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Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize

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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}
This commit is contained in:
kwiberg
2015-09-09 00:54:07 -07:00
committed by Commit bot
parent d944067a03
commit c99ebc1490
13 changed files with 0 additions and 158 deletions
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24
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
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@ -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")) {

4
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
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@ -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

34
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
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@ -722,40 +722,6 @@ class AudioCodingModule {
// Codec specific
//
///////////////////////////////////////////////////////////////////////////
// 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

44
webrtc/modules/audio_coding/main/test/iSACTest.cc
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@ -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);

2
webrtc/modules/audio_coding/main/test/iSACTest.h
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@ -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;