Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 60ab669c4c545b328b5c8b0453eb2cdecf851651
This commit was generated by merge_from_chromium.py.
Change-Id: I7efe0f9a490c88eb3ec522902a351f9662e8753e
diff --git a/base/helpers.cc b/base/helpers.cc
index 8b14cdf..84d1c93 100644
--- a/base/helpers.cc
+++ b/base/helpers.cc
@@ -47,36 +47,17 @@
};
#if defined(SSL_USE_OPENSSL)
-// The OpenSSL RNG. Need to make sure it doesn't run out of entropy.
+// The OpenSSL RNG.
class SecureRandomGenerator : public RandomGenerator {
public:
- SecureRandomGenerator() : inited_(false) {
- }
- ~SecureRandomGenerator() {
- }
+ SecureRandomGenerator() {}
+ ~SecureRandomGenerator() {}
virtual bool Init(const void* seed, size_t len) {
- // By default, seed from the system state.
- if (!inited_) {
- if (RAND_poll() <= 0) {
- return false;
- }
- inited_ = true;
- }
- // Allow app data to be mixed in, if provided.
- if (seed) {
- RAND_seed(seed, len);
- }
return true;
}
virtual bool Generate(void* buf, size_t len) {
- if (!inited_ && !Init(NULL, 0)) {
- return false;
- }
return (RAND_bytes(reinterpret_cast<unsigned char*>(buf), len) > 0);
}
-
- private:
- bool inited_;
};
#elif defined(SSL_USE_NSS_RNG)
diff --git a/base/openssladapter.cc b/base/openssladapter.cc
index 68a1fcb..feb01d3 100644
--- a/base/openssladapter.cc
+++ b/base/openssladapter.cc
@@ -34,6 +34,7 @@
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/openssl.h"
+#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/sslroots.h"
#include "webrtc/base/stringutils.h"
@@ -141,7 +142,7 @@
}
static int socket_puts(BIO* b, const char* str) {
- return socket_write(b, str, strlen(str));
+ return socket_write(b, str, rtc::checked_cast<int>(strlen(str)));
}
static long socket_ctrl(BIO* b, int cmd, long num, void* ptr) {
@@ -448,7 +449,7 @@
ssl_write_needs_read_ = false;
- int code = SSL_write(ssl_, pv, cb);
+ int code = SSL_write(ssl_, pv, checked_cast<int>(cb));
switch (SSL_get_error(ssl_, code)) {
case SSL_ERROR_NONE:
//LOG(LS_INFO) << " -- success";
@@ -503,7 +504,7 @@
ssl_read_needs_write_ = false;
- int code = SSL_read(ssl_, pv, cb);
+ int code = SSL_read(ssl_, pv, checked_cast<int>(cb));
switch (SSL_get_error(ssl_, code)) {
case SSL_ERROR_NONE:
//LOG(LS_INFO) << " -- success";
@@ -843,7 +844,8 @@
for (int i = 0; i < ARRAY_SIZE(kSSLCertCertificateList); i++) {
const unsigned char* cert_buffer = kSSLCertCertificateList[i];
size_t cert_buffer_len = kSSLCertCertificateSizeList[i];
- X509* cert = d2i_X509(NULL, &cert_buffer, cert_buffer_len);
+ X509* cert = d2i_X509(NULL, &cert_buffer,
+ checked_cast<long>(cert_buffer_len));
if (cert) {
int return_value = X509_STORE_add_cert(SSL_CTX_get_cert_store(ctx), cert);
if (return_value == 0) {
diff --git a/base/opensslstreamadapter.cc b/base/opensslstreamadapter.cc
index 133eb72..d790e4e 100644
--- a/base/opensslstreamadapter.cc
+++ b/base/opensslstreamadapter.cc
@@ -26,6 +26,7 @@
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/stream.h"
#include "webrtc/base/openssl.h"
#include "webrtc/base/openssladapter.h"
@@ -114,7 +115,7 @@
int error;
StreamResult result = stream->Read(out, outl, &read, &error);
if (result == SR_SUCCESS) {
- return read;
+ return checked_cast<int>(read);
} else if (result == SR_EOS) {
b->num = 1;
} else if (result == SR_BLOCK) {
@@ -132,7 +133,7 @@
int error;
StreamResult result = stream->Write(in, inl, &written, &error);
if (result == SR_SUCCESS) {
- return written;
+ return checked_cast<int>(written);
} else if (result == SR_BLOCK) {
BIO_set_retry_write(b);
}
@@ -140,7 +141,7 @@
}
static int stream_puts(BIO* b, const char* str) {
- return stream_write(b, str, strlen(str));
+ return stream_write(b, str, checked_cast<int>(strlen(str)));
}
static long stream_ctrl(BIO* b, int cmd, long num, void* ptr) {
@@ -364,7 +365,7 @@
ssl_write_needs_read_ = false;
- int code = SSL_write(ssl_, data, data_len);
+ int code = SSL_write(ssl_, data, checked_cast<int>(data_len));
int ssl_error = SSL_get_error(ssl_, code);
switch (ssl_error) {
case SSL_ERROR_NONE:
@@ -425,7 +426,7 @@
ssl_read_needs_write_ = false;
- int code = SSL_read(ssl_, data, data_len);
+ int code = SSL_read(ssl_, data, checked_cast<int>(data_len));
int ssl_error = SSL_get_error(ssl_, code);
switch (ssl_error) {
case SSL_ERROR_NONE:
diff --git a/base/safe_conversions_impl.h b/base/safe_conversions_impl.h
index 2950f97..77b053a 100644
--- a/base/safe_conversions_impl.h
+++ b/base/safe_conversions_impl.h
@@ -15,6 +15,8 @@
#include <limits>
+#include "webrtc/base/compile_assert.h"
+
namespace rtc {
namespace internal {
diff --git a/config.cc b/config.cc
index 70bd870..357f636 100644
--- a/config.cc
+++ b/config.cc
@@ -39,13 +39,13 @@
ss << ", max_bitrate_bps:" << max_bitrate_bps;
ss << ", max_qp: " << max_qp;
- ss << ", temporal_layer_thresholds_bps: {";
+ ss << ", temporal_layer_thresholds_bps: [";
for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
ss << temporal_layer_thresholds_bps[i];
if (i != temporal_layer_thresholds_bps.size() - 1)
- ss << "}, {";
+ ss << ", ";
}
- ss << '}';
+ ss << ']';
ss << '}';
return ss.str();
@@ -54,13 +54,13 @@
std::string VideoEncoderConfig::ToString() const {
std::stringstream ss;
- ss << "{streams: {";
+ ss << "{streams: [";
for (size_t i = 0; i < streams.size(); ++i) {
ss << streams[i].ToString();
if (i != streams.size() - 1)
- ss << "}, {";
+ ss << ", ";
}
- ss << '}';
+ ss << ']';
ss << ", content_type: ";
switch (content_type) {
case kRealtimeVideo:
diff --git a/modules/audio_coding/main/acm2/acm_isac.cc b/modules/audio_coding/main/acm2/acm_isac.cc
index bc20c96..8fa96e5 100644
--- a/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/modules/audio_coding/main/acm2/acm_isac.cc
@@ -277,7 +277,6 @@
return;
}
codec_inst_ptr_->inst = NULL;
- state_ = codec_inst_ptr_;
}
ACMISAC::~ACMISAC() {
diff --git a/modules/audio_coding/neteq/audio_decoder.cc b/modules/audio_coding/neteq/audio_decoder.cc
index 04a74ee..d5a2762 100644
--- a/modules/audio_coding/neteq/audio_decoder.cc
+++ b/modules/audio_coding/neteq/audio_decoder.cc
@@ -12,6 +12,7 @@
#include <assert.h>
+#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
namespace webrtc {
@@ -51,6 +52,11 @@
return false;
}
+CNG_dec_inst* AudioDecoder::CngDecoderInstance() {
+ FATAL() << "Not a CNG decoder";
+ return NULL;
+}
+
bool AudioDecoder::CodecSupported(NetEqDecoder codec_type) {
switch (codec_type) {
case kDecoderPCMu:
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.cc b/modules/audio_coding/neteq/audio_decoder_impl.cc
index 07b1b4b..eb07823 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -103,17 +103,17 @@
// iLBC
#ifdef WEBRTC_CODEC_ILBC
AudioDecoderIlbc::AudioDecoderIlbc() {
- WebRtcIlbcfix_DecoderCreate(reinterpret_cast<iLBC_decinst_t**>(&state_));
+ WebRtcIlbcfix_DecoderCreate(&dec_state_);
}
AudioDecoderIlbc::~AudioDecoderIlbc() {
- WebRtcIlbcfix_DecoderFree(static_cast<iLBC_decinst_t*>(state_));
+ WebRtcIlbcfix_DecoderFree(dec_state_);
}
int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcIlbcfix_Decode(static_cast<iLBC_decinst_t*>(state_),
+ int16_t ret = WebRtcIlbcfix_Decode(dec_state_,
reinterpret_cast<const int16_t*>(encoded),
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
@@ -122,12 +122,11 @@
}
int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) {
- return WebRtcIlbcfix_NetEqPlc(static_cast<iLBC_decinst_t*>(state_),
- decoded, num_frames);
+ return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
}
int AudioDecoderIlbc::Init() {
- return WebRtcIlbcfix_Decoderinit30Ms(static_cast<iLBC_decinst_t*>(state_));
+ return WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
}
#endif
@@ -135,19 +134,18 @@
#ifdef WEBRTC_CODEC_ISAC
AudioDecoderIsac::AudioDecoderIsac(int decode_sample_rate_hz) {
DCHECK(decode_sample_rate_hz == 16000 || decode_sample_rate_hz == 32000);
- WebRtcIsac_Create(reinterpret_cast<ISACStruct**>(&state_));
- WebRtcIsac_SetDecSampRate(static_cast<ISACStruct*>(state_),
- decode_sample_rate_hz);
+ WebRtcIsac_Create(&isac_state_);
+ WebRtcIsac_SetDecSampRate(isac_state_, decode_sample_rate_hz);
}
AudioDecoderIsac::~AudioDecoderIsac() {
- WebRtcIsac_Free(static_cast<ISACStruct*>(state_));
+ WebRtcIsac_Free(isac_state_);
}
int AudioDecoderIsac::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcIsac_Decode(static_cast<ISACStruct*>(state_),
+ int16_t ret = WebRtcIsac_Decode(isac_state_,
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
@@ -159,7 +157,7 @@
size_t encoded_len, int16_t* decoded,
SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcIsac_DecodeRcu(static_cast<ISACStruct*>(state_),
+ int16_t ret = WebRtcIsac_DecodeRcu(isac_state_,
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
@@ -168,12 +166,11 @@
}
int AudioDecoderIsac::DecodePlc(int num_frames, int16_t* decoded) {
- return WebRtcIsac_DecodePlc(static_cast<ISACStruct*>(state_),
- decoded, num_frames);
+ return WebRtcIsac_DecodePlc(isac_state_, decoded, num_frames);
}
int AudioDecoderIsac::Init() {
- return WebRtcIsac_DecoderInit(static_cast<ISACStruct*>(state_));
+ return WebRtcIsac_DecoderInit(isac_state_);
}
int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
@@ -181,7 +178,7 @@
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
- return WebRtcIsac_UpdateBwEstimate(static_cast<ISACStruct*>(state_),
+ return WebRtcIsac_UpdateBwEstimate(isac_state_,
payload,
static_cast<int32_t>(payload_len),
rtp_sequence_number,
@@ -190,24 +187,24 @@
}
int AudioDecoderIsac::ErrorCode() {
- return WebRtcIsac_GetErrorCode(static_cast<ISACStruct*>(state_));
+ return WebRtcIsac_GetErrorCode(isac_state_);
}
#endif
// iSAC fix
#ifdef WEBRTC_CODEC_ISACFX
AudioDecoderIsacFix::AudioDecoderIsacFix() {
- WebRtcIsacfix_Create(reinterpret_cast<ISACFIX_MainStruct**>(&state_));
+ WebRtcIsacfix_Create(&isac_state_);
}
AudioDecoderIsacFix::~AudioDecoderIsacFix() {
- WebRtcIsacfix_Free(static_cast<ISACFIX_MainStruct*>(state_));
+ WebRtcIsacfix_Free(isac_state_);
}
int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcIsacfix_Decode(static_cast<ISACFIX_MainStruct*>(state_),
+ int16_t ret = WebRtcIsacfix_Decode(isac_state_,
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
@@ -216,7 +213,7 @@
}
int AudioDecoderIsacFix::Init() {
- return WebRtcIsacfix_DecoderInit(static_cast<ISACFIX_MainStruct*>(state_));
+ return WebRtcIsacfix_DecoderInit(isac_state_);
}
int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload,
@@ -225,32 +222,32 @@
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
return WebRtcIsacfix_UpdateBwEstimate(
- static_cast<ISACFIX_MainStruct*>(state_),
+ isac_state_,
payload,
static_cast<int32_t>(payload_len),
rtp_sequence_number, rtp_timestamp, arrival_timestamp);
}
int AudioDecoderIsacFix::ErrorCode() {
- return WebRtcIsacfix_GetErrorCode(static_cast<ISACFIX_MainStruct*>(state_));
+ return WebRtcIsacfix_GetErrorCode(isac_state_);
}
#endif
// G.722
#ifdef WEBRTC_CODEC_G722
AudioDecoderG722::AudioDecoderG722() {
- WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_));
+ WebRtcG722_CreateDecoder(&dec_state_);
}
AudioDecoderG722::~AudioDecoderG722() {
- WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_));
+ WebRtcG722_FreeDecoder(dec_state_);
}
int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcG722_Decode(
- static_cast<G722DecInst*>(state_),
+ dec_state_,
const_cast<int16_t*>(reinterpret_cast<const int16_t*>(encoded)),
static_cast<int16_t>(encoded_len), decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
@@ -258,7 +255,7 @@
}
int AudioDecoderG722::Init() {
- return WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_));
+ return WebRtcG722_DecoderInit(dec_state_);
}
int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
@@ -267,18 +264,15 @@
return static_cast<int>(2 * encoded_len / channels_);
}
-AudioDecoderG722Stereo::AudioDecoderG722Stereo()
- : AudioDecoderG722(),
- state_left_(state_), // Base member |state_| is used for left channel.
- state_right_(NULL) {
+AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
channels_ = 2;
- // |state_left_| already created by the base class AudioDecoderG722.
- WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_right_));
+ WebRtcG722_CreateDecoder(&dec_state_left_);
+ WebRtcG722_CreateDecoder(&dec_state_right_);
}
AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
- // |state_left_| will be freed by the base class AudioDecoderG722.
- WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_right_));
+ WebRtcG722_FreeDecoder(dec_state_left_);
+ WebRtcG722_FreeDecoder(dec_state_right_);
}
int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
@@ -289,13 +283,13 @@
SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
// Decode left and right.
int16_t ret = WebRtcG722_Decode(
- static_cast<G722DecInst*>(state_left_),
+ dec_state_left_,
reinterpret_cast<int16_t*>(encoded_deinterleaved),
static_cast<int16_t>(encoded_len / 2), decoded, &temp_type);
if (ret >= 0) {
int decoded_len = ret;
ret = WebRtcG722_Decode(
- static_cast<G722DecInst*>(state_right_),
+ dec_state_right_,
reinterpret_cast<int16_t*>(&encoded_deinterleaved[encoded_len / 2]),
static_cast<int16_t>(encoded_len / 2), &decoded[decoded_len], &temp_type);
if (ret == decoded_len) {
@@ -317,11 +311,10 @@
}
int AudioDecoderG722Stereo::Init() {
- int ret = WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_right_));
- if (ret != 0) {
- return ret;
- }
- return AudioDecoderG722::Init();
+ int r = WebRtcG722_DecoderInit(dec_state_left_);
+ if (r != 0)
+ return r;
+ return WebRtcG722_DecoderInit(dec_state_right_);
}
// Split the stereo packet and place left and right channel after each other
@@ -401,18 +394,17 @@
AudioDecoderOpus::AudioDecoderOpus(int num_channels) {
DCHECK(num_channels == 1 || num_channels == 2);
channels_ = num_channels;
- WebRtcOpus_DecoderCreate(reinterpret_cast<OpusDecInst**>(&state_),
- static_cast<int>(channels_));
+ WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_));
}
AudioDecoderOpus::~AudioDecoderOpus() {
- WebRtcOpus_DecoderFree(static_cast<OpusDecInst*>(state_));
+ WebRtcOpus_DecoderFree(dec_state_);
}
int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcOpus_DecodeNew(static_cast<OpusDecInst*>(state_), encoded,
+ int16_t ret = WebRtcOpus_DecodeNew(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
if (ret > 0)
@@ -425,7 +417,7 @@
size_t encoded_len, int16_t* decoded,
SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
- int16_t ret = WebRtcOpus_DecodeFec(static_cast<OpusDecInst*>(state_), encoded,
+ int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
if (ret > 0)
@@ -435,12 +427,12 @@
}
int AudioDecoderOpus::Init() {
- return WebRtcOpus_DecoderInitNew(static_cast<OpusDecInst*>(state_));
+ return WebRtcOpus_DecoderInitNew(dec_state_);
}
int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
size_t encoded_len) {
- return WebRtcOpus_DurationEst(static_cast<OpusDecInst*>(state_),
+ return WebRtcOpus_DurationEst(dec_state_,
encoded, static_cast<int>(encoded_len));
}
@@ -458,19 +450,15 @@
#endif
AudioDecoderCng::AudioDecoderCng() {
- WebRtcCng_CreateDec(reinterpret_cast<CNG_dec_inst**>(&state_));
- assert(state_);
+ CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
}
AudioDecoderCng::~AudioDecoderCng() {
- if (state_) {
- WebRtcCng_FreeDec(static_cast<CNG_dec_inst*>(state_));
- }
+ WebRtcCng_FreeDec(dec_state_);
}
int AudioDecoderCng::Init() {
- assert(state_);
- return WebRtcCng_InitDec(static_cast<CNG_dec_inst*>(state_));
+ return WebRtcCng_InitDec(dec_state_);
}
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.h b/modules/audio_coding/neteq/audio_decoder_impl.h
index 214392e..b30331f 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -19,6 +19,22 @@
#include "webrtc/engine_configurations.h"
#endif
#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
+#ifdef WEBRTC_CODEC_G722
+#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
+#endif
+#ifdef WEBRTC_CODEC_ISACFX
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#endif
#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
#include "webrtc/typedefs.h"
@@ -109,6 +125,7 @@
virtual int Init();
private:
+ iLBC_decinst_t* dec_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbc);
};
#endif
@@ -133,6 +150,7 @@
virtual int ErrorCode();
private:
+ ISACStruct* isac_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsac);
};
#endif
@@ -153,6 +171,7 @@
virtual int ErrorCode();
private:
+ ISACFIX_MainStruct* isac_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacFix);
};
#endif
@@ -169,10 +188,11 @@
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len);
private:
+ G722DecInst* dec_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722);
};
-class AudioDecoderG722Stereo : public AudioDecoderG722 {
+class AudioDecoderG722Stereo : public AudioDecoder {
public:
AudioDecoderG722Stereo();
virtual ~AudioDecoderG722Stereo();
@@ -189,8 +209,8 @@
void SplitStereoPacket(const uint8_t* encoded, size_t encoded_len,
uint8_t* encoded_deinterleaved);
- void* const state_left_;
- void* state_right_;
+ G722DecInst* dec_state_left_;
+ G722DecInst* dec_state_right_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Stereo);
};
@@ -229,6 +249,7 @@
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
private:
+ OpusDecInst* dec_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpus);
};
#endif
@@ -252,7 +273,10 @@
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) { return -1; }
+ virtual CNG_dec_inst* CngDecoderInstance() OVERRIDE { return dec_state_; }
+
private:
+ CNG_dec_inst* dec_state_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoderCng);
};
diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc
index 31bb40c..e2be066 100644
--- a/modules/audio_coding/neteq/comfort_noise.cc
+++ b/modules/audio_coding/neteq/comfort_noise.cc
@@ -36,7 +36,7 @@
return kUnknownPayloadType;
}
decoder_database_->SetActiveCngDecoder(packet->header.payloadType);
- CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
+ CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
int16_t ret = WebRtcCng_UpdateSid(cng_inst,
packet->payload,
packet->payload_length);
@@ -72,7 +72,7 @@
if (!cng_decoder) {
return kUnknownPayloadType;
}
- CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
+ CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
// The expression &(*output)[0][0] is a pointer to the first element in
// the first channel.
if (WebRtcCng_Generate(cng_inst, &(*output)[0][0],
diff --git a/modules/audio_coding/neteq/interface/audio_decoder.h b/modules/audio_coding/neteq/interface/audio_decoder.h
index 16d78c9..be85c4d 100644
--- a/modules/audio_coding/neteq/interface/audio_decoder.h
+++ b/modules/audio_coding/neteq/interface/audio_decoder.h
@@ -14,6 +14,7 @@
#include <stdlib.h> // NULL
#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -63,7 +64,7 @@
// Used by PacketDuration below. Save the value -1 for errors.
enum { kNotImplemented = -2 };
- AudioDecoder() : channels_(1), state_(NULL) {}
+ AudioDecoder() : channels_(1) {}
virtual ~AudioDecoder() {}
// Decodes |encode_len| bytes from |encoded| and writes the result in
@@ -114,8 +115,9 @@
// Returns true if the packet has FEC and false otherwise.
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
- // Returns the underlying decoder state.
- void* state() { return state_; }
+ // If this is a CNG decoder, return the underlying CNG_dec_inst*. If this
+ // isn't a CNG decoder, don't call this method.
+ virtual CNG_dec_inst* CngDecoderInstance();
// Returns true if |codec_type| is supported.
static bool CodecSupported(NetEqDecoder codec_type);
@@ -134,7 +136,6 @@
static SpeechType ConvertSpeechType(int16_t type);
size_t channels_;
- void* state_;
private:
DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 46d03fb..ca2c1ee 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -147,9 +147,9 @@
AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder) {
- CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
// Generate long enough for 32kHz.
- if (WebRtcCng_Generate(cng_inst, cng_output, kCngLength, 0) < 0) {
+ if (WebRtcCng_Generate(cng_decoder->CngDecoderInstance(), cng_output,
+ kCngLength, 0) < 0) {
// Error returned; set return vector to all zeros.
memset(cng_output, 0, sizeof(cng_output));
}
diff --git a/modules/audio_device/android/opensles_input.cc b/modules/audio_device/android/opensles_input.cc
index f22d8bf..e68a6aa 100644
--- a/modules/audio_device/android/opensles_input.cc
+++ b/modules/audio_device/android/opensles_input.cc
@@ -360,6 +360,24 @@
req),
false);
+ SLAndroidConfigurationItf recorder_config;
+ OPENSL_RETURN_ON_FAILURE(
+ (*sles_recorder_)->GetInterface(sles_recorder_,
+ SL_IID_ANDROIDCONFIGURATION,
+ &recorder_config),
+ false);
+
+ // Set audio recorder configuration to
+ // SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION which ensures that we
+ // use the main microphone tuned for audio communications.
+ SLint32 stream_type = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
+ OPENSL_RETURN_ON_FAILURE(
+ (*recorder_config)->SetConfiguration(recorder_config,
+ SL_ANDROID_KEY_RECORDING_PRESET,
+ &stream_type,
+ sizeof(SLint32)),
+ false);
+
// Realize the recorder in synchronous mode.
OPENSL_RETURN_ON_FAILURE((*sles_recorder_)->Realize(sles_recorder_,
SL_BOOLEAN_FALSE),
diff --git a/modules/audio_device/android/opensles_output.cc b/modules/audio_device/android/opensles_output.cc
index 377789b..487e284 100644
--- a/modules/audio_device/android/opensles_output.cc
+++ b/modules/audio_device/android/opensles_output.cc
@@ -407,6 +407,24 @@
&audio_source, &audio_sink,
kNumInterfaces, ids, req),
false);
+
+ SLAndroidConfigurationItf player_config;
+ OPENSL_RETURN_ON_FAILURE(
+ (*sles_player_)->GetInterface(sles_player_,
+ SL_IID_ANDROIDCONFIGURATION,
+ &player_config),
+ false);
+
+ // Set audio player configuration to SL_ANDROID_STREAM_VOICE which corresponds
+ // to android.media.AudioManager.STREAM_VOICE_CALL.
+ SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
+ OPENSL_RETURN_ON_FAILURE(
+ (*player_config)->SetConfiguration(player_config,
+ SL_ANDROID_KEY_STREAM_TYPE,
+ &stream_type,
+ sizeof(SLint32)),
+ false);
+
// Realize the player in synchronous mode.
OPENSL_RETURN_ON_FAILURE((*sles_player_)->Realize(sles_player_,
SL_BOOLEAN_FALSE),
diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h
index faef7a0..18225f9 100644
--- a/modules/rtp_rtcp/source/rtp_format.h
+++ b/modules/rtp_rtcp/source/rtp_format.h
@@ -53,12 +53,10 @@
class RtpDepacketizer {
public:
struct ParsedPayload {
- explicit ParsedPayload(WebRtcRTPHeader* rtp_header)
- : payload(NULL), payload_length(0), header(rtp_header) {}
-
const uint8_t* payload;
size_t payload_length;
- WebRtcRTPHeader* header;
+ FrameType frame_type;
+ RTPTypeHeader type;
};
static RtpDepacketizer* Create(RtpVideoCodecTypes type);
diff --git a/modules/rtp_rtcp/source/rtp_format_h264.cc b/modules/rtp_rtcp/source/rtp_format_h264.cc
index b6af1ad..0d20b30 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -37,12 +37,15 @@
// Bit masks for FU (A and B) headers.
enum FuDefs { kSBit = 0x80, kEBit = 0x40, kRBit = 0x20 };
-void ParseSingleNalu(WebRtcRTPHeader* rtp_header,
+void ParseSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) {
- rtp_header->type.Video.codec = kRtpVideoH264;
- rtp_header->type.Video.isFirstPacket = true;
- RTPVideoHeaderH264* h264_header = &rtp_header->type.Video.codecHeader.H264;
+ parsed_payload->type.Video.width = 0;
+ parsed_payload->type.Video.height = 0;
+ parsed_payload->type.Video.codec = kRtpVideoH264;
+ parsed_payload->type.Video.isFirstPacket = true;
+ RTPVideoHeaderH264* h264_header =
+ &parsed_payload->type.Video.codecHeader.H264;
h264_header->single_nalu = true;
h264_header->stap_a = false;
@@ -56,15 +59,15 @@
case kSps:
case kPps:
case kIdr:
- rtp_header->frameType = kVideoFrameKey;
+ parsed_payload->frame_type = kVideoFrameKey;
break;
default:
- rtp_header->frameType = kVideoFrameDelta;
+ parsed_payload->frame_type = kVideoFrameDelta;
break;
}
}
-void ParseFuaNalu(WebRtcRTPHeader* rtp_header,
+void ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length,
size_t* offset) {
@@ -82,13 +85,16 @@
}
if (original_nal_type == kIdr) {
- rtp_header->frameType = kVideoFrameKey;
+ parsed_payload->frame_type = kVideoFrameKey;
} else {
- rtp_header->frameType = kVideoFrameDelta;
+ parsed_payload->frame_type = kVideoFrameDelta;
}
- rtp_header->type.Video.codec = kRtpVideoH264;
- rtp_header->type.Video.isFirstPacket = first_fragment;
- RTPVideoHeaderH264* h264_header = &rtp_header->type.Video.codecHeader.H264;
+ parsed_payload->type.Video.width = 0;
+ parsed_payload->type.Video.height = 0;
+ parsed_payload->type.Video.codec = kRtpVideoH264;
+ parsed_payload->type.Video.isFirstPacket = first_fragment;
+ RTPVideoHeaderH264* h264_header =
+ &parsed_payload->type.Video.codecHeader.H264;
h264_header->single_nalu = false;
h264_header->stap_a = false;
}
@@ -298,12 +304,11 @@
size_t offset = 0;
if (nal_type == kFuA) {
// Fragmented NAL units (FU-A).
- ParseFuaNalu(
- parsed_payload->header, payload_data, payload_data_length, &offset);
+ ParseFuaNalu(parsed_payload, payload_data, payload_data_length, &offset);
} else {
// We handle STAP-A and single NALU's the same way here. The jitter buffer
// will depacketize the STAP-A into NAL units later.
- ParseSingleNalu(parsed_payload->header, payload_data, payload_data_length);
+ ParseSingleNalu(parsed_payload, payload_data, payload_data_length);
}
parsed_payload->payload = payload_data + offset;
diff --git a/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index fb29b5a..eb690ea 100644
--- a/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -399,17 +399,15 @@
TEST_F(RtpDepacketizerH264Test, TestSingleNalu) {
uint8_t packet[2] = {0x05, 0xFF}; // F=0, NRI=0, Type=5.
-
- WebRtcRTPHeader expected_header;
- memset(&expected_header, 0, sizeof(expected_header));
- RtpDepacketizer::ParsedPayload payload(&expected_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(&payload, packet, sizeof(packet));
- EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
- EXPECT_TRUE(payload.header->type.Video.isFirstPacket);
- EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.single_nalu);
- EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.stap_a);
+ EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+ EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+ EXPECT_TRUE(payload.type.Video.isFirstPacket);
+ EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
+ EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
}
TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
@@ -417,17 +415,15 @@
// Length, nal header, payload.
0, 0x02, kIdr, 0xFF, 0, 0x03, kIdr, 0xFF,
0x00, 0, 0x04, kIdr, 0xFF, 0x00, 0x11};
-
- WebRtcRTPHeader expected_header;
- memset(&expected_header, 0, sizeof(expected_header));
- RtpDepacketizer::ParsedPayload payload(&expected_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(&payload, packet, sizeof(packet));
- EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
- EXPECT_TRUE(payload.header->type.Video.isFirstPacket);
- EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.single_nalu);
- EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.stap_a);
+ EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+ EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+ EXPECT_TRUE(payload.type.Video.isFirstPacket);
+ EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
+ EXPECT_TRUE(payload.type.Video.codecHeader.H264.stap_a);
}
TEST_F(RtpDepacketizerH264Test, TestStapADelta) {
@@ -435,17 +431,15 @@
// Length, nal header, payload.
0, 0x02, kSlice, 0xFF, 0, 0x03, kSlice, 0xFF,
0x00, 0, 0x04, kSlice, 0xFF, 0x00, 0x11};
-
- WebRtcRTPHeader expected_header;
- memset(&expected_header, 0, sizeof(expected_header));
- RtpDepacketizer::ParsedPayload payload(&expected_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(&payload, packet, sizeof(packet));
- EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
- EXPECT_TRUE(payload.header->type.Video.isFirstPacket);
- EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.single_nalu);
- EXPECT_TRUE(payload.header->type.Video.codecHeader.H264.stap_a);
+ EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+ EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+ EXPECT_TRUE(payload.type.Video.isFirstPacket);
+ EXPECT_TRUE(payload.type.Video.codecHeader.H264.single_nalu);
+ EXPECT_TRUE(payload.type.Video.codecHeader.H264.stap_a);
}
TEST_F(RtpDepacketizerH264Test, TestFuA) {
@@ -470,33 +464,36 @@
};
const uint8_t kExpected3[1] = {0x03};
- WebRtcRTPHeader expected_header;
- memset(&expected_header, 0, sizeof(expected_header));
- RtpDepacketizer::ParsedPayload payload(&expected_header);
+ RtpDepacketizer::ParsedPayload payload;
// We expect that the first packet is one byte shorter since the FU-A header
// has been replaced by the original nal header.
ASSERT_TRUE(depacketizer_->Parse(&payload, packet1, sizeof(packet1)));
ExpectPacket(&payload, kExpected1, sizeof(kExpected1));
- EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
- EXPECT_TRUE(payload.header->type.Video.isFirstPacket);
- EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.single_nalu);
- EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.stap_a);
+ EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+ EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+ EXPECT_TRUE(payload.type.Video.isFirstPacket);
+ EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
+ EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
// Following packets will be 2 bytes shorter since they will only be appended
// onto the first packet.
+ payload = RtpDepacketizer::ParsedPayload();
ASSERT_TRUE(depacketizer_->Parse(&payload, packet2, sizeof(packet2)));
ExpectPacket(&payload, kExpected2, sizeof(kExpected2));
- EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
- EXPECT_FALSE(payload.header->type.Video.isFirstPacket);
- EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.single_nalu);
- EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.stap_a);
+ EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+ EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+ EXPECT_FALSE(payload.type.Video.isFirstPacket);
+ EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
+ EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
+ payload = RtpDepacketizer::ParsedPayload();
ASSERT_TRUE(depacketizer_->Parse(&payload, packet3, sizeof(packet3)));
ExpectPacket(&payload, kExpected3, sizeof(kExpected3));
- EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
- EXPECT_FALSE(payload.header->type.Video.isFirstPacket);
- EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.single_nalu);
- EXPECT_FALSE(payload.header->type.Video.codecHeader.H264.stap_a);
+ EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+ EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
+ EXPECT_FALSE(payload.type.Video.isFirstPacket);
+ EXPECT_FALSE(payload.type.Video.codecHeader.H264.single_nalu);
+ EXPECT_FALSE(payload.type.Video.codecHeader.H264.stap_a);
}
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index 4907846..ab210ec 100644
--- a/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -90,17 +90,19 @@
const uint8_t* payload_data,
size_t payload_data_length) {
assert(parsed_payload != NULL);
- assert(parsed_payload->header != NULL);
uint8_t generic_header = *payload_data++;
--payload_data_length;
- parsed_payload->header->frameType =
+ parsed_payload->frame_type =
((generic_header & RtpFormatVideoGeneric::kKeyFrameBit) != 0)
? kVideoFrameKey
: kVideoFrameDelta;
- parsed_payload->header->type.Video.isFirstPacket =
+ parsed_payload->type.Video.isFirstPacket =
(generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0;
+ parsed_payload->type.Video.codec = kRtpVideoGeneric;
+ parsed_payload->type.Video.width = 0;
+ parsed_payload->type.Video.height = 0;
parsed_payload->payload = payload_data;
parsed_payload->payload_length = payload_data_length;
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8.cc b/modules/rtp_rtcp/source/rtp_format_vp8.cc
index 86bdd8b..d74e04f 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp8.cc
@@ -121,11 +121,11 @@
return parsed_bytes;
}
-int ParseVP8FrameSize(WebRtcRTPHeader* rtp_header,
+int ParseVP8FrameSize(RtpDepacketizer::ParsedPayload* parsed_payload,
const uint8_t* data,
int data_length) {
- assert(rtp_header != NULL);
- if (rtp_header->frameType != kVideoFrameKey) {
+ assert(parsed_payload != NULL);
+ if (parsed_payload->frame_type != kVideoFrameKey) {
// Included in payload header for I-frames.
return 0;
}
@@ -134,8 +134,8 @@
// in the beginning of the partition.
return -1;
}
- rtp_header->type.Video.width = ((data[7] << 8) + data[6]) & 0x3FFF;
- rtp_header->type.Video.height = ((data[9] << 8) + data[8]) & 0x3FFF;
+ parsed_payload->type.Video.width = ((data[7] << 8) + data[6]) & 0x3FFF;
+ parsed_payload->type.Video.height = ((data[9] << 8) + data[8]) & 0x3FFF;
return 0;
}
} // namespace
@@ -664,27 +664,27 @@
const uint8_t* payload_data,
size_t payload_data_length) {
assert(parsed_payload != NULL);
- assert(parsed_payload->header != NULL);
// Parse mandatory first byte of payload descriptor.
bool extension = (*payload_data & 0x80) ? true : false; // X bit
bool beginning_of_partition = (*payload_data & 0x10) ? true : false; // S bit
int partition_id = (*payload_data & 0x0F); // PartID field
- parsed_payload->header->type.Video.isFirstPacket =
+ parsed_payload->type.Video.width = 0;
+ parsed_payload->type.Video.height = 0;
+ parsed_payload->type.Video.isFirstPacket =
beginning_of_partition && (partition_id == 0);
-
- parsed_payload->header->type.Video.codecHeader.VP8.nonReference =
+ parsed_payload->type.Video.codec = kRtpVideoVp8;
+ parsed_payload->type.Video.codecHeader.VP8.nonReference =
(*payload_data & 0x20) ? true : false; // N bit
- parsed_payload->header->type.Video.codecHeader.VP8.partitionId = partition_id;
- parsed_payload->header->type.Video.codecHeader.VP8.beginningOfPartition =
+ parsed_payload->type.Video.codecHeader.VP8.partitionId = partition_id;
+ parsed_payload->type.Video.codecHeader.VP8.beginningOfPartition =
beginning_of_partition;
- parsed_payload->header->type.Video.codecHeader.VP8.pictureId = kNoPictureId;
- parsed_payload->header->type.Video.codecHeader.VP8.tl0PicIdx = kNoTl0PicIdx;
- parsed_payload->header->type.Video.codecHeader.VP8.temporalIdx =
- kNoTemporalIdx;
- parsed_payload->header->type.Video.codecHeader.VP8.layerSync = false;
- parsed_payload->header->type.Video.codecHeader.VP8.keyIdx = kNoKeyIdx;
+ parsed_payload->type.Video.codecHeader.VP8.pictureId = kNoPictureId;
+ parsed_payload->type.Video.codecHeader.VP8.tl0PicIdx = kNoTl0PicIdx;
+ parsed_payload->type.Video.codecHeader.VP8.temporalIdx = kNoTemporalIdx;
+ parsed_payload->type.Video.codecHeader.VP8.layerSync = false;
+ parsed_payload->type.Video.codecHeader.VP8.keyIdx = kNoKeyIdx;
if (partition_id > 8) {
// Weak check for corrupt payload_data: PartID MUST NOT be larger than 8.
@@ -697,7 +697,7 @@
if (extension) {
const int parsed_bytes =
- ParseVP8Extension(&parsed_payload->header->type.Video.codecHeader.VP8,
+ ParseVP8Extension(&parsed_payload->type.Video.codecHeader.VP8,
payload_data,
payload_data_length);
if (parsed_bytes < 0)
@@ -713,14 +713,14 @@
// Read P bit from payload header (only at beginning of first partition).
if (payload_data_length > 0 && beginning_of_partition && partition_id == 0) {
- parsed_payload->header->frameType =
+ parsed_payload->frame_type =
(*payload_data & 0x01) ? kVideoFrameDelta : kVideoFrameKey;
} else {
- parsed_payload->header->frameType = kVideoFrameDelta;
+ parsed_payload->frame_type = kVideoFrameDelta;
}
- if (0 != ParseVP8FrameSize(
- parsed_payload->header, payload_data, payload_data_length)) {
+ if (ParseVP8FrameSize(parsed_payload, payload_data, payload_data_length) !=
+ 0) {
return false;
}
diff --git a/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
index b13f879..4382ac2 100644
--- a/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -56,24 +56,23 @@
// | padding |
// : :
// +-+-+-+-+-+-+-+-+
-
-void VerifyBasicHeader(WebRtcRTPHeader* header, bool N, bool S, int part_id) {
- ASSERT_TRUE(header != NULL);
- EXPECT_EQ(N, header->type.Video.codecHeader.VP8.nonReference);
- EXPECT_EQ(S, header->type.Video.codecHeader.VP8.beginningOfPartition);
- EXPECT_EQ(part_id, header->type.Video.codecHeader.VP8.partitionId);
+void VerifyBasicHeader(RTPTypeHeader* type, bool N, bool S, int part_id) {
+ ASSERT_TRUE(type != NULL);
+ EXPECT_EQ(N, type->Video.codecHeader.VP8.nonReference);
+ EXPECT_EQ(S, type->Video.codecHeader.VP8.beginningOfPartition);
+ EXPECT_EQ(part_id, type->Video.codecHeader.VP8.partitionId);
}
-void VerifyExtensions(WebRtcRTPHeader* header,
+void VerifyExtensions(RTPTypeHeader* type,
int16_t picture_id, /* I */
int16_t tl0_pic_idx, /* L */
uint8_t temporal_idx, /* T */
int key_idx /* K */) {
- ASSERT_TRUE(header != NULL);
- EXPECT_EQ(picture_id, header->type.Video.codecHeader.VP8.pictureId);
- EXPECT_EQ(tl0_pic_idx, header->type.Video.codecHeader.VP8.tl0PicIdx);
- EXPECT_EQ(temporal_idx, header->type.Video.codecHeader.VP8.temporalIdx);
- EXPECT_EQ(key_idx, header->type.Video.codecHeader.VP8.keyIdx);
+ ASSERT_TRUE(type != NULL);
+ EXPECT_EQ(picture_id, type->Video.codecHeader.VP8.pictureId);
+ EXPECT_EQ(tl0_pic_idx, type->Video.codecHeader.VP8.tl0PicIdx);
+ EXPECT_EQ(temporal_idx, type->Video.codecHeader.VP8.temporalIdx);
+ EXPECT_EQ(key_idx, type->Video.codecHeader.VP8.keyIdx);
}
} // namespace
@@ -405,18 +404,16 @@
uint8_t packet[4] = {0};
packet[0] = 0x14; // Binary 0001 0100; S = 1, PartID = 4.
packet[1] = 0x01; // P frame.
-
- WebRtcRTPHeader rtp_header;
- memset(&rtp_header, 0, sizeof(rtp_header));
- RtpDepacketizer::ParsedPayload payload(&rtp_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(
&payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
- EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
- VerifyBasicHeader(payload.header, 0, 1, 4);
+ EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+ EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+ VerifyBasicHeader(&payload.type, 0, 1, 4);
VerifyExtensions(
- payload.header, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
+ &payload.type, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
}
TEST_F(RtpDepacketizerVp8Test, PictureID) {
@@ -427,29 +424,27 @@
packet[0] = 0xA0;
packet[1] = 0x80;
packet[2] = kPictureId;
-
- WebRtcRTPHeader rtp_header;
- memset(&rtp_header, 0, sizeof(rtp_header));
- RtpDepacketizer::ParsedPayload payload(&rtp_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(
&payload, packet + kHeaderLength1, sizeof(packet) - kHeaderLength1);
- EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
- VerifyBasicHeader(payload.header, 1, 0, 0);
+ EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+ EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+ VerifyBasicHeader(&payload.type, 1, 0, 0);
VerifyExtensions(
- payload.header, kPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
+ &payload.type, kPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
// Re-use packet, but change to long PictureID.
packet[2] = 0x80 | kPictureId;
packet[3] = kPictureId;
- memset(payload.header, 0, sizeof(rtp_header));
+ payload = RtpDepacketizer::ParsedPayload();
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(
&payload, packet + kHeaderLength2, sizeof(packet) - kHeaderLength2);
- VerifyBasicHeader(payload.header, 1, 0, 0);
- VerifyExtensions(payload.header,
+ VerifyBasicHeader(&payload.type, 1, 0, 0);
+ VerifyExtensions(&payload.type,
(kPictureId << 8) + kPictureId,
kNoTl0PicIdx,
kNoTemporalIdx,
@@ -463,18 +458,16 @@
packet[0] = 0x90;
packet[1] = 0x40;
packet[2] = kTl0PicIdx;
-
- WebRtcRTPHeader rtp_header;
- memset(&rtp_header, 0, sizeof(rtp_header));
- RtpDepacketizer::ParsedPayload payload(&rtp_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(
&payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
- EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
- VerifyBasicHeader(payload.header, 0, 1, 0);
+ EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+ EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+ VerifyBasicHeader(&payload.type, 0, 1, 0);
VerifyExtensions(
- payload.header, kNoPictureId, kTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
+ &payload.type, kNoPictureId, kTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
}
TEST_F(RtpDepacketizerVp8Test, TIDAndLayerSync) {
@@ -483,18 +476,16 @@
packet[0] = 0x88;
packet[1] = 0x20;
packet[2] = 0x80; // TID(2) + LayerSync(false)
-
- WebRtcRTPHeader rtp_header;
- memset(&rtp_header, 0, sizeof(rtp_header));
- RtpDepacketizer::ParsedPayload payload(&rtp_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(
&payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
- EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
- VerifyBasicHeader(payload.header, 0, 0, 8);
- VerifyExtensions(payload.header, kNoPictureId, kNoTl0PicIdx, 2, kNoKeyIdx);
- EXPECT_FALSE(payload.header->type.Video.codecHeader.VP8.layerSync);
+ EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+ EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+ VerifyBasicHeader(&payload.type, 0, 0, 8);
+ VerifyExtensions(&payload.type, kNoPictureId, kNoTl0PicIdx, 2, kNoKeyIdx);
+ EXPECT_FALSE(payload.type.Video.codecHeader.VP8.layerSync);
}
TEST_F(RtpDepacketizerVp8Test, KeyIdx) {
@@ -504,18 +495,16 @@
packet[0] = 0x88;
packet[1] = 0x10; // K = 1.
packet[2] = kKeyIdx;
-
- WebRtcRTPHeader rtp_header;
- memset(&rtp_header, 0, sizeof(rtp_header));
- RtpDepacketizer::ParsedPayload payload(&rtp_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(
&payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
- EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
- VerifyBasicHeader(payload.header, 0, 0, 8);
+ EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+ EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+ VerifyBasicHeader(&payload.type, 0, 0, 8);
VerifyExtensions(
- payload.header, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kKeyIdx);
+ &payload.type, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kKeyIdx);
}
TEST_F(RtpDepacketizerVp8Test, MultipleExtensions) {
@@ -527,17 +516,15 @@
packet[3] = 17; // PictureID, low 8 bits.
packet[4] = 42; // Tl0PicIdx.
packet[5] = 0x40 | 0x20 | 0x11; // TID(1) + LayerSync(true) + KEYIDX(17).
-
- WebRtcRTPHeader rtp_header;
- memset(&rtp_header, 0, sizeof(rtp_header));
- RtpDepacketizer::ParsedPayload payload(&rtp_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(
&payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
- EXPECT_EQ(kVideoFrameDelta, payload.header->frameType);
- VerifyBasicHeader(payload.header, 0, 0, 8);
- VerifyExtensions(payload.header, (17 << 8) + 17, 42, 1, 17);
+ EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
+ EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+ VerifyBasicHeader(&payload.type, 0, 0, 8);
+ VerifyExtensions(&payload.type, (17 << 8) + 17, 42, 1, 17);
}
TEST_F(RtpDepacketizerVp8Test, TooShortHeader) {
@@ -546,10 +533,7 @@
packet[1] = 0x80 | 0x40 | 0x20 | 0x10; // All extensions are enabled...
packet[2] = 0x80 | 17; // ... but only 2 bytes PictureID is provided.
packet[3] = 17; // PictureID, low 8 bits.
-
- WebRtcRTPHeader rtp_header;
- memset(&rtp_header, 0, sizeof(rtp_header));
- RtpDepacketizer::ParsedPayload payload(&rtp_header);
+ RtpDepacketizer::ParsedPayload payload;
EXPECT_FALSE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
}
@@ -571,23 +555,20 @@
size_t send_bytes;
ASSERT_TRUE(packetizer.NextPacket(packet, &send_bytes, &last));
ASSERT_TRUE(last);
-
- WebRtcRTPHeader rtp_header;
- memset(&rtp_header, 0, sizeof(rtp_header));
- RtpDepacketizer::ParsedPayload payload(&rtp_header);
+ RtpDepacketizer::ParsedPayload payload;
ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
ExpectPacket(
&payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
- EXPECT_EQ(kVideoFrameKey, payload.header->frameType);
- VerifyBasicHeader(payload.header, 1, 1, 0);
- VerifyExtensions(payload.header,
+ EXPECT_EQ(kVideoFrameKey, payload.frame_type);
+ EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
+ VerifyBasicHeader(&payload.type, 1, 1, 0);
+ VerifyExtensions(&payload.type,
input_header.pictureId,
input_header.tl0PicIdx,
input_header.temporalIdx,
input_header.keyIdx);
- EXPECT_EQ(payload.header->type.Video.codecHeader.VP8.layerSync,
+ EXPECT_EQ(payload.type.Video.codecHeader.VP8.layerSync,
input_header.layerSync);
}
-
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.cc b/modules/rtp_rtcp/source/rtp_receiver_video.cc
index dfbf35a..6f6d647 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -79,13 +79,15 @@
}
rtp_header->type.Video.isFirstPacket = is_first_packet;
- RtpDepacketizer::ParsedPayload parsed_payload(rtp_header);
+ RtpDepacketizer::ParsedPayload parsed_payload;
if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
return -1;
+ rtp_header->frameType = parsed_payload.frame_type;
+ rtp_header->type = parsed_payload.type;
return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
parsed_payload.payload_length,
- parsed_payload.header) == 0
+ rtp_header) == 0
? 0
: -1;
}
diff --git a/p2p/base/basicpacketsocketfactory.cc b/p2p/base/basicpacketsocketfactory.cc
index 06dfe76..9b12e78 100644
--- a/p2p/base/basicpacketsocketfactory.cc
+++ b/p2p/base/basicpacketsocketfactory.cc
@@ -44,7 +44,7 @@
}
AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
- const SocketAddress& address, int min_port, int max_port) {
+ const SocketAddress& address, uint16 min_port, uint16 max_port) {
// UDP sockets are simple.
rtc::AsyncSocket* socket =
socket_factory()->CreateAsyncSocket(
@@ -62,7 +62,8 @@
}
AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
- const SocketAddress& local_address, int min_port, int max_port, int opts) {
+ const SocketAddress& local_address, uint16 min_port, uint16 max_port,
+ int opts) {
// Fail if TLS is required.
if (opts & PacketSocketFactory::OPT_TLS) {
@@ -177,7 +178,7 @@
int BasicPacketSocketFactory::BindSocket(
AsyncSocket* socket, const SocketAddress& local_address,
- int min_port, int max_port) {
+ uint16 min_port, uint16 max_port) {
int ret = -1;
if (min_port == 0 && max_port == 0) {
// If there's no port range, let the OS pick a port for us.
diff --git a/p2p/base/basicpacketsocketfactory.h b/p2p/base/basicpacketsocketfactory.h
index fb3a526..b23a677 100644
--- a/p2p/base/basicpacketsocketfactory.h
+++ b/p2p/base/basicpacketsocketfactory.h
@@ -24,21 +24,28 @@
BasicPacketSocketFactory();
explicit BasicPacketSocketFactory(Thread* thread);
explicit BasicPacketSocketFactory(SocketFactory* socket_factory);
- virtual ~BasicPacketSocketFactory();
+ ~BasicPacketSocketFactory() override;
- virtual AsyncPacketSocket* CreateUdpSocket(
- const SocketAddress& local_address, int min_port, int max_port);
- virtual AsyncPacketSocket* CreateServerTcpSocket(
- const SocketAddress& local_address, int min_port, int max_port, int opts);
- virtual AsyncPacketSocket* CreateClientTcpSocket(
- const SocketAddress& local_address, const SocketAddress& remote_address,
- const ProxyInfo& proxy_info, const std::string& user_agent, int opts);
+ AsyncPacketSocket* CreateUdpSocket(const SocketAddress& local_address,
+ uint16 min_port,
+ uint16 max_port) override;
+ AsyncPacketSocket* CreateServerTcpSocket(const SocketAddress& local_address,
+ uint16 min_port,
+ uint16 max_port,
+ int opts) override;
+ AsyncPacketSocket* CreateClientTcpSocket(const SocketAddress& local_address,
+ const SocketAddress& remote_address,
+ const ProxyInfo& proxy_info,
+ const std::string& user_agent,
+ int opts) override;
- virtual AsyncResolverInterface* CreateAsyncResolver();
+ AsyncResolverInterface* CreateAsyncResolver() override;
private:
- int BindSocket(AsyncSocket* socket, const SocketAddress& local_address,
- int min_port, int max_port);
+ int BindSocket(AsyncSocket* socket,
+ const SocketAddress& local_address,
+ uint16 min_port,
+ uint16 max_port);
SocketFactory* socket_factory();
diff --git a/p2p/base/packetsocketfactory.h b/p2p/base/packetsocketfactory.h
index 1f45fec..d2d7b1b 100644
--- a/p2p/base/packetsocketfactory.h
+++ b/p2p/base/packetsocketfactory.h
@@ -29,17 +29,23 @@
PacketSocketFactory() { }
virtual ~PacketSocketFactory() { }
- virtual AsyncPacketSocket* CreateUdpSocket(
- const SocketAddress& address, int min_port, int max_port) = 0;
+ virtual AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
+ uint16 min_port,
+ uint16 max_port) = 0;
virtual AsyncPacketSocket* CreateServerTcpSocket(
- const SocketAddress& local_address, int min_port, int max_port,
+ const SocketAddress& local_address,
+ uint16 min_port,
+ uint16 max_port,
int opts) = 0;
// TODO: |proxy_info| and |user_agent| should be set
// per-factory and not when socket is created.
virtual AsyncPacketSocket* CreateClientTcpSocket(
- const SocketAddress& local_address, const SocketAddress& remote_address,
- const ProxyInfo& proxy_info, const std::string& user_agent, int opts) = 0;
+ const SocketAddress& local_address,
+ const SocketAddress& remote_address,
+ const ProxyInfo& proxy_info,
+ const std::string& user_agent,
+ int opts) = 0;
virtual AsyncResolverInterface* CreateAsyncResolver() = 0;
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index f569d9f..a8357ad 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -152,9 +152,12 @@
return rtc::ToString<uint32>(rtc::ComputeCrc32(ost.str()));
}
-Port::Port(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- const std::string& username_fragment, const std::string& password)
+Port::Port(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ const std::string& username_fragment,
+ const std::string& password)
: thread_(thread),
factory_(factory),
send_retransmit_count_attribute_(false),
@@ -176,10 +179,14 @@
Construct();
}
-Port::Port(rtc::Thread* thread, const std::string& type,
+Port::Port(rtc::Thread* thread,
+ const std::string& type,
rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- int min_port, int max_port, const std::string& username_fragment,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username_fragment,
const std::string& password)
: thread_(thread),
factory_(factory),
diff --git a/p2p/base/port.h b/p2p/base/port.h
index 48b8530..87072e6 100644
--- a/p2p/base/port.h
+++ b/p2p/base/port.h
@@ -107,13 +107,20 @@
class Port : public PortInterface, public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
- Port(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- const std::string& username_fragment, const std::string& password);
- Port(rtc::Thread* thread, const std::string& type,
+ Port(rtc::Thread* thread,
rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- int min_port, int max_port, const std::string& username_fragment,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ const std::string& username_fragment,
+ const std::string& password);
+ Port(rtc::Thread* thread,
+ const std::string& type,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username_fragment,
const std::string& password);
virtual ~Port();
@@ -256,8 +263,8 @@
// Debugging description of this port
virtual std::string ToString() const;
rtc::IPAddress& ip() { return ip_; }
- int min_port() { return min_port_; }
- int max_port() { return max_port_; }
+ uint16 min_port() { return min_port_; }
+ uint16 max_port() { return max_port_; }
// Timeout shortening function to speed up unit tests.
void set_timeout_delay(int delay) { timeout_delay_ = delay; }
@@ -354,8 +361,8 @@
bool send_retransmit_count_attribute_;
rtc::Network* network_;
rtc::IPAddress ip_;
- int min_port_;
- int max_port_;
+ uint16 min_port_;
+ uint16 max_port_;
std::string content_name_;
int component_;
uint32 generation_;
diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc
index 8805709..f09db28 100644
--- a/p2p/base/port_unittest.cc
+++ b/p2p/base/port_unittest.cc
@@ -100,12 +100,17 @@
// Stub port class for testing STUN generation and processing.
class TestPort : public Port {
public:
- TestPort(rtc::Thread* thread, const std::string& type,
- rtc::PacketSocketFactory* factory, rtc::Network* network,
- const rtc::IPAddress& ip, int min_port, int max_port,
- const std::string& username_fragment, const std::string& password)
- : Port(thread, type, factory, network, ip,
- min_port, max_port, username_fragment, password) {
+ TestPort(rtc::Thread* thread,
+ const std::string& type,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username_fragment,
+ const std::string& password)
+ : Port(thread, type, factory, network, ip, min_port, max_port,
+ username_fragment, password) {
}
~TestPort() {}
@@ -762,19 +767,21 @@
next_server_tcp_socket_(NULL),
next_client_tcp_socket_(NULL) {
}
- virtual ~FakePacketSocketFactory() { }
+ ~FakePacketSocketFactory() override { }
- virtual AsyncPacketSocket* CreateUdpSocket(
- const SocketAddress& address, int min_port, int max_port) {
+ AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
+ uint16 min_port,
+ uint16 max_port) override {
EXPECT_TRUE(next_udp_socket_ != NULL);
AsyncPacketSocket* result = next_udp_socket_;
next_udp_socket_ = NULL;
return result;
}
- virtual AsyncPacketSocket* CreateServerTcpSocket(
- const SocketAddress& local_address, int min_port, int max_port,
- int opts) {
+ AsyncPacketSocket* CreateServerTcpSocket(const SocketAddress& local_address,
+ uint16 min_port,
+ uint16 max_port,
+ int opts) override {
EXPECT_TRUE(next_server_tcp_socket_ != NULL);
AsyncPacketSocket* result = next_server_tcp_socket_;
next_server_tcp_socket_ = NULL;
@@ -783,10 +790,11 @@
// TODO: |proxy_info| and |user_agent| should be set
// per-factory and not when socket is created.
- virtual AsyncPacketSocket* CreateClientTcpSocket(
- const SocketAddress& local_address, const SocketAddress& remote_address,
- const rtc::ProxyInfo& proxy_info,
- const std::string& user_agent, int opts) {
+ AsyncPacketSocket* CreateClientTcpSocket(const SocketAddress& local_address,
+ const SocketAddress& remote_address,
+ const rtc::ProxyInfo& proxy_info,
+ const std::string& user_agent,
+ int opts) override {
EXPECT_TRUE(next_client_tcp_socket_ != NULL);
AsyncPacketSocket* result = next_client_tcp_socket_;
next_client_tcp_socket_ = NULL;
diff --git a/p2p/base/relayport.cc b/p2p/base/relayport.cc
index 4c40b3d..1a07f8f 100644
--- a/p2p/base/relayport.cc
+++ b/p2p/base/relayport.cc
@@ -172,11 +172,14 @@
uint32 start_time_;
};
-RelayPort::RelayPort(
- rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- int min_port, int max_port, const std::string& username,
- const std::string& password)
+RelayPort::RelayPort(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username,
+ const std::string& password)
: Port(thread, RELAY_PORT_TYPE, factory, network, ip, min_port, max_port,
username, password),
ready_(false),
diff --git a/p2p/base/relayport.h b/p2p/base/relayport.h
index 3d9538d..6297142 100644
--- a/p2p/base/relayport.h
+++ b/p2p/base/relayport.h
@@ -36,9 +36,13 @@
// RelayPort doesn't yet do anything fancy in the ctor.
static RelayPort* Create(
- rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- int min_port, int max_port, const std::string& username,
+ rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username,
const std::string& password) {
return new RelayPort(thread, factory, network, ip, min_port, max_port,
username, password);
@@ -66,9 +70,13 @@
sigslot::signal1<const ProtocolAddress*> SignalSoftTimeout;
protected:
- RelayPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network*, const rtc::IPAddress& ip,
- int min_port, int max_port, const std::string& username,
+ RelayPort(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network*,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username,
const std::string& password);
bool Init();
diff --git a/p2p/base/stunport.cc b/p2p/base/stunport.cc
index ec6232a..5ef9e9e 100644
--- a/p2p/base/stunport.cc
+++ b/p2p/base/stunport.cc
@@ -162,7 +162,8 @@
rtc::PacketSocketFactory* factory,
rtc::Network* network,
rtc::AsyncPacketSocket* socket,
- const std::string& username, const std::string& password)
+ const std::string& username,
+ const std::string& password)
: Port(thread, factory, network, socket->GetLocalAddress().ipaddr(),
username, password),
requests_(thread),
@@ -175,8 +176,11 @@
UDPPort::UDPPort(rtc::Thread* thread,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
- const rtc::IPAddress& ip, int min_port, int max_port,
- const std::string& username, const std::string& password)
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username,
+ const std::string& password)
: Port(thread, LOCAL_PORT_TYPE, factory, network, ip, min_port, max_port,
username, password),
requests_(thread),
diff --git a/p2p/base/stunport.h b/p2p/base/stunport.h
index eda7bb9..9ca6046 100644
--- a/p2p/base/stunport.h
+++ b/p2p/base/stunport.h
@@ -34,8 +34,8 @@
rtc::AsyncPacketSocket* socket,
const std::string& username,
const std::string& password) {
- UDPPort* port = new UDPPort(thread, factory, network, socket,
- username, password);
+ UDPPort* port =
+ new UDPPort(thread, factory, network, socket, username, password);
if (!port->Init()) {
delete port;
port = NULL;
@@ -47,12 +47,12 @@
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
- int min_port, int max_port,
+ uint16 min_port,
+ uint16 max_port,
const std::string& username,
const std::string& password) {
- UDPPort* port = new UDPPort(thread, factory, network,
- ip, min_port, max_port,
- username, password);
+ UDPPort* port = new UDPPort(thread, factory, network, ip, min_port,
+ max_port, username, password);
if (!port->Init()) {
delete port;
port = NULL;
@@ -98,14 +98,21 @@
}
protected:
- UDPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- int min_port, int max_port,
- const std::string& username, const std::string& password);
+ UDPPort(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username,
+ const std::string& password);
- UDPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network* network, rtc::AsyncPacketSocket* socket,
- const std::string& username, const std::string& password);
+ UDPPort(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ rtc::AsyncPacketSocket* socket,
+ const std::string& username,
+ const std::string& password);
bool Init();
@@ -194,18 +201,16 @@
class StunPort : public UDPPort {
public:
- static StunPort* Create(
- rtc::Thread* thread,
- rtc::PacketSocketFactory* factory,
- rtc::Network* network,
- const rtc::IPAddress& ip,
- int min_port, int max_port,
- const std::string& username,
- const std::string& password,
- const ServerAddresses& servers) {
- StunPort* port = new StunPort(thread, factory, network,
- ip, min_port, max_port,
- username, password, servers);
+ static StunPort* Create(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port, uint16 max_port,
+ const std::string& username,
+ const std::string& password,
+ const ServerAddresses& servers) {
+ StunPort* port = new StunPort(thread, factory, network, ip, min_port,
+ max_port, username, password, servers);
if (!port->Init()) {
delete port;
port = NULL;
@@ -220,10 +225,14 @@
}
protected:
- StunPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- int min_port, int max_port,
- const std::string& username, const std::string& password,
+ StunPort(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username,
+ const std::string& password,
const ServerAddresses& servers)
: UDPPort(thread, factory, network, ip, min_port, max_port, username,
password) {
diff --git a/p2p/base/tcpport.cc b/p2p/base/tcpport.cc
index be3068b..b37f4d3 100644
--- a/p2p/base/tcpport.cc
+++ b/p2p/base/tcpport.cc
@@ -18,9 +18,13 @@
TCPPort::TCPPort(rtc::Thread* thread,
rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- int min_port, int max_port, const std::string& username,
- const std::string& password, bool allow_listen)
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username,
+ const std::string& password,
+ bool allow_listen)
: Port(thread, LOCAL_PORT_TYPE, factory, network, ip, min_port, max_port,
username, password),
incoming_only_(false),
diff --git a/p2p/base/tcpport.h b/p2p/base/tcpport.h
index 43e4936..b3655a8 100644
--- a/p2p/base/tcpport.h
+++ b/p2p/base/tcpport.h
@@ -32,13 +32,13 @@
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
- int min_port, int max_port,
+ uint16 min_port,
+ uint16 max_port,
const std::string& username,
const std::string& password,
bool allow_listen) {
- TCPPort* port = new TCPPort(thread, factory, network,
- ip, min_port, max_port,
- username, password, allow_listen);
+ TCPPort* port = new TCPPort(thread, factory, network, ip, min_port,
+ max_port, username, password, allow_listen);
if (!port->Init()) {
delete port;
port = NULL;
@@ -57,10 +57,15 @@
virtual int GetError();
protected:
- TCPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
- rtc::Network* network, const rtc::IPAddress& ip,
- int min_port, int max_port, const std::string& username,
- const std::string& password, bool allow_listen);
+ TCPPort(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16 min_port,
+ uint16 max_port,
+ const std::string& username,
+ const std::string& password,
+ bool allow_listen);
bool Init();
// Handles sending using the local TCP socket.
diff --git a/p2p/base/turnport.cc b/p2p/base/turnport.cc
index e7626fe..fbdcfeb 100644
--- a/p2p/base/turnport.cc
+++ b/p2p/base/turnport.cc
@@ -184,7 +184,8 @@
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
- int min_port, int max_port,
+ uint16 min_port,
+ uint16 max_port,
const std::string& username,
const std::string& password,
const ProtocolAddress& server_address,
diff --git a/p2p/base/turnport.h b/p2p/base/turnport.h
index 17fad17..4ed77a0 100644
--- a/p2p/base/turnport.h
+++ b/p2p/base/turnport.h
@@ -42,16 +42,16 @@
const ProtocolAddress& server_address,
const RelayCredentials& credentials,
int server_priority) {
- return new TurnPort(thread, factory, network, socket,
- username, password, server_address,
- credentials, server_priority);
+ return new TurnPort(thread, factory, network, socket, username, password,
+ server_address, credentials, server_priority);
}
static TurnPort* Create(rtc::Thread* thread,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
- int min_port, int max_port,
+ uint16 min_port,
+ uint16 max_port,
const std::string& username, // ice username.
const std::string& password, // ice password.
const ProtocolAddress& server_address,
@@ -135,7 +135,8 @@
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
- int min_port, int max_port,
+ uint16 min_port,
+ uint16 max_port,
const std::string& username,
const std::string& password,
const ProtocolAddress& server_address,