Renaming opus_interface.c to opus_interface.cc.
This is to allow advanced features of WebRTC/Chrome e.g., field trials.
More style compliant changes may follow up. Only a minimal (not in terms of line changes) is applied, so that presubmit does not complain. These changes include
1. removing unused headers.
2. eliminating c-style casting.
Bug: b/143582588
Change-Id: I6d0fd926c542ab0afdc38cc4bf03aaf584ec13dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158670
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29657}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 29aa1e7..0dda20b 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -765,7 +765,7 @@
"//third_party/abseil-cpp/absl/types:optional",
]
public_deps = [ # no-presubmit-check TODO(webrtc:8603)
- ":webrtc_opus_c",
+ ":webrtc_opus_wrapper",
]
defines = audio_codec_defines
@@ -803,7 +803,7 @@
"//third_party/abseil-cpp/absl/types:optional",
]
public_deps = [ # no-presubmit-check TODO(webrtc:8603)
- ":webrtc_opus_c",
+ ":webrtc_opus_wrapper",
]
defines = audio_codec_defines
@@ -815,11 +815,11 @@
}
}
-rtc_library("webrtc_opus_c") {
+rtc_library("webrtc_opus_wrapper") {
poisonous = [ "audio_codecs" ]
sources = [
"codecs/opus/opus_inst.h",
- "codecs/opus/opus_interface.c",
+ "codecs/opus/opus_interface.cc",
"codecs/opus/opus_interface.h",
]
@@ -1296,7 +1296,7 @@
":audio_encoder_cng",
":pcm16b_c",
":red",
- ":webrtc_opus_c",
+ ":webrtc_opus_wrapper",
"..:module_api",
"../../api:rtp_headers",
"../../api/audio:audio_frame_api",
diff --git a/modules/audio_coding/codecs/opus/opus_interface.c b/modules/audio_coding/codecs/opus/opus_interface.cc
similarity index 79%
rename from modules/audio_coding/codecs/opus/opus_interface.c
rename to modules/audio_coding/codecs/opus/opus_interface.cc
index f8ff656..45eab2b 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -12,9 +12,6 @@
#include "rtc_base/checks.h"
-#include <stdlib.h>
-#include <string.h>
-
enum {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
/* Maximum supported frame size in WebRTC is 120 ms. */
@@ -67,15 +64,15 @@
return -1;
}
- OpusEncInst* state = (OpusEncInst*)calloc(1, sizeof(OpusEncInst));
+ OpusEncInst* state =
+ reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst)));
RTC_DCHECK(state);
int error;
- state->encoder = opus_encoder_create(sample_rate_hz, (int)channels, opus_app,
- &error);
+ state->encoder = opus_encoder_create(
+ sample_rate_hz, static_cast<int>(channels), opus_app, &error);
- if (error != OPUS_OK || (!state->encoder &&
- !state->multistream_encoder)) {
+ if (error != OPUS_OK || (!state->encoder && !state->multistream_encoder)) {
WebRtcOpus_EncoderFree(state);
return -1;
}
@@ -93,7 +90,7 @@
int32_t application,
size_t streams,
size_t coupled_streams,
- const unsigned char *channel_mapping) {
+ const unsigned char* channel_mapping) {
int opus_app;
if (!inst)
return -1;
@@ -109,22 +106,16 @@
return -1;
}
- OpusEncInst* state = (OpusEncInst*)calloc(1, sizeof(OpusEncInst));
+ OpusEncInst* state =
+ reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst)));
RTC_DCHECK(state);
int error;
state->multistream_encoder =
- opus_multistream_encoder_create(
- 48000,
- channels,
- streams,
- coupled_streams,
- channel_mapping,
- opus_app,
- &error);
+ opus_multistream_encoder_create(48000, channels, streams, coupled_streams,
+ channel_mapping, opus_app, &error);
- if (error != OPUS_OK || (!state->encoder &&
- !state->multistream_encoder)) {
+ if (error != OPUS_OK || (!state->encoder && !state->multistream_encoder)) {
WebRtcOpus_EncoderFree(state);
return -1;
}
@@ -162,17 +153,14 @@
}
if (inst->encoder) {
- res = opus_encode(inst->encoder,
- (const opus_int16*)audio_in,
- (int)samples,
- encoded,
- (opus_int32)length_encoded_buffer);
+ res = opus_encode(inst->encoder, (const opus_int16*)audio_in,
+ static_cast<int>(samples), encoded,
+ static_cast<opus_int32>(length_encoded_buffer));
} else {
- res = opus_multistream_encode(inst->multistream_encoder,
- (const opus_int16*)audio_in,
- (int)samples,
- encoded,
- (opus_int32)length_encoded_buffer);
+ res = opus_multistream_encode(
+ inst->multistream_encoder, (const opus_int16*)audio_in,
+ static_cast<int>(samples), encoded,
+ static_cast<opus_int32>(length_encoded_buffer));
}
if (res <= 0) {
@@ -195,11 +183,10 @@
return res;
}
-#define ENCODER_CTL(inst, vargs) ( \
- inst->encoder ? \
- opus_encoder_ctl(inst->encoder, vargs) \
- : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs))
-
+#define ENCODER_CTL(inst, vargs) \
+ (inst->encoder \
+ ? opus_encoder_ctl(inst->encoder, vargs) \
+ : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs))
int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
if (inst) {
@@ -240,9 +227,8 @@
int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
int32_t* result_hz) {
if (inst->encoder) {
- if (opus_encoder_ctl(
- inst->encoder,
- OPUS_GET_MAX_BANDWIDTH(result_hz)) == OPUS_OK) {
+ if (opus_encoder_ctl(inst->encoder, OPUS_GET_MAX_BANDWIDTH(result_hz)) ==
+ OPUS_OK) {
return 0;
}
return -1;
@@ -256,7 +242,7 @@
ret = OPUS_OK;
s = 0;
while (ret == OPUS_OK) {
- OpusEncoder *enc;
+ OpusEncoder* enc;
opus_int32 bandwidth;
ret = ENCODER_CTL(inst, OPUS_MULTISTREAM_GET_ENCODER_STATE(s, &enc));
@@ -303,8 +289,7 @@
// last long during a pure silence, if the signal type is not forced.
// TODO(minyue): Remove the signal type forcing when Opus DTX works properly
// without it.
- int ret = ENCODER_CTL(inst,
- OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
+ int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
if (ret != OPUS_OK)
return ret;
@@ -313,8 +298,7 @@
int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
if (inst) {
- int ret = ENCODER_CTL(inst,
- OPUS_SET_SIGNAL(OPUS_AUTO));
+ int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_AUTO));
if (ret != OPUS_OK)
return ret;
return ENCODER_CTL(inst, OPUS_SET_DTX(0));
@@ -341,8 +325,7 @@
int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
if (inst) {
- return ENCODER_CTL(inst,
- OPUS_SET_COMPLEXITY(complexity));
+ return ENCODER_CTL(inst, OPUS_SET_COMPLEXITY(complexity));
} else {
return -1;
}
@@ -353,19 +336,16 @@
return -1;
}
int32_t bandwidth;
- if (ENCODER_CTL(inst,
- OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
+ if (ENCODER_CTL(inst, OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
return bandwidth;
} else {
return -1;
}
-
}
int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth) {
if (inst) {
- return ENCODER_CTL(inst,
- OPUS_SET_BANDWIDTH(bandwidth));
+ return ENCODER_CTL(inst, OPUS_SET_BANDWIDTH(bandwidth));
} else {
return -1;
}
@@ -375,11 +355,9 @@
if (!inst)
return -1;
if (num_channels == 0) {
- return ENCODER_CTL(inst,
- OPUS_SET_FORCE_CHANNELS(OPUS_AUTO));
+ return ENCODER_CTL(inst, OPUS_SET_FORCE_CHANNELS(OPUS_AUTO));
} else if (num_channels == 1 || num_channels == 2) {
- return ENCODER_CTL(inst,
- OPUS_SET_FORCE_CHANNELS(num_channels));
+ return ENCODER_CTL(inst, OPUS_SET_FORCE_CHANNELS(num_channels));
} else {
return -1;
}
@@ -393,12 +371,13 @@
if (inst != NULL) {
// Create Opus decoder state.
- state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
+ state = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst)));
if (state == NULL) {
return -1;
}
- state->decoder = opus_decoder_create(sample_rate_hz, (int)channels, &error);
+ state->decoder =
+ opus_decoder_create(sample_rate_hz, static_cast<int>(channels), &error);
if (error == OPUS_OK && state->decoder) {
// Creation of memory all ok.
state->channels = channels;
@@ -419,7 +398,8 @@
}
int16_t WebRtcOpus_MultistreamDecoderCreate(
- OpusDecInst** inst, size_t channels,
+ OpusDecInst** inst,
+ size_t channels,
size_t streams,
size_t coupled_streams,
const unsigned char* channel_mapping) {
@@ -428,18 +408,14 @@
if (inst != NULL) {
// Create Opus decoder state.
- state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
+ state = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst)));
if (state == NULL) {
return -1;
}
// Create new memory, always at 48000 Hz.
state->multistream_decoder = opus_multistream_decoder_create(
- 48000, channels,
- streams,
- coupled_streams,
- channel_mapping,
- &error);
+ 48000, channels, streams, coupled_streams, channel_mapping, &error);
if (error == OPUS_OK && state->multistream_decoder) {
// Creation of memory all ok.
@@ -480,8 +456,7 @@
if (inst->decoder) {
opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
} else {
- opus_multistream_decoder_ctl(inst->multistream_decoder,
- OPUS_RESET_STATE);
+ opus_multistream_decoder_ctl(inst->multistream_decoder, OPUS_RESET_STATE);
}
inst->in_dtx_mode = 0;
}
@@ -512,17 +487,23 @@
/* |frame_size| is set to maximum Opus frame size in the normal case, and
* is set to the number of samples needed for PLC in case of losses.
* It is up to the caller to make sure the value is correct. */
-static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
- size_t encoded_bytes, int frame_size,
- int16_t* decoded, int16_t* audio_type, int decode_fec) {
+static int DecodeNative(OpusDecInst* inst,
+ const uint8_t* encoded,
+ size_t encoded_bytes,
+ int frame_size,
+ int16_t* decoded,
+ int16_t* audio_type,
+ int decode_fec) {
int res = -1;
if (inst->decoder) {
- res = opus_decode(inst->decoder, encoded, (opus_int32)encoded_bytes,
- (opus_int16*)decoded, frame_size, decode_fec);
+ res = opus_decode(
+ inst->decoder, encoded, static_cast<opus_int32>(encoded_bytes),
+ reinterpret_cast<opus_int16*>(decoded), frame_size, decode_fec);
} else {
- res = opus_multistream_decode(
- inst->multistream_decoder, encoded, (opus_int32)encoded_bytes,
- (opus_int16*)decoded, frame_size, decode_fec);
+ res = opus_multistream_decode(inst->multistream_decoder, encoded,
+ static_cast<opus_int32>(encoded_bytes),
+ reinterpret_cast<opus_int16*>(decoded),
+ frame_size, decode_fec);
}
if (res <= 0)
@@ -533,8 +514,10 @@
return res;
}
-int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
- size_t encoded_bytes, int16_t* decoded,
+int WebRtcOpus_Decode(OpusDecInst* inst,
+ const uint8_t* encoded,
+ size_t encoded_bytes,
+ int16_t* decoded,
int16_t* audio_type) {
int decoded_samples;
@@ -556,7 +539,8 @@
return decoded_samples;
}
-int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+int WebRtcOpus_DecodePlc(OpusDecInst* inst,
+ int16_t* decoded,
int number_of_lost_frames) {
int16_t audio_type = 0;
int decoded_samples;
@@ -571,8 +555,8 @@
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);
+ decoded_samples =
+ DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0);
if (decoded_samples < 0) {
return -1;
}
@@ -580,8 +564,10 @@
return decoded_samples;
}
-int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
- size_t encoded_bytes, int16_t* decoded,
+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
+ const uint8_t* encoded,
+ size_t encoded_bytes,
+ int16_t* decoded,
int16_t* audio_type) {
int decoded_samples;
int fec_samples;
@@ -593,8 +579,8 @@
fec_samples =
opus_packet_get_samples_per_frame(encoded, inst->sample_rate_hz);
- decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
- fec_samples, decoded, audio_type, 1);
+ decoded_samples = DecodeNative(inst, encoded, encoded_bytes, fec_samples,
+ decoded, audio_type, 1);
if (decoded_samples < 0) {
return -1;
}
@@ -612,7 +598,8 @@
}
int frames, samples;
- frames = opus_packet_get_nb_frames(payload, (opus_int32)payload_length_bytes);
+ frames = opus_packet_get_nb_frames(
+ payload, static_cast<opus_int32>(payload_length_bytes));
if (frames < 0) {
/* Invalid payload data. */
return 0;
@@ -667,12 +654,12 @@
// Max number of frames in an Opus packet is 48.
opus_int16 frame_sizes[48];
- const unsigned char *frame_data[48];
+ const unsigned char* frame_data[48];
// Parse packet to get the frames. But we only care about the first frame,
// since we can only decode the FEC from the first one.
- if (opus_packet_parse(payload, (opus_int32)payload_length_bytes, NULL,
- frame_data, frame_sizes, NULL) < 0) {
+ if (opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes),
+ NULL, frame_data, frame_sizes, NULL) < 0) {
return 0;
}
@@ -700,7 +687,7 @@
silk_frames = 3;
break;
default:
- return 0; // It is actually even an invalid packet.
+ return 0; // It is actually even an invalid packet.
}
const int channels = opus_packet_get_nb_channels(payload);