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;