Merge "Add Track Sink Format to AudioMixer" (4d3c5bde) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioMixer.cpp

+68 −22

Original line number	Diff line number	Diff line
		@@ -193,6 +193,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
		t->mainBuffer = NULL;
		t->auxBuffer = NULL;
		t->downmixerBufferProvider = NULL;
		t->mSinkFormat = AUDIO_FORMAT_PCM_16_BIT;

		status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
		if (status == OK) {
		@@ -440,6 +441,13 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
		// for a specific track? or per mixer?
		/* case DOWNMIX_TYPE:
		break */
		case SINK_FORMAT: {
		audio_format_t format = static_cast<audio_format_t>(valueInt);
		if (track.mSinkFormat != format) {
		track.mSinkFormat = format;
		ALOGV("setParameter(TRACK, SINK_FORMAT, %#x)", format);
		}
		} break;
		default:
		LOG_FATAL("bad param");
		}
		@@ -1043,7 +1051,7 @@ void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
		void AudioMixer::process__nop(state_t* state, int64_t pts)
		{
		uint32_t e0 = state->enabledTracks;
		size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
		size_t sampleCount = state->frameCount * MAX_NUM_CHANNELS;
		while (e0) {
		// process by group of tracks with same output buffer to
		// avoid multiple memset() on same buffer
		@@ -1062,7 +1070,8 @@ void AudioMixer::process__nop(state_t* state, int64_t pts)
		}
		e0 &= ~(e1);

		memset(t1.mainBuffer, 0, bufSize);
		memset(t1.mainBuffer, 0, sampleCount
		* audio_bytes_per_sample(t1.mSinkFormat));
		}

		while (e1) {
		@@ -1170,8 +1179,18 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
		}
		}
		}
		switch (t1.mSinkFormat) {
		case AUDIO_FORMAT_PCM_FLOAT:
		memcpy_to_float_from_q19_12(reinterpret_cast<float >(out), outTemp, BLOCKSIZE 2);
		out += BLOCKSIZE * 2; // output is 2 floats/frame.
		break;
		case AUDIO_FORMAT_PCM_16_BIT:
		ditherAndClamp(out, outTemp, BLOCKSIZE);
		out += BLOCKSIZE;
		out += BLOCKSIZE; // output is 1 int32_t (2 int16_t samples)/frame
		break;
		default:
		LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
		}
		numFrames += BLOCKSIZE;
		} while (numFrames < state->frameCount);
		}
		@@ -1253,7 +1272,16 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
		}
		}
		}
		switch (t1.mSinkFormat) {
		case AUDIO_FORMAT_PCM_FLOAT:
		memcpy_to_float_from_q19_12(reinterpret_cast<float>(out), outTemp, numFrames2);
		break;
		case AUDIO_FORMAT_PCM_16_BIT:
		ditherAndClamp(out, outTemp, numFrames);
		break;
		default:
		LOG_ALWAYS_FATAL("bad sink format: %d", t1.mSinkFormat);
		}
		}
		}

		@@ -1294,6 +1322,20 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
		}
		size_t outFrames = b.frameCount;

		switch (t.mSinkFormat) {
		case AUDIO_FORMAT_PCM_FLOAT: {
		float fout = reinterpret_cast<float>(out);
		static float scale = 1. / (32768. * 4096.); // exact when inverted
		do {
		uint32_t rl = reinterpret_cast<const uint32_t >(in);
		in += 2;
		int32_t l = mulRL(1, rl, vrl);
		int32_t r = mulRL(0, rl, vrl);
		fout++ = static_cast<float>(l) scale;
		fout++ = static_cast<float>(r) scale;
		} while (--outFrames);
		} break;
		case AUDIO_FORMAT_PCM_16_BIT:
		if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN \|\| uint32_t(vr) > UNITY_GAIN)) {
		// volume is boosted, so we might need to clamp even though
		// we process only one track.
		@@ -1316,6 +1358,10 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
		*out++ = (r<<16) \| (l & 0xFFFF);
		} while (--outFrames);
		}
		break;
		default:
		LOG_ALWAYS_FATAL("bad sink format: %d", t.mSinkFormat);
		}
		numFrames -= b.frameCount;
		t.bufferProvider->releaseBuffer(&b);
		}

services/audioflinger/AudioMixer.h

+4 −1

Original line number	Diff line number	Diff line
		@@ -77,6 +77,7 @@ public:
		MAIN_BUFFER = 0x4002,
		AUX_BUFFER = 0x4003,
		DOWNMIX_TYPE = 0X4004,
		SINK_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT\|16_BIT)
		// for target RESAMPLE
		SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
		// parameter 'value' is the new sample rate in Hz.
		@@ -193,7 +194,9 @@ private:

		int32_t sessionId;

		int32_t padding[2];
		audio_format_t mSinkFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT\|16_BIT)

		int32_t padding[1];

		// 16-byte boundary