zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Adding DTX to WebRTC Opus wrapper (relanding).

Browse Source 
This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.

See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/

Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org
2014-12-11 16:09:35 +00:00
parent 5f162c8509
commit 0ca768b131
5 changed files with 379 additions and 108 deletions
Download Patch File Download Diff File Expand all files Collapse all files

108
webrtc/modules/audio_coding/codecs/opus/opus_interface.c
View File

@ -43,6 +43,7 @@ int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels) {
state->encoder = opus_encoder_create(48000, channels, application,
&error);
state->in_dtx_mode = 0;
if (error == OPUS_OK && state->encoder != NULL) {
*inst = state;
return 0;
@ -80,9 +81,21 @@ int16_t WebRtcOpus_Encode(OpusEncInst* inst,
encoded,
length_encoded_buffer);
if (res > 0) {
if (res == 1) {
// Indicates DTX since the packet has nothing but a header. In principle,
// there is no need to send this packet. However, we do transmit the first
// occurrence to let the decoder know that the encoder enters DTX mode.
if (inst->in_dtx_mode) {
return 0;
} else {
inst->in_dtx_mode = 1;
return 1;
}
} else if (res > 1) {
inst->in_dtx_mode = 0;
return res;
}
return -1;
}
@ -140,6 +153,22 @@ int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) {
}
}
int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) {
if (inst) {
return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(1));
} else {
return -1;
}
}
int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
if (inst) {
return opus_encoder_ctl(inst->encoder, OPUS_SET_DTX(0));
} else {
return -1;
}
}
int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
if (inst) {
return opus_encoder_ctl(inst->encoder, OPUS_SET_COMPLEXITY(complexity));
@ -165,6 +194,7 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
/* Creation of memory all ok. */
state->channels = channels;
state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
state->in_dtx_mode = 0;
*inst = state;
return 0;
}
@ -195,53 +225,61 @@ int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
int error = opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
if (error == OPUS_OK) {
inst->in_dtx_mode = 0;
return 0;
}
return -1;
}
/* For decoder to determine if it is to output speech or comfort noise. */
static int16_t DetermineAudioType(OpusDecInst* inst, int16_t encoded_bytes) {
// Audio type becomes comfort noise if |encoded_byte| is 1 and keeps
// to be so if the following |encoded_byte| are 0 or 1.
if (encoded_bytes == 0 && inst->in_dtx_mode) {
return 2; // Comfort noise.
} else if (encoded_bytes == 1) {
inst->in_dtx_mode = 1;
return 2; // Comfort noise.
} else {
inst->in_dtx_mode = 0;
return 0; // Speech.
}
}
/* |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(OpusDecoder* inst, const uint8_t* encoded,
static int DecodeNative(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) {
int res = opus_decode(
inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 0);
int16_t* decoded, int16_t* audio_type, int decode_fec) {
int res = opus_decode(inst->decoder, encoded, encoded_bytes,
(opus_int16*)decoded, frame_size, decode_fec);
/* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0;
if (res <= 0)
return -1;
if (res > 0) {
return res;
}
return -1;
}
*audio_type = DetermineAudioType(inst, encoded_bytes);
static int DecodeFec(OpusDecoder* inst, const uint8_t* encoded,
int16_t encoded_bytes, int frame_size,
int16_t* decoded, int16_t* audio_type) {
int res = opus_decode(
inst, encoded, encoded_bytes, (opus_int16*)decoded, frame_size, 1);
/* TODO(tlegrand): set to DTX for zero-length packets? */
*audio_type = 0;
if (res > 0) {
return res;
}
return -1;
return res;
}
int16_t WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
int16_t encoded_bytes, int16_t* decoded,
int16_t* audio_type) {
int decoded_samples = DecodeNative(inst->decoder,
encoded,
encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel,
decoded,
audio_type);
int decoded_samples;
if (encoded_bytes == 0) {
*audio_type = DetermineAudioType(inst, encoded_bytes);
decoded_samples = WebRtcOpus_DecodePlc(inst, decoded, 1);
} else {
decoded_samples = DecodeNative(inst,
encoded,
encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel,
decoded,
audio_type,
0);
}
if (decoded_samples < 0) {
return -1;
}
@ -264,8 +302,8 @@ int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
plc_samples = number_of_lost_frames * inst->prev_decoded_samples;
plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
decoded_samples = DecodeNative(inst->decoder, NULL, 0, plc_samples,
decoded, &audio_type);
decoded_samples = DecodeNative(inst, NULL, 0, plc_samples,
decoded, &audio_type, 0);
if (decoded_samples < 0) {
return -1;
}
@ -285,8 +323,8 @@ int16_t WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
fec_samples = opus_packet_get_samples_per_frame(encoded, 48000);
decoded_samples = DecodeFec(inst->decoder, encoded, encoded_bytes,
fec_samples, decoded, audio_type);
decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
fec_samples, decoded, audio_type, 1);
if (decoded_samples < 0) {
return -1;
}