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Removing unused Opus wrapper API: WebRTCOpus_DecodePlc.

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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}
This commit is contained in:
Minyue Li
2019-10-29 21:38:15 +01:00
committed by Commit Bot
parent 0cbb58e046
commit fb075d558d
5 changed files with 33 additions and 51 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
modules/audio_coding/codecs/opus/opus_fec_test.cc
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@ -154,7 +154,8 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
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);
}

50
modules/audio_coding/codecs/opus/opus_interface.cc
View File

@ -514,6 +514,29 @@ static int DecodeNative(OpusDecInst* inst,
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 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
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 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
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,

18
modules/audio_coding/codecs/opus/opus_interface.h
View File

@ -406,24 +406,6 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
int16_t* decoded,
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(...)
*

2
modules/audio_coding/codecs/opus/opus_unittest.cc
View File

@ -810,7 +810,7 @@ TEST_P(OpusTest, OpusDecodePlc) {
// 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;