Removing unused Opus wrapper API: WebRTCOpus_DecodePlc. Bug: None Change-Id: I5b613b4c13ec5f6ad13d8430043d006f6d83c11f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158671 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29664}
diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc index 1ab4d86..47e40c6 100644 --- a/modules/audio_coding/codecs/opus/opus_fec_test.cc +++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -154,7 +154,8 @@ WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_, &out_data_[0], &audio_type); } else { - value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1); + value_1 = + WebRtcOpus_Decode(opus_decoder_, NULL, 0, &out_data_[0], &audio_type); } EXPECT_EQ(static_cast<int>(block_length_sample_), value_1); }
diff --git a/modules/audio_coding/codecs/opus/opus_interface.cc b/modules/audio_coding/codecs/opus/opus_interface.cc index 45eab2b..fc3d3ff 100644 --- a/modules/audio_coding/codecs/opus/opus_interface.cc +++ b/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -514,6 +514,29 @@ return res; } +static int DecodePlc(OpusDecInst* inst, int16_t* decoded) { + int16_t audio_type = 0; + int decoded_samples; + int plc_samples; + + /* The number of samples we ask for is |number_of_lost_frames| times + * |prev_decoded_samples_|. Limit the number of samples to maximum + * |MaxFrameSizePerChannel()|. */ + plc_samples = inst->prev_decoded_samples; + const int max_samples_per_channel = + MaxFrameSizePerChannel(inst->sample_rate_hz); + plc_samples = plc_samples <= max_samples_per_channel + ? plc_samples + : max_samples_per_channel; + decoded_samples = + DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0); + if (decoded_samples < 0) { + return -1; + } + + return decoded_samples; +} + int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded, size_t encoded_bytes, @@ -523,7 +546,7 @@ if (encoded_bytes == 0) { *audio_type = DetermineAudioType(inst, encoded_bytes); - decoded_samples = WebRtcOpus_DecodePlc(inst, decoded, 1); + decoded_samples = DecodePlc(inst, decoded); } else { decoded_samples = DecodeNative(inst, encoded, encoded_bytes, MaxFrameSizePerChannel(inst->sample_rate_hz), @@ -539,31 +562,6 @@ return decoded_samples; } -int WebRtcOpus_DecodePlc(OpusDecInst* inst, - int16_t* decoded, - int number_of_lost_frames) { - int16_t audio_type = 0; - int decoded_samples; - int plc_samples; - - /* The number of samples we ask for is |number_of_lost_frames| times - * |prev_decoded_samples_|. Limit the number of samples to maximum - * |MaxFrameSizePerChannel()|. */ - plc_samples = number_of_lost_frames * inst->prev_decoded_samples; - const int max_samples_per_channel = - MaxFrameSizePerChannel(inst->sample_rate_hz); - plc_samples = plc_samples <= max_samples_per_channel - ? plc_samples - : max_samples_per_channel; - decoded_samples = - DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0); - if (decoded_samples < 0) { - return -1; - } - - return decoded_samples; -} - int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded, size_t encoded_bytes,
diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h index cf95a69..ef62e0d 100644 --- a/modules/audio_coding/codecs/opus/opus_interface.h +++ b/modules/audio_coding/codecs/opus/opus_interface.h
@@ -407,24 +407,6 @@ int16_t* audio_type); /**************************************************************************** - * WebRtcOpus_DecodePlc(...) - * - * This function processes PLC for opus frame(s). - * Input: - * - inst : Decoder context - * - number_of_lost_frames : Number of PLC frames to produce - * - * Output: - * - decoded : The decoded vector - * - * Return value : >0 - number of samples in decoded PLC vector - * -1 - Error - */ -int WebRtcOpus_DecodePlc(OpusDecInst* inst, - int16_t* decoded, - int number_of_lost_frames); - -/**************************************************************************** * WebRtcOpus_DecodeFec(...) * * This function decodes the FEC data from an Opus packet into one or more audio
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc index f0f2ef0..10897fb 100644 --- a/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -810,7 +810,7 @@ // Call decoder PLC. int16_t* plc_buffer = new int16_t[decode_samples_per_channel * channels_]; EXPECT_EQ(decode_samples_per_channel, - WebRtcOpus_DecodePlc(opus_decoder_, plc_buffer, 1)); + WebRtcOpus_Decode(opus_decoder_, NULL, 0, plc_buffer, &audio_type)); // Free memory. delete[] plc_buffer;
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc index 7f0bdd2..10644e2 100644 --- a/modules/audio_coding/test/opus_test.cc +++ b/modules/audio_coding/test/opus_test.cc
@@ -299,8 +299,9 @@ opus_mono_decoder_, bitstream, bitstream_len_byte, &out_audio[decoded_samples * channels], &audio_type); } else { - decoded_samples += WebRtcOpus_DecodePlc( - opus_mono_decoder_, &out_audio[decoded_samples * channels], 1); + decoded_samples += WebRtcOpus_Decode( + opus_mono_decoder_, NULL, 0, + &out_audio[decoded_samples * channels], &audio_type); } } else { if (!lost_packet) { @@ -308,9 +309,9 @@ opus_stereo_decoder_, bitstream, bitstream_len_byte, &out_audio[decoded_samples * channels], &audio_type); } else { - decoded_samples += - WebRtcOpus_DecodePlc(opus_stereo_decoder_, - &out_audio[decoded_samples * channels], 1); + decoded_samples += WebRtcOpus_Decode( + opus_stereo_decoder_, NULL, 0, + &out_audio[decoded_samples * channels], &audio_type); } }