Revert "Opus multistream."
This reverts commit 83ed89a45f4578ca07efef48e772b9aafb263163.
Reason for revert: breaks downstream project
Original change's description:
> Opus multistream.
>
> This is a backwards-compatible change. It makes WebRTC use the Opus
> multistream decoder for all Opus packets. Single-stream packets are a
> special case of multistream ones (with stream=1).
>
> The tricky parts are 'WebRtcOpus_GetMaxPlaybackRate' and
> 'WebRtcOpus_GetSurroundParameters'. GetMaxPlaybackRate is supposed to
> do what opus_encoder_ctl(encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth))
> did when we had single-stream encoders. Now there may be several
> independent encoders with possibly different BANDWIDTH. The new
> GetMaxPlaybackRate queries all of them, and returns a playback rate if
> all the encoder's rates are equal.
>
> WebRtcOpus_GetSurroundParameters is a configuration convention. It
> maps the number of channels to a multi-stream encoder/decoder
> configuration. As described in RFC 7845
> https://tools.ietf.org/html/rfc7845#section-5.1.1, a multi-stream
> encoder/decoder needs a number of streams, number of coupled streams
> and a 255-byte mapping array. The function GetSurroundParameters
> computes all of these from the number of channels. [1, 2, 4, 6, 8]
> channels are supported.
>
> Bug: webrtc:8649
> Change-Id: I271de8e387d738254d6aa53af7fcf8644a53edb5
> Reviewed-on: https://webrtc-review.googlesource.com/c/111750
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26293}
TBR=aleloi@webrtc.org,minyue@webrtc.org
Change-Id: I1002e3273b44d3cccacdba84b8c363eefd537c4b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/118201
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26306}diff --git a/modules/audio_coding/codecs/opus/opus_interface.c b/modules/audio_coding/codecs/opus/opus_interface.c
index c657a14..d219098 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/modules/audio_coding/codecs/opus/opus_interface.c
@@ -37,40 +37,6 @@
kWebRtcOpusDefaultFrameSize = 960,
};
-int16_t GetSurroundParameters(int channels,
- int *streams,
- int *coupled_streams,
- unsigned char *mapping) {
- int opus_error;
- int ret = 0;
- // Use 'surround encoder create' to get values for 'coupled_streams',
- // 'streams' and 'mapping'.
- OpusMSEncoder* ms_encoder_ptr = opus_multistream_surround_encoder_create(
- 48000,
- channels,
- /* mapping family */ channels <= 2 ? 0 : 1,
- streams,
- coupled_streams,
- mapping,
- OPUS_APPLICATION_VOIP, // Application type shouldn't affect
- // streams/mapping values.
- &opus_error);
-
- // This shouldn't fail; if it fails,
- // signal an error and return invalid values.
- if (opus_error != OPUS_OK || ms_encoder_ptr == NULL) {
- ret = -1;
- *streams = -1;
- *coupled_streams = -1;
- }
-
- // We don't need the encoder.
- if (ms_encoder_ptr != NULL) {
- opus_multistream_encoder_destroy(ms_encoder_ptr);
- }
- return ret;
-}
-
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
size_t channels,
int32_t application) {
@@ -89,26 +55,12 @@
return -1;
}
- unsigned char mapping[255];
- memset(mapping, 0, 255);
- int streams = -1;
- int coupled_streams = -1;
-
-
OpusEncInst* state = calloc(1, sizeof(OpusEncInst));
RTC_DCHECK(state);
int error;
- state->encoder = opus_multistream_surround_encoder_create(
- 48000,
- channels,
- /* mapping family */ channels <= 2 ? 0 : 1,
- &streams,
- &coupled_streams,
- mapping,
- opus_app,
- &error);
-
+ state->encoder = opus_encoder_create(48000, (int)channels, opus_app,
+ &error);
if (error != OPUS_OK || !state->encoder) {
WebRtcOpus_EncoderFree(state);
return -1;
@@ -123,7 +75,7 @@
int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
if (inst) {
- opus_multistream_encoder_destroy(inst->encoder);
+ opus_encoder_destroy(inst->encoder);
free(inst);
return 0;
} else {
@@ -142,11 +94,11 @@
return -1;
}
- res = opus_multistream_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,
+ (int)samples,
+ encoded,
+ (opus_int32)length_encoded_buffer);
if (res <= 0) {
return -1;
@@ -170,7 +122,7 @@
int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
if (inst) {
- return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate));
} else {
return -1;
}
@@ -178,8 +130,8 @@
int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) {
if (inst) {
- return opus_multistream_encoder_ctl(inst->encoder,
- OPUS_SET_PACKET_LOSS_PERC(loss_rate));
+ return opus_encoder_ctl(inst->encoder,
+ OPUS_SET_PACKET_LOSS_PERC(loss_rate));
} else {
return -1;
}
@@ -202,46 +154,13 @@
} else {
set_bandwidth = OPUS_BANDWIDTH_FULLBAND;
}
- return opus_multistream_encoder_ctl(inst->encoder,
- OPUS_SET_MAX_BANDWIDTH(set_bandwidth));
-}
-
-int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
- int32_t* result_hz) {
- opus_int32 max_bandwidth;
- int s;
- int ret;
-
- max_bandwidth = 0;
- ret = OPUS_OK;
- s = 0;
- while (ret == OPUS_OK) {
- OpusEncoder *enc;
- opus_int32 bandwidth;
-
- ret = opus_multistream_encoder_ctl(
- inst->encoder,
- OPUS_MULTISTREAM_GET_ENCODER_STATE(s, &enc));
- if (ret == OPUS_BAD_ARG)
- break;
- if (ret != OPUS_OK)
- return -1;
- if (opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&bandwidth)) != OPUS_OK)
- return -1;
-
- if (max_bandwidth != 0 && max_bandwidth != bandwidth)
- return -1;
-
- max_bandwidth = bandwidth;
- s++;
- }
- *result_hz = max_bandwidth;
- return 0;
+ return opus_encoder_ctl(inst->encoder,
+ OPUS_SET_MAX_BANDWIDTH(set_bandwidth));
}
int16_t WebRtcOpus_EnableFec(OpusEncInst* inst) {
if (inst) {
- return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(1));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(1));
} else {
return -1;
}
@@ -249,7 +168,7 @@
int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) {
if (inst) {
- return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(0));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_INBAND_FEC(0));
} else {
return -1;
}
@@ -265,21 +184,21 @@
// 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 = opus_multistream_encoder_ctl(inst->encoder,
- OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
+ int ret = opus_encoder_ctl(inst->encoder,
+ OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
if (ret != OPUS_OK)
return ret;
- return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_DTX(1));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(1));
}
int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
if (inst) {
- int ret = opus_multistream_encoder_ctl(inst->encoder,
- OPUS_SET_SIGNAL(OPUS_AUTO));
+ int ret = opus_encoder_ctl(inst->encoder,
+ OPUS_SET_SIGNAL(OPUS_AUTO));
if (ret != OPUS_OK)
return ret;
- return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_DTX(0));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(0));
} else {
return -1;
}
@@ -287,7 +206,7 @@
int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst) {
if (inst) {
- return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_VBR(0));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_VBR(0));
} else {
return -1;
}
@@ -295,7 +214,7 @@
int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst) {
if (inst) {
- return opus_multistream_encoder_ctl(inst->encoder, OPUS_SET_VBR(1));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_VBR(1));
} else {
return -1;
}
@@ -303,8 +222,7 @@
int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
if (inst) {
- return opus_multistream_encoder_ctl(inst->encoder,
- OPUS_SET_COMPLEXITY(complexity));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_COMPLEXITY(complexity));
} else {
return -1;
}
@@ -315,8 +233,7 @@
return -1;
}
int32_t bandwidth;
- if (opus_multistream_encoder_ctl(inst->encoder,
- OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
+ if (opus_encoder_ctl(inst->encoder, OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
return bandwidth;
} else {
return -1;
@@ -326,8 +243,7 @@
int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth) {
if (inst) {
- return opus_multistream_encoder_ctl(inst->encoder,
- OPUS_SET_BANDWIDTH(bandwidth));
+ return opus_encoder_ctl(inst->encoder, OPUS_SET_BANDWIDTH(bandwidth));
} else {
return -1;
}
@@ -337,10 +253,10 @@
if (!inst)
return -1;
if (num_channels == 0) {
- return opus_multistream_encoder_ctl(inst->encoder,
+ return opus_encoder_ctl(inst->encoder,
OPUS_SET_FORCE_CHANNELS(OPUS_AUTO));
} else if (num_channels == 1 || num_channels == 2) {
- return opus_multistream_encoder_ctl(inst->encoder,
+ return opus_encoder_ctl(inst->encoder,
OPUS_SET_FORCE_CHANNELS(num_channels));
} else {
return -1;
@@ -352,31 +268,16 @@
OpusDecInst* state;
if (inst != NULL) {
- // Create Opus decoder state.
+ /* Create Opus decoder state. */
state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
if (state == NULL) {
return -1;
}
- unsigned char mapping[255];
- memset(mapping, 0, 255);
- int streams = -1;
- int coupled_streams = -1;
- if (GetSurroundParameters(channels, &streams,
- &coupled_streams, mapping) != 0) {
- free(state);
- return -1;
- }
-
- // Create new memory, always at 48000 Hz.
- state->decoder = opus_multistream_decoder_create(
- 48000, (int)channels,
- /* streams = */ streams,
- /* coupled streams = */ coupled_streams,
- mapping,
- &error);
+ /* Create new memory, always at 48000 Hz. */
+ state->decoder = opus_decoder_create(48000, (int)channels, &error);
if (error == OPUS_OK && state->decoder != NULL) {
- // Creation of memory all ok.
+ /* Creation of memory all ok. */
state->channels = channels;
state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
state->in_dtx_mode = 0;
@@ -384,9 +285,9 @@
return 0;
}
- // If memory allocation was unsuccessful, free the entire state.
+ /* If memory allocation was unsuccessful, free the entire state. */
if (state->decoder) {
- opus_multistream_decoder_destroy(state->decoder);
+ opus_decoder_destroy(state->decoder);
}
free(state);
}
@@ -395,7 +296,7 @@
int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
if (inst) {
- opus_multistream_decoder_destroy(inst->decoder);
+ opus_decoder_destroy(inst->decoder);
free(inst);
return 0;
} else {
@@ -408,7 +309,7 @@
}
void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
- opus_multistream_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+ opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
inst->in_dtx_mode = 0;
}
@@ -423,10 +324,6 @@
// fact a 1-byte TOC with a 1-byte payload. That will be erroneously
// interpreted as comfort noise output, but such a payload is probably
// faulty anyway.
-
- // TODO(webrtc:10218): This is wrong for multistream opus. Then are several
- // single-stream packets glued together with some packet size bytes in
- // between. See https://tools.ietf.org/html/rfc6716#appendix-B
inst->in_dtx_mode = 1;
return 2; // Comfort noise.
} else {
@@ -441,9 +338,8 @@
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 = opus_multistream_decode(
- inst->decoder, encoded, (opus_int32)encoded_bytes,
- (opus_int16*)decoded, frame_size, decode_fec);
+ int res = opus_decode(inst->decoder, encoded, (opus_int32)encoded_bytes,
+ (opus_int16*)decoded, frame_size, decode_fec);
if (res <= 0)
return -1;