From 33ccdfa1f555e00170e2b98cd0f575eed3e46236 Mon Sep 17 00:00:00 2001 From: "minyue@webrtc.org" Date: Thu, 4 Dec 2014 12:14:12 +0000 Subject: [PATCH] Relanding r7807. r7807 was reverted to be excluded from the cause of a failure. It has been verified and can reland now. BUG= TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../codecs/opus/interface/opus_interface.h | 15 -- .../audio_coding/codecs/opus/opus_fec_test.cc | 6 +- .../audio_coding/codecs/opus/opus_inst.h | 3 +- .../audio_coding/codecs/opus/opus_interface.c | 191 ++---------------- .../codecs/opus/opus_speed_test.cc | 4 +- .../audio_coding/codecs/opus/opus_unittest.cc | 134 +----------- .../audio_coding/main/test/opus_test.cc | 8 +- .../audio_coding/neteq/audio_decoder_impl.cc | 8 +- 8 files changed, 34 insertions(+), 335 deletions(-) diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h index c1348565f..38ff00dbd 100644 --- a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h +++ b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h @@ -166,9 +166,7 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst); * Return value : 0 - Success * -1 - Error */ -int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst); int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst); -int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst); /**************************************************************************** * WebRtcOpus_Decode(...) @@ -190,21 +188,12 @@ int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst); * Return value : >0 - Samples per channel in decoded vector * -1 - Error */ -int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded, - int16_t encoded_bytes, int16_t* decoded, - int16_t* audio_type); int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type); -int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const uint8_t* encoded, - int16_t encoded_bytes, int16_t* decoded, - int16_t* audio_type); /**************************************************************************** * WebRtcOpus_DecodePlc(...) - * TODO(tlegrand): Remove master and slave functions when NetEq4 is in place. - * WebRtcOpus_DecodePlcMaster(...) - * WebRtcOpus_DecodePlcSlave(...) * * This function processes PLC for opus frame(s). * Input: @@ -219,10 +208,6 @@ int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const uint8_t* encoded, */ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames); -int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded, - int16_t number_of_lost_frames); -int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded, - int16_t number_of_lost_frames); /**************************************************************************** * WebRtcOpus_DecodeFec(...) diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc index 57d834354..35f39729f 100644 --- a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc +++ b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc @@ -156,10 +156,8 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) { if (!lost_current) { // Decode current frame. - value_2 = WebRtcOpus_DecodeNew(opus_decoder_, &bit_stream_[0], - encoded_bytes_, - &out_data_[value_1 * channels_], - &audio_type); + value_2 = WebRtcOpus_Decode(opus_decoder_, &bit_stream_[0], encoded_bytes_, + &out_data_[value_1 * channels_], &audio_type); EXPECT_EQ(block_length_sample_, value_2); } } diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_inst.h b/webrtc/modules/audio_coding/codecs/opus/opus_inst.h index 50caf8373..4597ab8eb 100644 --- a/webrtc/modules/audio_coding/codecs/opus/opus_inst.h +++ b/webrtc/modules/audio_coding/codecs/opus/opus_inst.h @@ -18,8 +18,7 @@ struct WebRtcOpusEncInst { }; struct WebRtcOpusDecInst { - OpusDecoder* decoder_left; - OpusDecoder* decoder_right; + OpusDecoder* decoder; int prev_decoded_samples; int channels; }; diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c index 0c2644ba3..ba7fe13ac 100644 --- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c +++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c @@ -149,8 +149,7 @@ int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) { } int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) { - int error_l; - int error_r; + int error; OpusDecInst* state; if (inst != NULL) { @@ -160,11 +159,9 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) { return -1; } - /* Create new memory for left and right channel, always at 48000 Hz. */ - state->decoder_left = opus_decoder_create(48000, channels, &error_l); - state->decoder_right = opus_decoder_create(48000, channels, &error_r); - if (error_l == OPUS_OK && error_r == OPUS_OK && state->decoder_left != NULL - && state->decoder_right != NULL) { + /* Create new memory, always at 48000 Hz. */ + state->decoder = opus_decoder_create(48000, channels, &error); + if (error == OPUS_OK && state->decoder != NULL) { /* Creation of memory all ok. */ state->channels = channels; state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize; @@ -173,11 +170,8 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) { } /* If memory allocation was unsuccessful, free the entire state. */ - if (state->decoder_left) { - opus_decoder_destroy(state->decoder_left); - } - if (state->decoder_right) { - opus_decoder_destroy(state->decoder_right); + if (state->decoder) { + opus_decoder_destroy(state->decoder); } free(state); } @@ -186,8 +180,7 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) { int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) { if (inst) { - opus_decoder_destroy(inst->decoder_left); - opus_decoder_destroy(inst->decoder_right); + opus_decoder_destroy(inst->decoder); free(inst); return 0; } else { @@ -199,24 +192,8 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst) { return inst->channels; } -int16_t WebRtcOpus_DecoderInitNew(OpusDecInst* inst) { - int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE); - if (error == OPUS_OK) { - return 0; - } - return -1; -} - int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) { - int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE); - if (error == OPUS_OK) { - return 0; - } - return -1; -} - -int16_t WebRtcOpus_DecoderInitSlave(OpusDecInst* inst) { - int error = opus_decoder_ctl(inst->decoder_right, OPUS_RESET_STATE); + int error = opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE); if (error == OPUS_OK) { return 0; } @@ -256,10 +233,10 @@ static int DecodeFec(OpusDecoder* inst, const uint8_t* encoded, return -1; } -int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded, - int16_t encoded_bytes, int16_t* decoded, - int16_t* audio_type) { - int decoded_samples = DecodeNative(inst->decoder_left, +int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded, + int16_t encoded_bytes, int16_t* decoded, + int16_t* audio_type) { + int decoded_samples = DecodeNative(inst->decoder, encoded, encoded_bytes, kWebRtcOpusMaxFrameSizePerChannel, @@ -275,70 +252,6 @@ int16_t WebRtcOpus_DecodeNew(OpusDecInst* inst, const uint8_t* encoded, return decoded_samples; } -int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded, - int16_t encoded_bytes, int16_t* decoded, - int16_t* audio_type) { - int decoded_samples; - int i; - - /* If mono case, just do a regular call to the decoder. - * If stereo, call to WebRtcOpus_Decode() gives left channel as output, and - * calls to WebRtcOpus_Decode_slave() give right channel as output. - * This is to make stereo work with the current setup of NetEQ, which - * requires two calls to the decoder to produce stereo. */ - - decoded_samples = DecodeNative(inst->decoder_left, encoded, encoded_bytes, - kWebRtcOpusMaxFrameSizePerChannel, decoded, - audio_type); - if (decoded_samples < 0) { - return -1; - } - if (inst->channels == 2) { - /* The parameter |decoded_samples| holds the number of samples pairs, in - * case of stereo. Number of samples in |decoded| equals |decoded_samples| - * times 2. */ - for (i = 0; i < decoded_samples; i++) { - /* Take every second sample, starting at the first sample. This gives - * the left channel. */ - decoded[i] = decoded[i * 2]; - } - } - - /* Update decoded sample memory, to be used by the PLC in case of losses. */ - inst->prev_decoded_samples = decoded_samples; - - return decoded_samples; -} - -int16_t WebRtcOpus_DecodeSlave(OpusDecInst* inst, const uint8_t* encoded, - int16_t encoded_bytes, int16_t* decoded, - int16_t* audio_type) { - int decoded_samples; - int i; - - decoded_samples = DecodeNative(inst->decoder_right, encoded, encoded_bytes, - kWebRtcOpusMaxFrameSizePerChannel, decoded, - audio_type); - if (decoded_samples < 0) { - return -1; - } - if (inst->channels == 2) { - /* The parameter |decoded_samples| holds the number of samples pairs, in - * case of stereo. Number of samples in |decoded| equals |decoded_samples| - * times 2. */ - for (i = 0; i < decoded_samples; i++) { - /* Take every second sample, starting at the second sample. This gives - * the right channel. */ - decoded[i] = decoded[i * 2 + 1]; - } - } else { - /* Decode slave should never be called for mono packets. */ - return -1; - } - - return decoded_samples; -} - int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded, int16_t number_of_lost_frames) { int16_t audio_type = 0; @@ -351,7 +264,7 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded, plc_samples = number_of_lost_frames * inst->prev_decoded_samples; plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ? plc_samples : kWebRtcOpusMaxFrameSizePerChannel; - decoded_samples = DecodeNative(inst->decoder_left, NULL, 0, plc_samples, + decoded_samples = DecodeNative(inst->decoder, NULL, 0, plc_samples, decoded, &audio_type); if (decoded_samples < 0) { return -1; @@ -360,82 +273,6 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded, return decoded_samples; } -int16_t WebRtcOpus_DecodePlcMaster(OpusDecInst* inst, int16_t* decoded, - int16_t number_of_lost_frames) { - int decoded_samples; - int16_t audio_type = 0; - int plc_samples; - int i; - - /* If mono case, just do a regular call to the decoder. - * If stereo, call to WebRtcOpus_DecodePlcMaster() gives left channel as - * output, and calls to WebRtcOpus_DecodePlcSlave() give right channel as - * output. This is to make stereo work with the current setup of NetEQ, which - * requires two calls to the decoder to produce stereo. */ - - /* The number of samples we ask for is |number_of_lost_frames| times - * |prev_decoded_samples_|. Limit the number of samples to maximum - * |kWebRtcOpusMaxFrameSizePerChannel|. */ - plc_samples = number_of_lost_frames * inst->prev_decoded_samples; - plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ? - plc_samples : kWebRtcOpusMaxFrameSizePerChannel; - decoded_samples = DecodeNative(inst->decoder_left, NULL, 0, plc_samples, - decoded, &audio_type); - if (decoded_samples < 0) { - return -1; - } - - if (inst->channels == 2) { - /* The parameter |decoded_samples| holds the number of sample pairs, in - * case of stereo. The original number of samples in |decoded| equals - * |decoded_samples| times 2. */ - for (i = 0; i < decoded_samples; i++) { - /* Take every second sample, starting at the first sample. This gives - * the left channel. */ - decoded[i] = decoded[i * 2]; - } - } - - return decoded_samples; -} - -int16_t WebRtcOpus_DecodePlcSlave(OpusDecInst* inst, int16_t* decoded, - int16_t number_of_lost_frames) { - int decoded_samples; - int16_t audio_type = 0; - int plc_samples; - int i; - - /* Calls to WebRtcOpus_DecodePlcSlave() give right channel as output. - * The function should never be called in the mono case. */ - if (inst->channels != 2) { - return -1; - } - - /* The number of samples we ask for is |number_of_lost_frames| times - * |prev_decoded_samples_|. Limit the number of samples to maximum - * |kWebRtcOpusMaxFrameSizePerChannel|. */ - plc_samples = number_of_lost_frames * inst->prev_decoded_samples; - plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) - ? plc_samples : kWebRtcOpusMaxFrameSizePerChannel; - decoded_samples = DecodeNative(inst->decoder_right, NULL, 0, plc_samples, - decoded, &audio_type); - if (decoded_samples < 0) { - return -1; - } - - /* The parameter |decoded_samples| holds the number of sample pairs, - * The original number of samples in |decoded| equals |decoded_samples| - * times 2. */ - for (i = 0; i < decoded_samples; i++) { - /* Take every second sample, starting at the second sample. This gives - * the right channel. */ - decoded[i] = decoded[i * 2 + 1]; - } - - return decoded_samples; -} - int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, int16_t encoded_bytes, int16_t* decoded, int16_t* audio_type) { @@ -448,7 +285,7 @@ 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, encoded, encoded_bytes, + decoded_samples = DecodeFec(inst->decoder, encoded, encoded_bytes, fec_samples, decoded, audio_type); if (decoded_samples < 0) { return -1; diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc index afbaee38e..e7811e349 100644 --- a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc +++ b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc @@ -72,8 +72,8 @@ float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream, int value; int16_t audio_type; clock_t clocks = clock(); - value = WebRtcOpus_DecodeNew(opus_decoder_, bit_stream, encoded_bytes, - out_data, &audio_type); + value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data, + &audio_type); clocks = clock() - clocks; EXPECT_EQ(output_length_sample_, value); return 1000.0 * clocks / CLOCKS_PER_SEC; diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc index e76dcdcd5..9c34f5169 100644 --- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc @@ -35,9 +35,7 @@ class OpusTest : public ::testing::Test { WebRtcOpusEncInst* opus_mono_encoder_; WebRtcOpusEncInst* opus_stereo_encoder_; WebRtcOpusDecInst* opus_mono_decoder_; - WebRtcOpusDecInst* opus_mono_decoder_new_; WebRtcOpusDecInst* opus_stereo_decoder_; - WebRtcOpusDecInst* opus_stereo_decoder_new_; int16_t speech_data_[kOpusMaxFrameSamples]; int16_t output_data_[kOpusMaxFrameSamples]; @@ -48,9 +46,7 @@ OpusTest::OpusTest() : opus_mono_encoder_(NULL), opus_stereo_encoder_(NULL), opus_mono_decoder_(NULL), - opus_mono_decoder_new_(NULL), - opus_stereo_decoder_(NULL), - opus_stereo_decoder_new_(NULL) { + opus_stereo_decoder_(NULL) { } void OpusTest::SetUp() { @@ -117,91 +113,56 @@ TEST_F(OpusTest, OpusEncodeDecodeMono) { // Create encoder memory. EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1)); EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1)); - EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_new_, 1)); // Set bitrate. EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, 32000)); // Check number of channels for decoder. EXPECT_EQ(1, WebRtcOpus_DecoderChannels(opus_mono_decoder_)); - EXPECT_EQ(1, WebRtcOpus_DecoderChannels(opus_mono_decoder_new_)); // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples]; encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, kOpus20msFrameSamples, kMaxBytes, bitstream_); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); EXPECT_EQ(kOpus20msFrameSamples, WebRtcOpus_Decode(opus_mono_decoder_, bitstream_, encoded_bytes, output_data_decode, &audio_type)); - // Data in |output_data_decode_new| should be the same as in - // |output_data_decode|. - for (int i = 0; i < kOpus20msFrameSamples; i++) { - EXPECT_EQ(output_data_decode_new[i], output_data_decode[i]); - } - // Free memory. EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_)); EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_mono_decoder_)); - EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_mono_decoder_new_)); } TEST_F(OpusTest, OpusEncodeDecodeStereo) { // Create encoder memory. EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2)); EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2)); - EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_new_, 2)); // Set bitrate. EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, 64000)); // Check number of channels for decoder. EXPECT_EQ(2, WebRtcOpus_DecoderChannels(opus_stereo_decoder_)); - EXPECT_EQ(2, WebRtcOpus_DecoderChannels(opus_stereo_decoder_new_)); // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples]; - int16_t output_data_decode_slave[kOpusMaxFrameSamples]; encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, kOpus20msFrameSamples, kMaxBytes, bitstream_); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); EXPECT_EQ(kOpus20msFrameSamples, WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_, encoded_bytes, output_data_decode, &audio_type)); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_, - encoded_bytes, output_data_decode_slave, - &audio_type)); - - // Data in |output_data_decode_new| should be the same as in - // |output_data_decode| and |output_data_decode_slave| interleaved to a - // stereo signal. - for (int i = 0; i < kOpus20msFrameSamples; i++) { - EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]); - EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]); - } // Free memory. EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_)); EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_)); - EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_new_)); } TEST_F(OpusTest, OpusSetBitRate) { @@ -249,67 +210,30 @@ TEST_F(OpusTest, OpusDecodeInit) { // Create encoder memory. EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2)); EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2)); - EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_new_, 2)); // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples]; - int16_t output_data_decode_slave[kOpusMaxFrameSamples]; + encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, kOpus20msFrameSamples, kMaxBytes, bitstream_); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); EXPECT_EQ(kOpus20msFrameSamples, WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_, encoded_bytes, output_data_decode, &audio_type)); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_, - encoded_bytes, output_data_decode_slave, - &audio_type)); - // Data in |output_data_decode_new| should be the same as in - // |output_data_decode| and |output_data_decode_slave| interleaved to a - // stereo signal. - for (int i = 0; i < kOpus20msFrameSamples; i++) { - EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]); - EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]); - } - - EXPECT_EQ(0, WebRtcOpus_DecoderInitNew(opus_stereo_decoder_new_)); EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_stereo_decoder_)); - EXPECT_EQ(0, WebRtcOpus_DecoderInitSlave(opus_stereo_decoder_)); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); EXPECT_EQ(kOpus20msFrameSamples, WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_, encoded_bytes, output_data_decode, &audio_type)); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_, - encoded_bytes, output_data_decode_slave, - &audio_type)); - - // Data in |output_data_decode_new| should be the same as in - // |output_data_decode| and |output_data_decode_slave| interleaved to a - // stereo signal. - for (int i = 0; i < kOpus20msFrameSamples; i++) { - EXPECT_EQ(output_data_decode_new[i * 2], output_data_decode[i]); - EXPECT_EQ(output_data_decode_new[i * 2 + 1], output_data_decode_slave[i]); - } // Free memory. EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_)); EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_)); - EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_new_)); } TEST_F(OpusTest, OpusEnableDisableFec) { @@ -382,49 +306,33 @@ TEST_F(OpusTest, OpusDecodePlcMono) { // Create encoder memory. EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1)); EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1)); - EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_mono_decoder_new_, 1)); // Set bitrate. EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, 32000)); // Check number of channels for decoder. EXPECT_EQ(1, WebRtcOpus_DecoderChannels(opus_mono_decoder_)); - EXPECT_EQ(1, WebRtcOpus_DecoderChannels(opus_mono_decoder_new_)); // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples]; encoded_bytes = WebRtcOpus_Encode(opus_mono_encoder_, speech_data_, kOpus20msFrameSamples, kMaxBytes, bitstream_); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeNew(opus_mono_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); EXPECT_EQ(kOpus20msFrameSamples, WebRtcOpus_Decode(opus_mono_decoder_, bitstream_, encoded_bytes, output_data_decode, &audio_type)); - // Call decoder PLC for both versions of the decoder. + // Call decoder PLC. int16_t plc_buffer[kOpusMaxFrameSamples]; - int16_t plc_buffer_new[kOpusMaxFrameSamples]; EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodePlcMaster(opus_mono_decoder_, plc_buffer, 1)); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodePlc(opus_mono_decoder_new_, plc_buffer_new, 1)); - - // Data in |plc_buffer| should be the same as in |plc_buffer_new|. - for (int i = 0; i < kOpus20msFrameSamples; i++) { - EXPECT_EQ(plc_buffer[i], plc_buffer_new[i]); - } + WebRtcOpus_DecodePlc(opus_mono_decoder_, plc_buffer, 1)); // Free memory. EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_mono_encoder_)); EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_mono_decoder_)); - EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_mono_decoder_new_)); } // PLC in stereo mode. @@ -432,61 +340,33 @@ TEST_F(OpusTest, OpusDecodePlcStereo) { // Create encoder memory. EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2)); EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2)); - EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_stereo_decoder_new_, 2)); // Set bitrate. EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, 64000)); // Check number of channels for decoder. EXPECT_EQ(2, WebRtcOpus_DecoderChannels(opus_stereo_decoder_)); - EXPECT_EQ(2, WebRtcOpus_DecoderChannels(opus_stereo_decoder_new_)); // Encode & decode. int16_t encoded_bytes; int16_t audio_type; - int16_t output_data_decode_new[kOpusMaxFrameSamples]; int16_t output_data_decode[kOpusMaxFrameSamples]; - int16_t output_data_decode_slave[kOpusMaxFrameSamples]; encoded_bytes = WebRtcOpus_Encode(opus_stereo_encoder_, speech_data_, kOpus20msFrameSamples, kMaxBytes, bitstream_); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeNew(opus_stereo_decoder_new_, bitstream_, - encoded_bytes, output_data_decode_new, - &audio_type)); EXPECT_EQ(kOpus20msFrameSamples, WebRtcOpus_Decode(opus_stereo_decoder_, bitstream_, encoded_bytes, output_data_decode, &audio_type)); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodeSlave(opus_stereo_decoder_, bitstream_, - encoded_bytes, - output_data_decode_slave, - &audio_type)); - // Call decoder PLC for both versions of the decoder. - int16_t plc_buffer_left[kOpusMaxFrameSamples]; - int16_t plc_buffer_right[kOpusMaxFrameSamples]; - int16_t plc_buffer_new[kOpusMaxFrameSamples]; + // Call decoder PLC. + int16_t plc_buffer[kOpusMaxFrameSamples]; EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodePlcMaster(opus_stereo_decoder_, - plc_buffer_left, 1)); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodePlcSlave(opus_stereo_decoder_, - plc_buffer_right, 1)); - EXPECT_EQ(kOpus20msFrameSamples, - WebRtcOpus_DecodePlc(opus_stereo_decoder_new_, plc_buffer_new, 1)); - // Data in |plc_buffer_left| and |plc_buffer_right|should be the same as the - // interleaved samples in |plc_buffer_new|. - for (int i = 0, j = 0; i < kOpus20msFrameSamples; i++) { - EXPECT_EQ(plc_buffer_left[i], plc_buffer_new[j++]); - EXPECT_EQ(plc_buffer_right[i], plc_buffer_new[j++]); - } + WebRtcOpus_DecodePlc(opus_stereo_decoder_, plc_buffer, 1)); // Free memory. EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_stereo_encoder_)); EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_)); - EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_stereo_decoder_new_)); } // Duration estimation. diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc index 398d59da4..ecc056a36 100644 --- a/webrtc/modules/audio_coding/main/test/opus_test.cc +++ b/webrtc/modules/audio_coding/main/test/opus_test.cc @@ -85,8 +85,8 @@ void OpusTest::Perform() { // Create Opus decoders for mono and stereo for stand-alone testing of Opus. ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1); ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1); - ASSERT_GT(WebRtcOpus_DecoderInitNew(opus_mono_decoder_), -1); - ASSERT_GT(WebRtcOpus_DecoderInitNew(opus_stereo_decoder_), -1); + ASSERT_GT(WebRtcOpus_DecoderInit(opus_mono_decoder_), -1); + ASSERT_GT(WebRtcOpus_DecoderInit(opus_stereo_decoder_), -1); ASSERT_TRUE(acm_receiver_.get() != NULL); EXPECT_EQ(0, acm_receiver_->InitializeReceiver()); @@ -304,7 +304,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, // Run stand-alone Opus decoder, or decode PLC. if (channels == 1) { if (!lost_packet) { - decoded_samples += WebRtcOpus_DecodeNew( + decoded_samples += WebRtcOpus_Decode( opus_mono_decoder_, bitstream, bitstream_len_byte, &out_audio[decoded_samples * channels], &audio_type); } else { @@ -313,7 +313,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, } } else { if (!lost_packet) { - decoded_samples += WebRtcOpus_DecodeNew( + decoded_samples += WebRtcOpus_Decode( opus_stereo_decoder_, bitstream, bitstream_len_byte, &out_audio[decoded_samples * channels], &audio_type); } else { diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc index eb0782344..799ac7bdd 100644 --- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc @@ -404,9 +404,9 @@ AudioDecoderOpus::~AudioDecoderOpus() { int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len, int16_t* decoded, SpeechType* speech_type) { int16_t temp_type = 1; // Default is speech. - int16_t ret = WebRtcOpus_DecodeNew(dec_state_, encoded, - static_cast(encoded_len), decoded, - &temp_type); + int16_t ret = WebRtcOpus_Decode(dec_state_, encoded, + static_cast(encoded_len), decoded, + &temp_type); if (ret > 0) ret *= static_cast(channels_); // Return total number of samples. *speech_type = ConvertSpeechType(temp_type); @@ -427,7 +427,7 @@ int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded, } int AudioDecoderOpus::Init() { - return WebRtcOpus_DecoderInitNew(dec_state_); + return WebRtcOpus_DecoderInit(dec_state_); } int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,