type; int16_t audio_type;
int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes, kOpus20msFrameSamples, kMaxBytes,
bitstream_); bitstream_);
@@ -408,7 +404,7 @@ TEST_F(OpusTest, OpusDecodePlcMono) {
encoded_bytes, output_data_decode_new, encoded_bytes, output_data_decode_new,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_mono_decoder_, coded, WebRtcOpus_Decode(opus_mono_decoder_, bitstream_,
encoded_bytes, output_data_decode, encoded_bytes, output_data_decode,
&audio_type)); &audio_type));
@@ -451,7 +447,6 @@ TEST_F(OpusTest, OpusDecodePlcStereo) {
int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode_new[kOpusMaxFrameSamples];
int16_t output_data_decode[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples];
int16_t output_data_decode_slave[kOpusMaxFrameSamples]; int16_t output_data_decode_slave[kOpusMaxFrameSamples];
int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
kOpus20msFrameSamples, kMaxBytes, kOpus20msFrameSamples, kMaxBytes,
bitstream_); bitstream_);
@@ -460,11 +455,11 @@ TEST_F(OpusTest, OpusDecodePlcStereo) {
encoded_bytes, output_data_decode_new, encoded_bytes, output_data_decode_new,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_Decode(opus_stereo_decoder_, coded, WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
encoded_bytes, output_data_decode, encoded_bytes, output_data_decode,
&audio_type)); &audio_type));
EXPECT_EQ(kOpus20msFrameSamples, EXPECT_EQ(kOpus20msFrameSamples,
WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded, WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_,
encoded_bytes, encoded_bytes,
output_data_decode_slave, output_data_decode_slave,
&audio_type)); &audio_type));