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authorkwiberg <kwiberg@webrtc.org>2015-09-09 00:54:07 -0700
committerCommit bot <commit-bot@chromium.org>2015-09-09 07:54:10 +0000
commitc99ebc1490ec689f5932d7731a215ca02ab30af6 (patch)
tree0d3b45fdbd3b20927492b41ad8c1058803bbee12
parentd944067a03e4820586e2ad868da9e54d0fa04e08 (diff)
downloadwebrtc-c99ebc1490ec689f5932d7731a215ca02ab30af6.tar.gz
Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize
And the corresponding ACM methods SetISACMaxRate and SetISACMaxPayloadSize. They were only used in tests. Review URL: https://codereview.webrtc.org/1311533010 Cr-Commit-Position: refs/heads/master@{#9903}
-rw-r--r--webrtc/modules/audio_coding/codecs/audio_encoder.cc4
-rw-r--r--webrtc/modules/audio_coding/codecs/audio_encoder.h10
-rw-r--r--webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc8
-rw-r--r--webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h2
-rw-r--r--webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h2
-rw-r--r--webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h14
-rw-r--r--webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc8
-rw-r--r--webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h2
-rw-r--r--webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc24
-rw-r--r--webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h4
-rw-r--r--webrtc/modules/audio_coding/main/interface/audio_coding_module.h34
-rw-r--r--webrtc/modules/audio_coding/main/test/iSACTest.cc44
-rw-r--r--webrtc/modules/audio_coding/main/test/iSACTest.h2
13 files changed, 0 insertions, 158 deletions
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 8974cf18bc..c0c20bec4d 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -52,8 +52,4 @@ void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
void AudioEncoder::SetTargetBitrate(int target_bps) {}
-void AudioEncoder::SetMaxBitrate(int max_bps) {}
-
-void AudioEncoder::SetMaxPayloadSize(int max_payload_size_bytes) {}
-
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index c053b7fdef..cda9d86f2e 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -138,16 +138,6 @@ class AudioEncoder {
// encoder is free to adjust or disregard the given bitrate (the default
// implementation does the latter).
virtual void SetTargetBitrate(int target_bps);
-
- // Sets the maximum bitrate which must not be exceeded for any packet. The
- // encoder is free to adjust or disregard this value (the default
- // implementation does the latter).
- virtual void SetMaxBitrate(int max_bps);
-
- // Sets an upper limit on the size of packet payloads produced by the
- // encoder. The encoder is free to adjust or disregard this value (the
- // default implementation does the latter).
- virtual void SetMaxPayloadSize(int max_payload_size_bytes);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index ab3bd770e6..2fe58c9ba5 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -199,14 +199,6 @@ void AudioEncoderCng::SetTargetBitrate(int bits_per_second) {
speech_encoder_->SetTargetBitrate(bits_per_second);
}
-void AudioEncoderCng::SetMaxBitrate(int max_bps) {
- speech_encoder_->SetMaxBitrate(max_bps);
-}
-
-void AudioEncoderCng::SetMaxPayloadSize(int max_payload_size_bytes) {
- speech_encoder_->SetMaxPayloadSize(max_payload_size_bytes);
-}
-
AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
size_t frames_to_encode,
size_t max_encoded_bytes,
diff --git a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
index fd2aa129c8..b025bc2e44 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
@@ -67,8 +67,6 @@ class AudioEncoderCng final : public AudioEncoder {
void SetMaxPlaybackRate(int frequency_hz) override;
void SetProjectedPacketLossRate(double fraction) override;
void SetTargetBitrate(int target_bps) override;
- void SetMaxBitrate(int max_bps) override;
- void SetMaxPayloadSize(int max_payload_size_bytes) override;
private:
EncodedInfo EncodePassive(size_t frames_to_encode,
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 5484395ad8..686b45a742 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -67,8 +67,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
- void SetMaxPayloadSize(int max_payload_size_bytes) override;
- void SetMaxBitrate(int max_rate_bps) override;
private:
// This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index ad09c3f90d..3cc635c612 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -158,20 +158,6 @@ void AudioEncoderIsacT<T>::Reset() {
}
template <typename T>
-void AudioEncoderIsacT<T>::SetMaxPayloadSize(int max_payload_size_bytes) {
- auto conf = config_;
- conf.max_payload_size_bytes = max_payload_size_bytes;
- RecreateEncoderInstance(conf);
-}
-
-template <typename T>
-void AudioEncoderIsacT<T>::SetMaxBitrate(int max_rate_bps) {
- auto conf = config_;
- conf.max_bit_rate = max_rate_bps;
- RecreateEncoderInstance(conf);
-}
-
-template <typename T>
void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
CHECK(config.IsOk());
packet_in_progress_ = false;
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 46febf7662..c8ae53fe29 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -121,12 +121,4 @@ void AudioEncoderCopyRed::SetTargetBitrate(int bits_per_second) {
speech_encoder_->SetTargetBitrate(bits_per_second);
}
-void AudioEncoderCopyRed::SetMaxBitrate(int max_bps) {
- speech_encoder_->SetMaxBitrate(max_bps);
-}
-
-void AudioEncoderCopyRed::SetMaxPayloadSize(int max_payload_size_bytes) {
- speech_encoder_->SetMaxPayloadSize(max_payload_size_bytes);
-}
-
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index d0fcd41ed8..1d35f95877 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -54,8 +54,6 @@ class AudioEncoderCopyRed final : public AudioEncoder {
void SetMaxPlaybackRate(int frequency_hz) override;
void SetProjectedPacketLossRate(double fraction) override;
void SetTargetBitrate(int target_bps) override;
- void SetMaxBitrate(int max_bps) override;
- void SetMaxPayloadSize(int max_payload_size_bytes) override;
private:
AudioEncoder* speech_encoder_;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 571a509b0e..cb07cd6eb7 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -739,30 +739,6 @@ int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
return 0;
}
-// TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine.
-int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
-
- if (!HaveValidEncoder("SetISACMaxRate")) {
- return -1;
- }
-
- codec_manager_.CurrentEncoder()->SetMaxBitrate(max_bit_per_sec);
- return 0;
-}
-
-// TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine.
-int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
-
- if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
- return -1;
- }
-
- codec_manager_.CurrentEncoder()->SetMaxPayloadSize(max_size_bytes);
- return 0;
-}
-
int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
CriticalSectionScoped lock(acm_crit_sect_.get());
if (!HaveValidEncoder("SetOpusApplication")) {
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 568bf92d3f..837cd11004 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -185,10 +185,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
int GetNetworkStatistics(NetworkStatistics* statistics) override;
- int SetISACMaxRate(int max_bit_per_sec) override;
-
- int SetISACMaxPayloadSize(int max_size_bytes) override;
-
int SetOpusApplication(OpusApplicationMode application) override;
// If current send codec is Opus, informs it about the maximum playback rate
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index 085dd619ce..0d3d5da818 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -723,40 +723,6 @@ class AudioCodingModule {
//
///////////////////////////////////////////////////////////////////////////
- // int32_t SetISACMaxRate()
- // Set the maximum instantaneous rate of iSAC. For a payload of B bits
- // with a frame-size of T sec the instantaneous rate is B/T bits per
- // second. Therefore, (B/T < |max_rate_bps|) and
- // (B < |max_payload_len_bytes| * 8) are always satisfied for iSAC payloads,
- // c.f SetISACMaxPayloadSize().
- //
- // Input:
- // -max_rate_bps : maximum instantaneous bit-rate given in bits/sec.
- //
- // Return value:
- // -1 if failed to set the maximum rate.
- // 0 if the maximum rate is set successfully.
- //
- virtual int SetISACMaxRate(int max_rate_bps) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int32_t SetISACMaxPayloadSize()
- // Set the maximum payload size of iSAC packets. No iSAC payload,
- // regardless of its frame-size, may exceed the given limit. For
- // an iSAC payload of size B bits and frame-size T seconds we have;
- // (B < |max_payload_len_bytes| * 8) and (B/T < |max_rate_bps|), c.f.
- // SetISACMaxRate().
- //
- // Input:
- // -max_payload_len_bytes : maximum payload size in bytes.
- //
- // Return value:
- // -1 if failed to set the maximum payload-size.
- // 0 if the given length is set successfully.
- //
- virtual int SetISACMaxPayloadSize(int max_payload_len_bytes) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
// int SetOpusApplication()
// Sets the intended application if current send codec is Opus. Opus uses this
// to optimize the encoding for applications like VOIP and music. Currently,
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/main/test/iSACTest.cc
index cc41e3bc1b..bd796d1ce7 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/main/test/iSACTest.cc
@@ -35,8 +35,6 @@ namespace webrtc {
void SetISACConfigDefault(ACMTestISACConfig& isacConfig) {
isacConfig.currentRateBitPerSec = 0;
isacConfig.currentFrameSizeMsec = 0;
- isacConfig.maxRateBitPerSec = 0;
- isacConfig.maxPayloadSizeByte = 0;
isacConfig.encodingMode = -1;
isacConfig.initRateBitPerSec = 0;
isacConfig.initFrameSizeInMsec = 0;
@@ -67,15 +65,6 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
}
}
- if (isacConfig.maxRateBitPerSec > 0) {
- // Set max rate.
- EXPECT_EQ(0, acm->SetISACMaxRate(isacConfig.maxRateBitPerSec));
- }
- if (isacConfig.maxPayloadSizeByte > 0) {
- // Set max payload size.
- EXPECT_EQ(0, acm->SetISACMaxPayloadSize(isacConfig.maxPayloadSizeByte));
- }
-
return 0;
}
@@ -193,39 +182,6 @@ void ISACTest::Perform() {
testNr++;
EncodeDecode(testNr, wbISACConfig, swbISACConfig);
- int user_input;
- if ((_testMode == 0) || (_testMode == 1)) {
- swbISACConfig.maxPayloadSizeByte = static_cast<uint16_t>(200);
- wbISACConfig.maxPayloadSizeByte = static_cast<uint16_t>(200);
- } else {
- printf("Enter the max payload-size for side A: ");
- CHECK_ERROR(scanf("%d", &user_input));
- swbISACConfig.maxPayloadSizeByte = (uint16_t) user_input;
- printf("Enter the max payload-size for side B: ");
- CHECK_ERROR(scanf("%d", &user_input));
- wbISACConfig.maxPayloadSizeByte = (uint16_t) user_input;
- }
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
- SetISACConfigDefault(wbISACConfig);
- SetISACConfigDefault(swbISACConfig);
-
- if ((_testMode == 0) || (_testMode == 1)) {
- swbISACConfig.maxRateBitPerSec = static_cast<uint32_t>(48000);
- wbISACConfig.maxRateBitPerSec = static_cast<uint32_t>(48000);
- } else {
- printf("Enter the max rate for side A: ");
- CHECK_ERROR(scanf("%d", &user_input));
- swbISACConfig.maxRateBitPerSec = (uint32_t) user_input;
- printf("Enter the max rate for side B: ");
- CHECK_ERROR(scanf("%d", &user_input));
- wbISACConfig.maxRateBitPerSec = (uint32_t) user_input;
- }
-
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
testNr++;
if (_testMode == 0) {
SwitchingSamplingRate(testNr, 4);
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.h b/webrtc/modules/audio_coding/main/test/iSACTest.h
index f4223f7512..8f892d907b 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.h
+++ b/webrtc/modules/audio_coding/main/test/iSACTest.h
@@ -29,8 +29,6 @@ namespace webrtc {
struct ACMTestISACConfig {
int32_t currentRateBitPerSec;
int16_t currentFrameSizeMsec;
- uint32_t maxRateBitPerSec;
- int16_t maxPayloadSizeByte;
int16_t encodingMode;
uint32_t initRateBitPerSec;
int16_t initFrameSizeInMsec;