Opus wrapper: Use const for inputs and uint8[] for byte streams
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
parent
1bada48401
commit
4bd2db9a55
@ -42,8 +42,11 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
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* Return value : >0 - Length (in bytes) of coded data
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* -1 - Error
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*/
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int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
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int16_t length_encoded_buffer, uint8_t* encoded);
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int16_t WebRtcOpus_Encode(OpusEncInst* inst,
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const int16_t* audio_in,
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int16_t samples,
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int16_t length_encoded_buffer,
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uint8_t* encoded);
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/****************************************************************************
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* WebRtcOpus_SetBitRate(...)
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@ -190,10 +193,10 @@ int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst);
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int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int16_t* decoded,
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int16_t* audio_type);
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int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
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int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int16_t* decoded,
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int16_t* audio_type);
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int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
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int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int16_t* decoded,
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int16_t* audio_type);
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@ -63,17 +63,21 @@ int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
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}
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}
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int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
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int16_t length_encoded_buffer, uint8_t* encoded) {
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opus_int16* audio = (opus_int16*) audio_in;
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unsigned char* coded = encoded;
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int16_t WebRtcOpus_Encode(OpusEncInst* inst,
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const int16_t* audio_in,
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int16_t samples,
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int16_t length_encoded_buffer,
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uint8_t* encoded) {
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int res;
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if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
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return -1;
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}
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res = opus_encode(inst->encoder, audio, samples, coded,
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res = opus_encode(inst->encoder,
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(const opus_int16*)audio_in,
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samples,
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encoded,
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length_encoded_buffer);
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if (res > 0) {
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@ -222,13 +226,11 @@ int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst) {
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/* |frame_size| is set to maximum Opus frame size in the normal case, and
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* is set to the number of samples needed for PLC in case of losses.
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* It is up to the caller to make sure the value is correct. */
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static int DecodeNative(OpusDecoder* inst, const int16_t* encoded,
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static int DecodeNative(OpusDecoder* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int frame_size,
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int16_t* decoded, int16_t* audio_type) {
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unsigned char* coded = (unsigned char*) encoded;
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opus_int16* audio = (opus_int16*) decoded;
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int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 0);
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int res = opus_decode(
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inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 0);
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/* TODO(tlegrand): set to DTX for zero-length packets? */
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*audio_type = 0;
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@ -239,13 +241,11 @@ static int DecodeNative(OpusDecoder* inst, const int16_t* encoded,
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return -1;
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}
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static int DecodeFec(OpusDecoder* inst, const int16_t* encoded,
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static int DecodeFec(OpusDecoder* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int frame_size,
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int16_t* decoded, int16_t* audio_type) {
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unsigned char* coded = (unsigned char*) encoded;
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opus_int16* audio = (opus_int16*) decoded;
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int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 1);
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int res = opus_decode(
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inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 1);
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/* TODO(tlegrand): set to DTX for zero-length packets? */
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*audio_type = 0;
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@ -259,12 +259,12 @@ static int DecodeFec(OpusDecoder* inst, const int16_t* encoded,
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int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int16_t* decoded,
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int16_t* audio_type) {
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int16_t* coded = (int16_t*)encoded;
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int decoded_samples;
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decoded_samples = DecodeNative(inst->decoder_left, coded, encoded_bytes,
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kWebRtcOpusMaxFrameSizePerChannel,
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decoded, audio_type);
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int decoded_samples = DecodeNative(inst->decoder_left,
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encoded,
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encoded_bytes,
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kWebRtcOpusMaxFrameSizePerChannel,
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decoded,
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audio_type);
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if (decoded_samples < 0) {
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return -1;
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}
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@ -275,7 +275,7 @@ int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded,
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return decoded_samples;
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}
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int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
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int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int16_t* decoded,
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int16_t* audio_type) {
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int decoded_samples;
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@ -310,7 +310,7 @@ int16_t WebRtcOpus_Decode(OpusDecInst* inst, const int16_t* encoded,
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return decoded_samples;
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}
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int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const int16_t* encoded,
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int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int16_t* decoded,
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int16_t* audio_type) {
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int decoded_samples;
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@ -439,7 +439,6 @@ int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded,
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int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
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int16_t encoded_bytes, int16_t* decoded,
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int16_t* audio_type) {
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int16_t* coded = (int16_t*)encoded;
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int decoded_samples;
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int fec_samples;
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@ -449,7 +448,7 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
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fec_samples = opus_packet_get_samples_per_frame(encoded, 48000);
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decoded_samples = DecodeFec(inst->decoder_left, coded, encoded_bytes,
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decoded_samples = DecodeFec(inst->decoder_left, encoded, encoded_bytes,
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fec_samples, decoded, audio_type);
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if (decoded_samples < 0) {
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return -1;
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@ -131,7 +131,6 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) {
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int16_t audio_type;
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int16_t output_data_decode_new[kOpusMaxFrameSamples];
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int16_t output_data_decode[kOpusMaxFrameSamples];
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int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
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encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_,
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kOpus20msFrameSamples, kMaxBytes,
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bitstream_);
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@ -140,7 +139,7 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) {
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encoded_bytes, output_data_decode_new,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_Decode(opus_mono_decoder_, coded,
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WebRtcOpus_Decode(opus_mono_decoder_, bitstream_,
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encoded_bytes, output_data_decode,
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&audio_type));
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@ -175,7 +174,6 @@ TEST_F(OpusTest, OpusEncodeDecodeStereo) {
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int16_t output_data_decode_new[kOpusMaxFrameSamples];
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int16_t output_data_decode[kOpusMaxFrameSamples];
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int16_t output_data_decode_slave[kOpusMaxFrameSamples];
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int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
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encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
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kOpus20msFrameSamples, kMaxBytes,
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bitstream_);
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@ -184,11 +182,11 @@ TEST_F(OpusTest, OpusEncodeDecodeStereo) {
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encoded_bytes, output_data_decode_new,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_Decode(opus_stereo_decoder_, coded,
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WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
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encoded_bytes, output_data_decode,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
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WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_,
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encoded_bytes, output_data_decode_slave,
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&audio_type));
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@ -259,7 +257,6 @@ TEST_F(OpusTest, OpusDecodeInit) {
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int16_t output_data_decode_new[kOpusMaxFrameSamples];
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int16_t output_data_decode[kOpusMaxFrameSamples];
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int16_t output_data_decode_slave[kOpusMaxFrameSamples];
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int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
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encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
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kOpus20msFrameSamples, kMaxBytes,
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bitstream_);
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@ -268,11 +265,11 @@ TEST_F(OpusTest, OpusDecodeInit) {
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encoded_bytes, output_data_decode_new,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_Decode(opus_stereo_decoder_, coded,
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WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
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encoded_bytes, output_data_decode,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
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WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_,
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encoded_bytes, output_data_decode_slave,
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&audio_type));
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@ -293,11 +290,11 @@ TEST_F(OpusTest, OpusDecodeInit) {
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encoded_bytes, output_data_decode_new,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_Decode(opus_stereo_decoder_, coded,
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WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
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encoded_bytes, output_data_decode,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
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WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_,
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encoded_bytes, output_data_decode_slave,
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&audio_type));
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@ -399,7 +396,6 @@ TEST_F(OpusTest, OpusDecodePlcMono) {
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int16_t audio_type;
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int16_t output_data_decode_new[kOpusMaxFrameSamples];
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int16_t output_data_decode[kOpusMaxFrameSamples];
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int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
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encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_,
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kOpus20msFrameSamples, kMaxBytes,
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bitstream_);
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@ -408,7 +404,7 @@ TEST_F(OpusTest, OpusDecodePlcMono) {
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encoded_bytes, output_data_decode_new,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_Decode(opus_mono_decoder_, coded,
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WebRtcOpus_Decode(opus_mono_decoder_, bitstream_,
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encoded_bytes, output_data_decode,
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&audio_type));
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@ -451,7 +447,6 @@ TEST_F(OpusTest, OpusDecodePlcStereo) {
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int16_t output_data_decode_new[kOpusMaxFrameSamples];
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int16_t output_data_decode[kOpusMaxFrameSamples];
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int16_t output_data_decode_slave[kOpusMaxFrameSamples];
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int16_t* coded = reinterpret_cast<int16_t*>(bitstream_);
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encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_,
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kOpus20msFrameSamples, kMaxBytes,
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bitstream_);
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@ -460,11 +455,11 @@ TEST_F(OpusTest, OpusDecodePlcStereo) {
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encoded_bytes, output_data_decode_new,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_Decode(opus_stereo_decoder_, coded,
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WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_,
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encoded_bytes, output_data_decode,
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&audio_type));
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EXPECT_EQ(kOpus20msFrameSamples,
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WebRtcOpus_DecodeSlave(opus_stereo_decoder_, coded,
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WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_,
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encoded_bytes,
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output_data_decode_slave,
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&audio_type));
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