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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h" | 11 #include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h" |
12 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h" | 12 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h" |
13 | 13 |
14 #include <assert.h> | |
14 #include <stdlib.h> | 15 #include <stdlib.h> |
15 #include <string.h> | 16 #include <string.h> |
16 | 17 |
17 enum { | 18 enum { |
18 /* Maximum supported frame size in WebRTC is 60 ms. */ | 19 /* Maximum supported frame size in WebRTC is 60 ms. */ |
19 kWebRtcOpusMaxEncodeFrameSizeMs = 60, | 20 kWebRtcOpusMaxEncodeFrameSizeMs = 60, |
20 | 21 |
21 /* The format allows up to 120 ms frames. Since we don't control the other | 22 /* The format allows up to 120 ms frames. Since we don't control the other |
22 * side, we must allow for packets of that size. NetEq is currently limited | 23 * side, we must allow for packets of that size. NetEq is currently limited |
23 * to 60 ms on the receive side. */ | 24 * to 60 ms on the receive side. */ |
24 kWebRtcOpusMaxDecodeFrameSizeMs = 120, | 25 kWebRtcOpusMaxDecodeFrameSizeMs = 120, |
25 | 26 |
26 /* Maximum sample count per channel is 48 kHz * maximum frame size in | 27 /* Maximum sample count per channel is 48 kHz * maximum frame size in |
27 * milliseconds. */ | 28 * milliseconds. */ |
28 kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs, | 29 kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs, |
29 | 30 |
30 /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */ | 31 /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */ |
31 kWebRtcOpusDefaultFrameSize = 960, | 32 kWebRtcOpusDefaultFrameSize = 960, |
33 | |
34 // Maximum number of consecutive zeros, beyond or equal to which DTX can fail. | |
35 kZeroBreakCount = 157, | |
32 }; | 36 }; |
33 | 37 |
34 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, | 38 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, |
35 int32_t channels, | 39 int32_t channels, |
36 int32_t application) { | 40 int32_t application) { |
37 OpusEncInst* state; | 41 if (!inst) |
38 if (inst != NULL) { | 42 return -1; |
39 state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst)); | |
40 if (state) { | |
41 int opus_app; | |
42 switch (application) { | |
43 case 0: { | |
44 opus_app = OPUS_APPLICATION_VOIP; | |
45 break; | |
46 } | |
47 case 1: { | |
48 opus_app = OPUS_APPLICATION_AUDIO; | |
49 break; | |
50 } | |
51 default: { | |
52 free(state); | |
53 return -1; | |
54 } | |
55 } | |
56 | 43 |
57 int error; | 44 int opus_app; |
kwiberg-webrtc
2015/11/01 02:01:55
You're declaring variables not immediately followi
minyue-webrtc
2015/11/06 10:33:17
Done.
| |
58 state->encoder = opus_encoder_create(48000, channels, opus_app, | 45 switch (application) { |
59 &error); | 46 case 0: { |
60 state->in_dtx_mode = 0; | 47 opus_app = OPUS_APPLICATION_VOIP; |
61 if (error == OPUS_OK && state->encoder != NULL) { | 48 break; |
62 *inst = state; | 49 } |
63 return 0; | 50 case 1: { |
64 } | 51 opus_app = OPUS_APPLICATION_AUDIO; |
65 free(state); | 52 break; |
53 } | |
54 default: { | |
55 return -1; | |
66 } | 56 } |
67 } | 57 } |
kwiberg-webrtc
2015/11/01 02:01:55
You don't need the braces for each case in the swi
minyue-webrtc
2015/11/06 10:33:17
I see. this existed before. But I can just change
| |
68 return -1; | 58 |
59 OpusEncInst* state = calloc(1, sizeof(OpusEncInst)); | |
60 assert(state); | |
61 | |
62 // Allocate zero counters. | |
63 state->zero_counts = calloc(channels, sizeof(size_t)); | |
64 assert(state->zero_counts); | |
65 | |
66 int error; | |
67 state->encoder = opus_encoder_create(48000, channels, opus_app, | |
68 &error); | |
69 if (error != OPUS_OK || !state->encoder) { | |
70 WebRtcOpus_EncoderFree(state); | |
71 return -1; | |
72 } | |
73 | |
74 state->in_dtx_mode = 0; | |
75 state->channels = channels; | |
76 | |
77 *inst = state; | |
78 return 0; | |
69 } | 79 } |
70 | 80 |
71 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) { | 81 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) { |
72 if (inst) { | 82 if (inst) { |
73 opus_encoder_destroy(inst->encoder); | 83 opus_encoder_destroy(inst->encoder); |
84 free(inst->zero_counts); | |
74 free(inst); | 85 free(inst); |
75 return 0; | 86 return 0; |
76 } else { | 87 } else { |
77 return -1; | 88 return -1; |
78 } | 89 } |
79 } | 90 } |
80 | 91 |
81 int WebRtcOpus_Encode(OpusEncInst* inst, | 92 int WebRtcOpus_Encode(OpusEncInst* inst, |
82 const int16_t* audio_in, | 93 const int16_t* audio_in, |
83 size_t samples, | 94 size_t samples, |
84 size_t length_encoded_buffer, | 95 size_t length_encoded_buffer, |
85 uint8_t* encoded) { | 96 uint8_t* encoded) { |
86 int res; | 97 int res; |
98 int16_t buffer[2 * 48 * kWebRtcOpusMaxEncodeFrameSizeMs]; | |
87 | 99 |
88 if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) { | 100 if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) { |
89 return -1; | 101 return -1; |
90 } | 102 } |
91 | 103 |
104 const int channels = inst->channels; | |
105 int use_buffer = 0; | |
106 | |
107 // Break long consecutive zeros by forcing a "1" every |kZeroBreakCount| | |
108 // samples. | |
109 if (inst->in_dtx_mode) { | |
110 size_t i; | |
111 int c; | |
112 for (i = 0; i < samples; ++i) { | |
113 for (c = 0; c < channels; ++c) { | |
114 if (audio_in[i * channels + c] == 0) { | |
115 ++inst->zero_counts[c]; | |
116 if (inst->zero_counts[c] == kZeroBreakCount) { | |
117 if (!use_buffer) { | |
118 memcpy(buffer, audio_in, samples * channels * sizeof(int16_t)); | |
119 use_buffer = 1; | |
120 } | |
121 buffer[i * channels + c] = 1; | |
122 inst->zero_counts[c] = 0; | |
123 } | |
124 } else { | |
125 inst->zero_counts[c] = 0; | |
126 } | |
127 } | |
128 } | |
129 } | |
130 | |
131 | |
132 | |
92 res = opus_encode(inst->encoder, | 133 res = opus_encode(inst->encoder, |
93 (const opus_int16*)audio_in, | 134 use_buffer ? buffer : audio_in, |
kwiberg-webrtc
2015/11/01 02:01:55
The loop looks good now. But you have three blank
minyue-webrtc
2015/11/06 10:33:17
:)
| |
94 (int)samples, | 135 (int)samples, |
95 encoded, | 136 encoded, |
96 (opus_int32)length_encoded_buffer); | 137 (opus_int32)length_encoded_buffer); |
97 | 138 |
98 if (res == 1) { | 139 if (res == 1) { |
99 // Indicates DTX since the packet has nothing but a header. In principle, | 140 // Indicates DTX since the packet has nothing but a header. In principle, |
100 // there is no need to send this packet. However, we do transmit the first | 141 // there is no need to send this packet. However, we do transmit the first |
101 // occurrence to let the decoder know that the encoder enters DTX mode. | 142 // occurrence to let the decoder know that the encoder enters DTX mode. |
102 if (inst->in_dtx_mode) { | 143 if (inst->in_dtx_mode) { |
103 return 0; | 144 return 0; |
(...skipping 348 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
452 return 0; | 493 return 0; |
453 } | 494 } |
454 | 495 |
455 for (n = 0; n < channels; n++) { | 496 for (n = 0; n < channels; n++) { |
456 if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1))) | 497 if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1))) |
457 return 1; | 498 return 1; |
458 } | 499 } |
459 | 500 |
460 return 0; | 501 return 0; |
461 } | 502 } |
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